From 65e470e0607bc6c24aa089cf5b64b5ba9301f752 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 24 Mar 2021 16:52:41 +0800 Subject: [PATCH 001/218] =?UTF-8?q?=E5=88=9D=E6=AD=A5=E6=B7=BB=E5=8A=A0rts?= =?UTF-8?q?p=E8=BD=ACwebrtc=E7=9B=B8=E5=85=B3=E5=8A=9F=E8=83=BD?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- CMakeLists.txt | 6 + cmake/FindSRTP.cmake | 55 ++ server/CMakeLists.txt | 1 + server/WebApi.cpp | 24 + webrtc/CMakeLists.txt | 16 + webrtc/dtls_transport.cc | 75 ++ webrtc/dtls_transport.h | 58 ++ webrtc/ice_server.cc | 201 ++++++ webrtc/ice_server.h | 40 + webrtc/logger.h | 18 + webrtc/rtc_dtls_transport.cc | 1323 ++++++++++++++++++++++++++++++++++ webrtc/rtc_dtls_transport.h | 187 +++++ webrtc/srtp_session.cc | 269 +++++++ webrtc/srtp_session.h | 54 ++ webrtc/stun_packet.cc | 710 ++++++++++++++++++ webrtc/stun_packet.h | 179 +++++ webrtc/utils.cc | 139 ++++ webrtc/utils.h | 318 ++++++++ webrtc/webrtc_transport.cc | 215 ++++++ webrtc/webrtc_transport.h | 112 +++ 20 files changed, 4000 insertions(+) create mode 100644 cmake/FindSRTP.cmake create mode 100644 webrtc/CMakeLists.txt create mode 100644 webrtc/dtls_transport.cc create mode 100644 webrtc/dtls_transport.h create mode 100644 webrtc/ice_server.cc create mode 100644 webrtc/ice_server.h create mode 100644 webrtc/logger.h create mode 100644 webrtc/rtc_dtls_transport.cc create mode 100644 webrtc/rtc_dtls_transport.h create mode 100644 webrtc/srtp_session.cc create mode 100644 webrtc/srtp_session.h create mode 100644 webrtc/stun_packet.cc create mode 100644 webrtc/stun_packet.h create mode 100644 webrtc/utils.cc create mode 100644 webrtc/utils.h create mode 100644 webrtc/webrtc_transport.cc create mode 100644 webrtc/webrtc_transport.h diff --git a/CMakeLists.txt b/CMakeLists.txt index 0db8dcea..36a93a12 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -55,6 +55,7 @@ option(ENABLE_TESTS "Enable Tests" true) option(ENABLE_SERVER "Enable Server" true) option(ENABLE_MEM_DEBUG "Enable Memory Debug" false) option(ENABLE_ASAN "Enable Address Sanitize" false) +option(ENABLE_WEBRTC "Enable WebRTC" true) if (ENABLE_MEM_DEBUG) set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -Wl,-wrap,free -Wl,-wrap,malloc -Wl,-wrap,realloc -Wl,-wrap,calloc") @@ -224,6 +225,11 @@ if(ENABLE_API) add_subdirectory(api) endif() +if(ENABLE_WEBRTC) + add_definitions(-DENABLE_WEBRTC) + add_subdirectory(webrtc) +endif() + if (NOT IOS) #测试程序 if(ENABLE_TESTS) diff --git a/cmake/FindSRTP.cmake b/cmake/FindSRTP.cmake new file mode 100644 index 00000000..3046020e --- /dev/null +++ b/cmake/FindSRTP.cmake @@ -0,0 +1,55 @@ +############################################################################ +# FindSRTP.txt +# Copyright (C) 2014 Belledonne Communications, Grenoble France +# +############################################################################ +# +# This program is free software; you can redistribute it and/or +# modify it under the terms of the GNU General Public License +# as published by the Free Software Foundation; either version 2 +# of the License, or (at your option) any later version. +# +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with this program; if not, write to the Free Software +# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +# +############################################################################ +# +# - Find the SRTP include file and library +# +# SRTP_FOUND - system has SRTP +# SRTP_INCLUDE_DIRS - the SRTP include directory +# SRTP_LIBRARIES - The libraries needed to use SRTP + +set(_SRTP_ROOT_PATHS + ${CMAKE_INSTALL_PREFIX} + ) + +find_path(SRTP_INCLUDE_DIRS + NAMES srtp2/srtp.h + HINTS _SRTP_ROOT_PATHS + PATH_SUFFIXES include + ) + +if(SRTP_INCLUDE_DIRS) + set(HAVE_SRTP_SRTP_H 1) +endif() + +find_library(SRTP_LIBRARIES + NAMES srtp2 + HINTS ${_SRTP_ROOT_PATHS} + PATH_SUFFIXES bin lib + ) + +include(FindPackageHandleStandardArgs) +find_package_handle_standard_args(SRTP + DEFAULT_MSG + SRTP_INCLUDE_DIRS SRTP_LIBRARIES HAVE_SRTP_SRTP_H + ) + +mark_as_advanced(SRTP_INCLUDE_DIRS SRTP_LIBRARIES HAVE_SRTP_SRTP_H) \ No newline at end of file diff --git a/server/CMakeLists.txt b/server/CMakeLists.txt index 6e0af5e1..c0b3a009 100644 --- a/server/CMakeLists.txt +++ b/server/CMakeLists.txt @@ -49,3 +49,4 @@ else() endif() target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) +message(${LINK_LIB_LIST}) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index a70b12d1..4d0dba28 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -35,6 +35,9 @@ #if defined(ENABLE_RTPPROXY) #include "Rtp/RtpServer.h" #endif +#ifdef ENABLE_WEBRTC +#include "../webrtc/webrtc_transport.h" +#endif using namespace toolkit; using namespace mediakit; @@ -1049,6 +1052,27 @@ void installWebApi() { #endif }); +#ifdef ENABLE_WEBRTC + static list rtcs; + api_regist("/webrtc",[](API_ARGS_MAP_ASYNC){ + CHECK_ARGS("app", "stream"); + auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, DEFAULT_VHOST, allArgs["app"], allArgs["stream"])); + if (!src) { + throw ApiRetException("流不存在", API::NotFound); + } + headerOut["Content-Type"] = "text/plain"; + headerOut["Access-Control-Allow-Origin"] = "*"; + auto poller = EventPollerPool::Instance().getFirstPoller(); + auto rtc = std::make_shared(poller); + poller->async([invoker, rtc, headerOut, src]() { + rtc->attach(src); + auto sdp = rtc->GetLocalSdp(); + invoker(200, headerOut, sdp); + rtcs.emplace_back(rtc); + }); + }); +#endif + ////////////以下是注册的Hook API//////////// api_regist("/index/hook/on_publish",[](API_ARGS_MAP){ //开始推流事件 diff --git a/webrtc/CMakeLists.txt b/webrtc/CMakeLists.txt new file mode 100644 index 00000000..03039aea --- /dev/null +++ b/webrtc/CMakeLists.txt @@ -0,0 +1,16 @@ +list(APPEND LINK_LIB_LIST webrtc) +#查找srtp是否安装 +find_package(SRTP QUIET) +if (SRTP_FOUND) + message(STATUS "found library:${SRTP_LIBRARIES}") + include_directories(${SRTP_INCLUDE_DIRS}) + list(APPEND LINK_LIB_LIST ${SRTP_LIBRARIES}) +else () + message(FATAL_ERROR "srtp未找到!") +endif () + +include_directories(./) +file(GLOB SRC_LIST ./*.*) +add_library(webrtc ${SRC_LIST}) +set(LINK_LIB_LIST ${LINK_LIB_LIST} PARENT_SCOPE) + diff --git a/webrtc/dtls_transport.cc b/webrtc/dtls_transport.cc new file mode 100644 index 00000000..69e1f402 --- /dev/null +++ b/webrtc/dtls_transport.cc @@ -0,0 +1,75 @@ +// +// Created by xueyuegui on 19-12-7. +// + +#include "dtls_transport.h" + +#include + +DtlsTransport::DtlsTransport(bool is_server) : is_server_(is_server) { + dtls_transport_.reset(new RTC::DtlsTransport(this)); +} + +DtlsTransport::~DtlsTransport() {} + +void DtlsTransport::Start() { + if (is_server_) { + dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); + } else { + dtls_transport_->Run(RTC::DtlsTransport::Role::CLIENT); + } +} + +void DtlsTransport::Close() {} + +void DtlsTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {} + +void DtlsTransport::OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, + RTC::CryptoSuite srtp_crypto_suite, + uint8_t *srtpLocalKey, size_t srtpLocalKeyLen, + uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, + std::string &remoteCert) { + std::string client_key; + std::string server_key; + server_key.assign((char *) srtpLocalKey, srtpLocalKeyLen); + client_key.assign((char *) srtpRemoteKey, srtpRemoteKeyLen); + if (is_server_) { + // If we are server, we swap the keys + client_key.swap(server_key); + } + if (handshake_completed_callback_) { + handshake_completed_callback_(client_key, server_key, srtp_crypto_suite); + } +} + +void DtlsTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) { + if (handshake_failed_callback_) { + handshake_failed_callback_(); + } +} + +void DtlsTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {} + +void DtlsTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, + const uint8_t *data, size_t len) { + if (output_callback_) { + output_callback_((char *) data, len); + } +} + +void DtlsTransport::OutputData(char *buf, size_t len) { + if (output_callback_) { + output_callback_(buf, len); + } +} + +void DtlsTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, + const uint8_t *data, size_t len) {} + +bool DtlsTransport::IsDtlsPacket(const char *buf, size_t len) { + return RTC::DtlsTransport::IsDtls((uint8_t *) buf, len); +} + +void DtlsTransport::InputData(char *buf, size_t len) { + dtls_transport_->ProcessDtlsData((uint8_t *) buf, len); +} diff --git a/webrtc/dtls_transport.h b/webrtc/dtls_transport.h new file mode 100644 index 00000000..41f4c65c --- /dev/null +++ b/webrtc/dtls_transport.h @@ -0,0 +1,58 @@ +// +// Created by xueyuegui on 19-12-7. +// + +#ifndef MYWEBRTC_MYDTLSTRANSPORT_H +#define MYWEBRTC_MYDTLSTRANSPORT_H + +#include +#include + +#include "rtc_dtls_transport.h" + +class DtlsTransport : RTC::DtlsTransport::Listener { +public: + typedef std::shared_ptr Ptr; + + DtlsTransport(bool bServer); + ~DtlsTransport(); + + void Start(); + void Close(); + void InputData(char *buf, size_t len); + void OutputData(char *buf, size_t len); + static bool IsDtlsPacket(const char *buf, size_t len); + std::string GetMyFingerprint() { + auto finger_prints = dtls_transport_->GetLocalFingerprints(); + for (size_t i = 0; i < finger_prints.size(); i++) { + if (finger_prints[i].algorithm == RTC::DtlsTransport::FingerprintAlgorithm::SHA256) { + return finger_prints[i].value; + } + } + return ""; + }; + + void SetHandshakeCompletedCB(std::function cb) { + handshake_completed_callback_ = std::move(cb); + } + void SetHandshakeFailedCB(std::function cb) { handshake_failed_callback_ = std::move(cb); } + void SetOutPutCB(std::function cb) { output_callback_ = std::move(cb); } + + /* Pure virtual methods inherited from RTC::DtlsTransport::Listener. */ +public: + void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override; + void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, RTC::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey, size_t srtpLocalKeyLen, uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, std::string &remoteCert) override; + void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override; + void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override; + void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data,size_t len) override; + void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; + +private: + bool is_server_ = false; + std::function handshake_failed_callback_; + std::shared_ptr dtls_transport_; + std::function output_callback_; + std::function handshake_completed_callback_; +}; + +#endif// MYWEBRTC_MYDTLSTRANSPORT_H diff --git a/webrtc/ice_server.cc b/webrtc/ice_server.cc new file mode 100644 index 00000000..cf7fb24b --- /dev/null +++ b/webrtc/ice_server.cc @@ -0,0 +1,201 @@ +#include "ice_server.h" + +#include + +static constexpr size_t StunSerializeBufferSize{65536}; +static uint8_t StunSerializeBuffer[StunSerializeBufferSize]; + +IceServer::IceServer() {} + +IceServer::~IceServer() {} + +IceServer::IceServer(const std::string &username_fragment, const std::string &password) + : username_fragment_(username_fragment), password_(password) {} + +void IceServer::ProcessStunPacket(RTC::StunPacket *packet, sockaddr_in *remote_address) { + // Must be a Binding method. + if (packet->GetMethod() != RTC::StunPacket::Method::BINDING) { + if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) { + ELOG_WARN("unknown method %#.3x in STUN Request => 400", + static_cast(packet->GetMethod())); + ELOG_WARN("unknown method %#.3x in STUN Request => 400", + static_cast(packet->GetMethod())); + // Reply 400. + RTC::StunPacket *response = packet->CreateErrorResponse(400); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + } else { + ELOG_WARN("ignoring STUN Indication or Response with unknown method %#.3x", + static_cast(packet->GetMethod())); + } + return; + } + + // Must use FINGERPRINT (optional for ICE STUN indications). + if (!packet->HasFingerprint() && packet->GetClass() != RTC::StunPacket::Class::INDICATION) { + if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) { + ELOG_WARN("STUN Binding Request without FINGERPRINT => 400"); + // Reply 400. + RTC::StunPacket *response = packet->CreateErrorResponse(400); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + } else { + ELOG_WARN("ignoring STUN Binding Response without FINGERPRINT"); + } + return; + } + + switch (packet->GetClass()) { + case RTC::StunPacket::Class::REQUEST: { + // USERNAME, MESSAGE-INTEGRITY and PRIORITY are required. + if (!packet->HasMessageIntegrity() || (packet->GetPriority() == 0u) || + packet->GetUsername().empty()) { + ELOG_WARN("mising required attributes in STUN Binding Request => 400"); + + // Reply 400. + RTC::StunPacket *response = packet->CreateErrorResponse(400); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + return; + } + + // Check authentication. + switch (packet->CheckAuthentication(this->username_fragment_, this->password_)) { + case RTC::StunPacket::Authentication::OK: { + if (!this->old_password_.empty()) { + ELOG_DEBUG("kNew ICE credentials applied"); + this->old_username_fragment_.clear(); + this->old_password_.clear(); + } + break; + } + + case RTC::StunPacket::Authentication::UNAUTHORIZED: { + // We may have changed our username_fragment_ and password_, so check + // the old ones. + // clang-format off + if (!this->old_username_fragment_.empty() && + !this->old_password_.empty() && + packet->CheckAuthentication(this->old_username_fragment_, this->old_password_) == + RTC::StunPacket::Authentication::OK) { + ELOG_DEBUG("using old ICE credentials"); + break; + } + ELOG_WARN("wrong authentication in STUN Binding Request => 401"); + // Reply 401. + RTC::StunPacket *response = packet->CreateErrorResponse(401); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + return; + } + + case RTC::StunPacket::Authentication::BAD_REQUEST: { + ELOG_WARN("cannot check authentication in STUN Binding Request => 400"); + // Reply 400. + RTC::StunPacket *response = packet->CreateErrorResponse(400); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + return; + } + } + +#if 0 + // NOTE: Should be rejected with 487, but this makes Chrome happy: + // https://bugs.chromium.org/p/webrtc/issues/detail?id=7478 + // The remote peer must be ICE controlling. + if (packet->GetIceControlled()) { + MS_WARN_TAG(ice, "peer indicates ICE-CONTROLLED in STUN Binding Request => 487"); + // Reply 487 (Role Conflict). + RTC::StunPacket *response = packet->CreateErrorResponse(487); + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + return; + } +#endif + + ELOG_DEBUG("processing STUN Binding Request [Priority:%d, UseCandidate:%s]", + static_cast(packet->GetPriority()), + (packet->HasUseCandidate() ? "true" : "false")); + // Create a success response. + RTC::StunPacket *response = packet->CreateSuccessResponse(); + // Add XOR-MAPPED-ADDRESS. + // response->SetXorMappedAddress(tuple->GetRemoteAddress()); + response->SetXorMappedAddress((struct sockaddr *) remote_address); + // Authenticate the response. + if (this->old_password_.empty()) { + response->Authenticate(this->password_); + } else { + response->Authenticate(this->old_password_); + } + + // Send back. + response->Serialize(StunSerializeBuffer); + if (send_callback_) { + send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); + } + delete response; + // Handle the tuple. + HandleTuple(remote_address, packet->HasUseCandidate()); + break; + } + + case RTC::StunPacket::Class::INDICATION: { + ELOG_DEBUG("STUN Binding Indication processed"); + break; + } + + case RTC::StunPacket::Class::SUCCESS_RESPONSE: { + ELOG_DEBUG("STUN Binding Success Response processed"); + break; + } + + case RTC::StunPacket::Class::ERROR_RESPONSE: { + ELOG_DEBUG("STUN Binding Error Response processed"); + break; + } + } +} +void IceServer::HandleTuple(sockaddr_in *remote_address, bool has_use_candidate) { + remote_address_ = *remote_address; + if (has_use_candidate) { + this->state = IceState::kCompleted; + } + if (ice_server_completed_callback_) { + ice_server_completed_callback_(); + ice_server_completed_callback_ = nullptr; + } +} + +const std::string &IceServer::GetUsernameFragment() const { return this->username_fragment_; } + +const std::string &IceServer::GetPassword() const { return this->password_; } + +inline void IceServer::SetUsernameFragment(const std::string &username_fragment) { + this->old_username_fragment_ = this->username_fragment_; + this->username_fragment_ = username_fragment; +} + +inline void IceServer::SetPassword(const std::string &password) { + this->old_password_ = this->password_; + this->password_ = password; +} + +inline IceServer::IceState IceServer::GetState() const { return this->state; } \ No newline at end of file diff --git a/webrtc/ice_server.h b/webrtc/ice_server.h new file mode 100644 index 00000000..d33f26d2 --- /dev/null +++ b/webrtc/ice_server.h @@ -0,0 +1,40 @@ +#pragma once + +#include +#include + +#include "logger.h" +#include "stun_packet.h" + +typedef std::function UdpSendCallback; + +class IceServer { +public: + enum class IceState { kNew = 1, kConnect, kCompleted, kDisconnected }; + typedef std::shared_ptr Ptr; + IceServer(); + IceServer(const std::string &username_fragment, const std::string &password); + const std::string &GetUsernameFragment() const; + const std::string &GetPassword() const; + void SetUsernameFragment(const std::string &username_fragment); + void SetPassword(const std::string &password); + IceState GetState() const; + void ProcessStunPacket(RTC::StunPacket *packet, struct sockaddr_in *remote_address); + void HandleTuple(struct sockaddr_in *remote_address, bool has_use_candidate); + ~IceServer(); + void SetSendCB(UdpSendCallback send_cb) { send_callback_ = send_cb; } + void SetIceServerCompletedCB(std::function cb) { ice_server_completed_callback_ = cb; }; + struct sockaddr_in *GetSelectAddr() { + return &remote_address_; + } + +private: + UdpSendCallback send_callback_; + std::function ice_server_completed_callback_; + std::string username_fragment_; + std::string password_; + std::string old_username_fragment_; + std::string old_password_; + IceState state{IceState::kNew}; + struct sockaddr_in remote_address_; +}; diff --git a/webrtc/logger.h b/webrtc/logger.h new file mode 100644 index 00000000..a165536b --- /dev/null +++ b/webrtc/logger.h @@ -0,0 +1,18 @@ +#pragma once +#include +#include + +#define ELOG_DEBUG(fmt, ...) printf(fmt "\n", ##__VA_ARGS__) +#define ELOG_WARN(fmt, ...) printf(fmt "\n", ##__VA_ARGS__) + +#define MS_TRACE() +#define MS_ERROR(fmt, ...) printf("error:" fmt "\n", ##__VA_ARGS__) +#define MS_THROW_ERROR(fmt, ...) do{ printf("throw:" fmt "\n", ##__VA_ARGS__); throw std::runtime_error("error"); } while(false); +#define MS_DUMP(fmt, ...) printf("dump:" fmt "\n", ##__VA_ARGS__) +#define MS_DEBUG_2TAGS(tag1, tag2,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) +#define MS_WARN_2TAGS(tag1, tag2,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__) +#define MS_DEBUG_TAG(tag,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) +#define MS_ASSERT(con, log) assert(con) +#define MS_ABORT(fmt, ...) do{ printf("abort:" fmt "\n", ##__VA_ARGS__); abort(); } while(false); +#define MS_WARN_TAG(tag,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__) +#define MS_DEBUG_DEV(fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) \ No newline at end of file diff --git a/webrtc/rtc_dtls_transport.cc b/webrtc/rtc_dtls_transport.cc new file mode 100644 index 00000000..1087af4a --- /dev/null +++ b/webrtc/rtc_dtls_transport.cc @@ -0,0 +1,1323 @@ +#define MS_CLASS "RTC::DtlsTransport" +// #define MS_LOG_DEV_LEVEL 3 + +#include "rtc_dtls_transport.h" + +#include +#include +#include +#include +#include + +#include // std::sprintf(), std::fopen() +#include // std::memcpy(), std::strcmp() + +#include "logger.h" + +typedef struct { + long tv_sec; + long tv_usec; +} uv_timeval_t; + +#define LOG_OPENSSL_ERROR(desc) \ + do { \ + if (ERR_peek_error() == 0) \ + MS_ERROR("OpenSSL error [desc:'%s']", desc); \ + else { \ + int64_t err; \ + while ((err = ERR_get_error()) != 0) { \ + MS_ERROR("OpenSSL error [desc:'%s', error:'%s']", desc, ERR_error_string(err, nullptr)); \ + } \ + ERR_clear_error(); \ + } \ + } while (false) + +/* Static methods for OpenSSL callbacks. */ + +inline static int onSslCertificateVerify(int /*preverifyOk*/, X509_STORE_CTX* /*ctx*/) { + MS_TRACE(); + + // Always valid since DTLS certificates are self-signed. + return 1; +} + +inline static void onSslInfo(const SSL* ssl, int where, int ret) { + static_cast(SSL_get_ex_data(ssl, 0))->OnSslInfo(where, ret); +} + +inline static unsigned int onSslDtlsTimer(SSL* /*ssl*/, unsigned int timerUs) { + if (timerUs == 0) + return 100000; + else if (timerUs >= 4000000) + return 4000000; + else + return 2 * timerUs; +} + +namespace RTC { +/* Static. */ + +// clang-format off + static constexpr int DtlsMtu{ 1350 }; + static constexpr int SslReadBufferSize{ 65536 }; + // AES-HMAC: http://tools.ietf.org/html/rfc3711 + static constexpr size_t SrtpMasterKeyLength{ 16 }; + static constexpr size_t SrtpMasterSaltLength{ 14 }; + static constexpr size_t SrtpMasterLength{ SrtpMasterKeyLength + SrtpMasterSaltLength }; + // AES-GCM: http://tools.ietf.org/html/rfc7714 + static constexpr size_t SrtpAesGcm256MasterKeyLength{ 32 }; + static constexpr size_t SrtpAesGcm256MasterSaltLength{ 12 }; + static constexpr size_t SrtpAesGcm256MasterLength{ SrtpAesGcm256MasterKeyLength + SrtpAesGcm256MasterSaltLength }; + static constexpr size_t SrtpAesGcm128MasterKeyLength{ 16 }; + static constexpr size_t SrtpAesGcm128MasterSaltLength{ 12 }; + static constexpr size_t SrtpAesGcm128MasterLength{ SrtpAesGcm128MasterKeyLength + SrtpAesGcm128MasterSaltLength }; +// clang-format on + +/* Class variables. */ + +X509* DtlsTransport::certificate{nullptr}; +EVP_PKEY* DtlsTransport::privateKey{nullptr}; +SSL_CTX* DtlsTransport::sslCtx{nullptr}; +uint8_t DtlsTransport::sslReadBuffer[SslReadBufferSize]; +// clang-format off + std::map DtlsTransport::string2FingerprintAlgorithm = + { + { "sha-1", DtlsTransport::FingerprintAlgorithm::SHA1 }, + { "sha-224", DtlsTransport::FingerprintAlgorithm::SHA224 }, + { "sha-256", DtlsTransport::FingerprintAlgorithm::SHA256 }, + { "sha-384", DtlsTransport::FingerprintAlgorithm::SHA384 }, + { "sha-512", DtlsTransport::FingerprintAlgorithm::SHA512 } + }; + std::map DtlsTransport::fingerprintAlgorithm2String = + { + { DtlsTransport::FingerprintAlgorithm::SHA1, "sha-1" }, + { DtlsTransport::FingerprintAlgorithm::SHA224, "sha-224" }, + { DtlsTransport::FingerprintAlgorithm::SHA256, "sha-256" }, + { DtlsTransport::FingerprintAlgorithm::SHA384, "sha-384" }, + { DtlsTransport::FingerprintAlgorithm::SHA512, "sha-512" } + }; + std::map DtlsTransport::string2Role = + { + { "auto", DtlsTransport::Role::AUTO }, + { "client", DtlsTransport::Role::CLIENT }, + { "server", DtlsTransport::Role::SERVER } + }; + std::vector DtlsTransport::localFingerprints; + std::vector DtlsTransport::srtpCryptoSuites = + { + { RTC::CryptoSuite::AEAD_AES_256_GCM, "SRTP_AEAD_AES_256_GCM" }, + { RTC::CryptoSuite::AEAD_AES_128_GCM, "SRTP_AEAD_AES_128_GCM" }, + { RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_80, "SRTP_AES128_CM_SHA1_80" }, + { RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_32, "SRTP_AES128_CM_SHA1_32" } + }; +// clang-format on + +/* Class methods. */ + +void DtlsTransport::ClassInit() { + MS_TRACE(); + +#if 0 + // Generate a X509 certificate and private key (unless PEM files are provided). + if (Settings::configuration.dtlsCertificateFile.empty() || + Settings::configuration.dtlsPrivateKeyFile.empty()) { + GenerateCertificateAndPrivateKey(); + } else { + ReadCertificateAndPrivateKeyFromFiles(); + } +#else + GenerateCertificateAndPrivateKey(); +#endif + + // Create a global SSL_CTX. + CreateSslCtx(); + + // Generate certificate fingerprints. + GenerateFingerprints(); +} + +void DtlsTransport::ClassDestroy() { + MS_TRACE(); + + if (DtlsTransport::privateKey) EVP_PKEY_free(DtlsTransport::privateKey); + if (DtlsTransport::certificate) X509_free(DtlsTransport::certificate); + if (DtlsTransport::sslCtx) SSL_CTX_free(DtlsTransport::sslCtx); +} + +void DtlsTransport::GenerateCertificateAndPrivateKey() { + MS_TRACE(); + + int ret{0}; + EC_KEY* ecKey{nullptr}; + X509_NAME* certName{nullptr}; + std::string subject = std::string("mediasoup") + std::to_string(rand() % 999999 + 100000); + // std::string("mediasoup") + std::to_string(Utils::Crypto::GetRandomUInt(100000, 999999)); + + // Create key with curve. + ecKey = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1); + + if (!ecKey) { + LOG_OPENSSL_ERROR("EC_KEY_new_by_curve_name() failed"); + + goto error; + } + + EC_KEY_set_asn1_flag(ecKey, OPENSSL_EC_NAMED_CURVE); + + // NOTE: This can take some time. + ret = EC_KEY_generate_key(ecKey); + + if (ret == 0) { + LOG_OPENSSL_ERROR("EC_KEY_generate_key() failed"); + + goto error; + } + + // Create a private key object. + DtlsTransport::privateKey = EVP_PKEY_new(); + + if (!DtlsTransport::privateKey) { + LOG_OPENSSL_ERROR("EVP_PKEY_new() failed"); + + goto error; + } + + // NOLINTNEXTLINE(cppcoreguidelines-pro-type-cstyle-cast) + ret = EVP_PKEY_assign_EC_KEY(DtlsTransport::privateKey, ecKey); + + if (ret == 0) { + LOG_OPENSSL_ERROR("EVP_PKEY_assign_EC_KEY() failed"); + + goto error; + } + + // The EC key now belongs to the private key, so don't clean it up separately. + ecKey = nullptr; + + // Create the X509 certificate. + DtlsTransport::certificate = X509_new(); + + if (!DtlsTransport::certificate) { + LOG_OPENSSL_ERROR("X509_new() failed"); + + goto error; + } + + // Set version 3 (note that 0 means version 1). + X509_set_version(DtlsTransport::certificate, 2); + + // Set serial number (avoid default 0). + // ASN1_INTEGER_set(X509_get_serialNumber(DtlsTransport::certificate), + // static_cast(Utils::Crypto::GetRandomUInt(1000000, 9999999))); + ASN1_INTEGER_set(X509_get_serialNumber(DtlsTransport::certificate), + static_cast(rand() % 999999 + 100000)); + + // Set valid period. + X509_gmtime_adj(X509_get_notBefore(DtlsTransport::certificate), -315360000); // -10 years. + X509_gmtime_adj(X509_get_notAfter(DtlsTransport::certificate), 315360000); // 10 years. + + // Set the public key for the certificate using the key. + ret = X509_set_pubkey(DtlsTransport::certificate, DtlsTransport::privateKey); + + if (ret == 0) { + LOG_OPENSSL_ERROR("X509_set_pubkey() failed"); + + goto error; + } + + // Set certificate fields. + certName = X509_get_subject_name(DtlsTransport::certificate); + + if (!certName) { + LOG_OPENSSL_ERROR("X509_get_subject_name() failed"); + + goto error; + } + + X509_NAME_add_entry_by_txt(certName, "O", MBSTRING_ASC, + reinterpret_cast(subject.c_str()), -1, -1, 0); + X509_NAME_add_entry_by_txt(certName, "CN", MBSTRING_ASC, + reinterpret_cast(subject.c_str()), -1, -1, 0); + + // It is self-signed so set the issuer name to be the same as the subject. + ret = X509_set_issuer_name(DtlsTransport::certificate, certName); + + if (ret == 0) { + LOG_OPENSSL_ERROR("X509_set_issuer_name() failed"); + + goto error; + } + + // Sign the certificate with its own private key. + ret = X509_sign(DtlsTransport::certificate, DtlsTransport::privateKey, EVP_sha1()); + + if (ret == 0) { + LOG_OPENSSL_ERROR("X509_sign() failed"); + + goto error; + } + + return; + +error: + + if (ecKey) EC_KEY_free(ecKey); + + if (DtlsTransport::privateKey) + EVP_PKEY_free(DtlsTransport::privateKey); // NOTE: This also frees the EC key. + + if (DtlsTransport::certificate) X509_free(DtlsTransport::certificate); + + MS_THROW_ERROR("DTLS certificate and private key generation failed"); +} + +void DtlsTransport::ReadCertificateAndPrivateKeyFromFiles() { +#if 0 + MS_TRACE(); + + FILE* file{nullptr}; + + file = fopen(Settings::configuration.dtlsCertificateFile.c_str(), "r"); + + if (!file) { + MS_ERROR("error reading DTLS certificate file: %s", std::strerror(errno)); + + goto error; + } + + DtlsTransport::certificate = PEM_read_X509(file, nullptr, nullptr, nullptr); + + if (!DtlsTransport::certificate) { + LOG_OPENSSL_ERROR("PEM_read_X509() failed"); + + goto error; + } + + fclose(file); + + file = fopen(Settings::configuration.dtlsPrivateKeyFile.c_str(), "r"); + + if (!file) { + MS_ERROR("error reading DTLS private key file: %s", std::strerror(errno)); + + goto error; + } + + DtlsTransport::privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr); + + if (!DtlsTransport::privateKey) { + LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed"); + + goto error; + } + + fclose(file); + + return; + + error: + + MS_THROW_ERROR("error reading DTLS certificate and private key PEM files"); +#endif +} + +void DtlsTransport::CreateSslCtx() { + MS_TRACE(); + + std::string dtlsSrtpCryptoSuites; + int ret; + + /* Set the global DTLS context. */ + + // Both DTLS 1.0 and 1.2 (requires OpenSSL >= 1.1.0). + DtlsTransport::sslCtx = SSL_CTX_new(DTLS_method()); + + if (!DtlsTransport::sslCtx) { + LOG_OPENSSL_ERROR("SSL_CTX_new() failed"); + + goto error; + } + + ret = SSL_CTX_use_certificate(DtlsTransport::sslCtx, DtlsTransport::certificate); + + if (ret == 0) { + LOG_OPENSSL_ERROR("SSL_CTX_use_certificate() failed"); + + goto error; + } + + ret = SSL_CTX_use_PrivateKey(DtlsTransport::sslCtx, DtlsTransport::privateKey); + + if (ret == 0) { + LOG_OPENSSL_ERROR("SSL_CTX_use_PrivateKey() failed"); + + goto error; + } + + ret = SSL_CTX_check_private_key(DtlsTransport::sslCtx); + + if (ret == 0) { + LOG_OPENSSL_ERROR("SSL_CTX_check_private_key() failed"); + + goto error; + } + + // Set options. + SSL_CTX_set_options(DtlsTransport::sslCtx, SSL_OP_CIPHER_SERVER_PREFERENCE | SSL_OP_NO_TICKET | + SSL_OP_SINGLE_ECDH_USE | SSL_OP_NO_QUERY_MTU); + + // Don't use sessions cache. + SSL_CTX_set_session_cache_mode(DtlsTransport::sslCtx, SSL_SESS_CACHE_OFF); + + // Read always as much into the buffer as possible. + // NOTE: This is the default for DTLS, but a bug in non latest OpenSSL + // versions makes this call required. + SSL_CTX_set_read_ahead(DtlsTransport::sslCtx, 1); + + SSL_CTX_set_verify_depth(DtlsTransport::sslCtx, 4); + + // Require certificate from peer. + SSL_CTX_set_verify(DtlsTransport::sslCtx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, + onSslCertificateVerify); + + // Set SSL info callback. + SSL_CTX_set_info_callback(DtlsTransport::sslCtx, onSslInfo); + + // Set ciphers. + ret = SSL_CTX_set_cipher_list(DtlsTransport::sslCtx, + "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); + + if (ret == 0) { + LOG_OPENSSL_ERROR("SSL_CTX_set_cipher_list() failed"); + + goto error; + } + + // Enable ECDH ciphers. + // DOC: http://en.wikibooks.org/wiki/OpenSSL/Diffie-Hellman_parameters + // NOTE: https://code.google.com/p/chromium/issues/detail?id=406458 + // NOTE: https://bugs.ruby-lang.org/issues/12324 + + // For OpenSSL >= 1.0.2. + SSL_CTX_set_ecdh_auto(DtlsTransport::sslCtx, 1); + + // Set the "use_srtp" DTLS extension. + for (auto it = DtlsTransport::srtpCryptoSuites.begin(); + it != DtlsTransport::srtpCryptoSuites.end(); ++it) { + if (it != DtlsTransport::srtpCryptoSuites.begin()) dtlsSrtpCryptoSuites += ":"; + + SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(*it); + dtlsSrtpCryptoSuites += cryptoSuiteEntry->name; + } + + MS_DEBUG_2TAGS(dtls, srtp, "setting SRTP cryptoSuites for DTLS: %s", + dtlsSrtpCryptoSuites.c_str()); + + // NOTE: This function returns 0 on success. + ret = SSL_CTX_set_tlsext_use_srtp(DtlsTransport::sslCtx, dtlsSrtpCryptoSuites.c_str()); + + if (ret != 0) { + MS_ERROR("SSL_CTX_set_tlsext_use_srtp() failed when entering '%s'", + dtlsSrtpCryptoSuites.c_str()); + LOG_OPENSSL_ERROR("SSL_CTX_set_tlsext_use_srtp() failed"); + + goto error; + } + + return; + +error: + + if (DtlsTransport::sslCtx) { + SSL_CTX_free(DtlsTransport::sslCtx); + DtlsTransport::sslCtx = nullptr; + } + + MS_THROW_ERROR("SSL context creation failed"); +} + +void DtlsTransport::GenerateFingerprints() { + MS_TRACE(); + + for (auto& kv : DtlsTransport::string2FingerprintAlgorithm) { + const std::string& algorithmString = kv.first; + FingerprintAlgorithm algorithm = kv.second; + uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; + unsigned int size{0}; + char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; + const EVP_MD* hashFunction; + int ret; + + switch (algorithm) { + case FingerprintAlgorithm::SHA1: + hashFunction = EVP_sha1(); + break; + + case FingerprintAlgorithm::SHA224: + hashFunction = EVP_sha224(); + break; + + case FingerprintAlgorithm::SHA256: + hashFunction = EVP_sha256(); + break; + + case FingerprintAlgorithm::SHA384: + hashFunction = EVP_sha384(); + break; + + case FingerprintAlgorithm::SHA512: + hashFunction = EVP_sha512(); + break; + + default: + MS_THROW_ERROR("unknown algorithm"); + } + + ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size); + + if (ret == 0) { + MS_ERROR("X509_digest() failed"); + MS_THROW_ERROR("Fingerprints generation failed"); + } + + // Convert to hexadecimal format in uppercase with colons. + for (unsigned int i{0}; i < size; ++i) { + std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); + } + hexFingerprint[(size * 3) - 1] = '\0'; + + MS_DEBUG_TAG(dtls, "%-7s fingerprint: %s", algorithmString.c_str(), hexFingerprint); + // Store it in the vector. + DtlsTransport::Fingerprint fingerprint; + + fingerprint.algorithm = DtlsTransport::GetFingerprintAlgorithm(algorithmString); + fingerprint.value = hexFingerprint; + + DtlsTransport::localFingerprints.push_back(fingerprint); + } +} + +/* Instance methods. */ + +DtlsTransport::DtlsTransport(Listener* listener) : listener(listener) { + MS_TRACE(); + + /* Set SSL. */ + + this->ssl = SSL_new(DtlsTransport::sslCtx); + + if (!this->ssl) { + LOG_OPENSSL_ERROR("SSL_new() failed"); + + goto error; + } + + // Set this as custom data. + SSL_set_ex_data(this->ssl, 0, static_cast(this)); + + this->sslBioFromNetwork = BIO_new(BIO_s_mem()); + + if (!this->sslBioFromNetwork) { + LOG_OPENSSL_ERROR("BIO_new() failed"); + + SSL_free(this->ssl); + + goto error; + } + + this->sslBioToNetwork = BIO_new(BIO_s_mem()); + + if (!this->sslBioToNetwork) { + LOG_OPENSSL_ERROR("BIO_new() failed"); + + BIO_free(this->sslBioFromNetwork); + SSL_free(this->ssl); + + goto error; + } + + SSL_set_bio(this->ssl, this->sslBioFromNetwork, this->sslBioToNetwork); + + // Set the MTU so that we don't send packets that are too large with no fragmentation. + SSL_set_mtu(this->ssl, DtlsMtu); + DTLS_set_link_mtu(this->ssl, DtlsMtu); + + // Set callback handler for setting DTLS timer interval. + DTLS_set_timer_cb(this->ssl, onSslDtlsTimer); + + // Set the DTLS timer. + // this->timer = new Timer(this); + + return; + +error: + + // NOTE: At this point SSL_set_bio() was not called so we must free BIOs as + // well. + if (this->sslBioFromNetwork) BIO_free(this->sslBioFromNetwork); + + if (this->sslBioToNetwork) BIO_free(this->sslBioToNetwork); + + if (this->ssl) SSL_free(this->ssl); + + // NOTE: If this is not catched by the caller the program will abort, but + // this should never happen. + MS_THROW_ERROR("DtlsTransport instance creation failed"); +} + +DtlsTransport::~DtlsTransport() { + MS_TRACE(); + + if (IsRunning()) { + // Send close alert to the peer. + SSL_shutdown(this->ssl); + SendPendingOutgoingDtlsData(); + } + + if (this->ssl) { + SSL_free(this->ssl); + + this->ssl = nullptr; + this->sslBioFromNetwork = nullptr; + this->sslBioToNetwork = nullptr; + } + + // Close the DTLS timer. + // delete this->timer; +} + +void DtlsTransport::Dump() const { + MS_TRACE(); + + std::string state{"new"}; + std::string role{"none "}; + + switch (this->state) { + case DtlsState::CONNECTING: + state = "connecting"; + break; + case DtlsState::CONNECTED: + state = "connected"; + break; + case DtlsState::FAILED: + state = "failed"; + break; + case DtlsState::CLOSED: + state = "closed"; + break; + default:; + } + + switch (this->localRole) { + case Role::AUTO: + role = "auto"; + break; + case Role::SERVER: + role = "server"; + break; + case Role::CLIENT: + role = "client"; + break; + default:; + } + + MS_DUMP(""); + MS_DUMP(" state : %s", state.c_str()); + MS_DUMP(" role : %s", role.c_str()); + MS_DUMP(" handshake done: : %s", this->handshakeDone ? "yes" : "no"); + MS_DUMP(""); +} + +void DtlsTransport::Run(Role localRole) { + MS_TRACE(); + + MS_ASSERT(localRole == Role::CLIENT || localRole == Role::SERVER, + "local DTLS role must be 'client' or 'server'"); + + Role previousLocalRole = this->localRole; + + if (localRole == previousLocalRole) { + MS_ERROR("same local DTLS role provided, doing nothing"); + + return; + } + + // If the previous local DTLS role was 'client' or 'server' do reset. + if (previousLocalRole == Role::CLIENT || previousLocalRole == Role::SERVER) { + MS_DEBUG_TAG(dtls, "resetting DTLS due to local role change"); + + Reset(); + } + + // Update local role. + this->localRole = localRole; + + // Set state and notify the listener. + this->state = DtlsState::CONNECTING; + this->listener->OnDtlsTransportConnecting(this); + + switch (this->localRole) { + case Role::CLIENT: { + MS_DEBUG_TAG(dtls, "running [role:client]"); + + SSL_set_connect_state(this->ssl); + SSL_do_handshake(this->ssl); + SendPendingOutgoingDtlsData(); + SetTimeout(); + + break; + } + + case Role::SERVER: { + MS_DEBUG_TAG(dtls, "running [role:server]"); + + SSL_set_accept_state(this->ssl); + SSL_do_handshake(this->ssl); + + break; + } + + default: { + MS_ABORT("invalid local DTLS role"); + } + } +} + +bool DtlsTransport::SetRemoteFingerprint(Fingerprint fingerprint) { + MS_TRACE(); + + MS_ASSERT(fingerprint.algorithm != FingerprintAlgorithm::NONE, + "no fingerprint algorithm provided"); + + this->remoteFingerprint = fingerprint; + + // The remote fingerpring may have been set after DTLS handshake was done, + // so we may need to process it now. + if (this->handshakeDone && this->state != DtlsState::CONNECTED) { + MS_DEBUG_TAG(dtls, "handshake already done, processing it right now"); + + return ProcessHandshake(); + } + + return true; +} + +void DtlsTransport::ProcessDtlsData(const uint8_t* data, size_t len) { + MS_TRACE(); + + int written; + int read; + + if (!IsRunning()) { + MS_ERROR("cannot process data while not running"); + + return; + } + + // Write the received DTLS data into the sslBioFromNetwork. + written = + BIO_write(this->sslBioFromNetwork, static_cast(data), static_cast(len)); + + if (written != static_cast(len)) { + MS_WARN_TAG(dtls, "OpenSSL BIO_write() wrote less (%zu bytes) than given data (%zu bytes)", + static_cast(written), len); + } + + // Must call SSL_read() to process received DTLS data. + read = SSL_read(this->ssl, static_cast(DtlsTransport::sslReadBuffer), SslReadBufferSize); + + // Send data if it's ready. + SendPendingOutgoingDtlsData(); + + // Check SSL status and return if it is bad/closed. + if (!CheckStatus(read)) return; + + // Set/update the DTLS timeout. + if (!SetTimeout()) return; + + // Application data received. Notify to the listener. + if (read > 0) { + // It is allowed to receive DTLS data even before validating remote fingerprint. + if (!this->handshakeDone) { + MS_WARN_TAG(dtls, "ignoring application data received while DTLS handshake not done"); + + return; + } + + // Notify the listener. + this->listener->OnDtlsTransportApplicationDataReceived( + this, (uint8_t*)DtlsTransport::sslReadBuffer, static_cast(read)); + } +} + +void DtlsTransport::SendApplicationData(const uint8_t* data, size_t len) { + MS_TRACE(); + + // We cannot send data to the peer if its remote fingerprint is not validated. + if (this->state != DtlsState::CONNECTED) { + MS_WARN_TAG(dtls, "cannot send application data while DTLS is not fully connected"); + + return; + } + + if (len == 0) { + MS_WARN_TAG(dtls, "ignoring 0 length data"); + + return; + } + + int written; + + written = SSL_write(this->ssl, static_cast(data), static_cast(len)); + + if (written < 0) { + LOG_OPENSSL_ERROR("SSL_write() failed"); + + if (!CheckStatus(written)) return; + } else if (written != static_cast(len)) { + MS_WARN_TAG(dtls, "OpenSSL SSL_write() wrote less (%d bytes) than given data (%zu bytes)", + written, len); + } + + // Send data. + SendPendingOutgoingDtlsData(); +} + +void DtlsTransport::Reset() { + MS_TRACE(); + + int ret; + + if (!IsRunning()) return; + + MS_WARN_TAG(dtls, "resetting DTLS transport"); + + // Stop the DTLS timer. + // this->timer->Stop(); + + // We need to reset the SSL instance so we need to "shutdown" it, but we + // don't want to send a Close Alert to the peer, so just don't call + // SendPendingOutgoingDTLSData(). + SSL_shutdown(this->ssl); + + this->localRole = Role::NONE; + this->state = DtlsState::NEW; + this->handshakeDone = false; + this->handshakeDoneNow = false; + + // Reset SSL status. + // NOTE: For this to properly work, SSL_shutdown() must be called before. + // NOTE: This may fail if not enough DTLS handshake data has been received, + // but we don't care so just clear the error queue. + ret = SSL_clear(this->ssl); + + if (ret == 0) ERR_clear_error(); +} + +inline bool DtlsTransport::CheckStatus(int returnCode) { + MS_TRACE(); + + int err; + bool wasHandshakeDone = this->handshakeDone; + + err = SSL_get_error(this->ssl, returnCode); + + switch (err) { + case SSL_ERROR_NONE: + break; + + case SSL_ERROR_SSL: + LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SSL"); + break; + + case SSL_ERROR_WANT_READ: + break; + + case SSL_ERROR_WANT_WRITE: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_WRITE"); + break; + + case SSL_ERROR_WANT_X509_LOOKUP: + MS_DEBUG_TAG(dtls, "SSL status: SSL_ERROR_WANT_X509_LOOKUP"); + break; + + case SSL_ERROR_SYSCALL: + LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SYSCALL"); + break; + + case SSL_ERROR_ZERO_RETURN: + break; + + case SSL_ERROR_WANT_CONNECT: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_CONNECT"); + break; + + case SSL_ERROR_WANT_ACCEPT: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_ACCEPT"); + break; + + default: + MS_WARN_TAG(dtls, "SSL status: unknown error"); + } + + // Check if the handshake (or re-handshake) has been done right now. + if (this->handshakeDoneNow) { + this->handshakeDoneNow = false; + this->handshakeDone = true; + + // Stop the timer. + // this->timer->Stop(); + + // Process the handshake just once (ignore if DTLS renegotiation). + // if (!wasHandshakeDone && this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE) + // return ProcessHandshake(); + if (!wasHandshakeDone) { + return ProcessHandshake(); + } + + return true; + } + // Check if the peer sent close alert or a fatal error happened. + else if (((SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN) != 0) || err == SSL_ERROR_SSL || + err == SSL_ERROR_SYSCALL) { + if (this->state == DtlsState::CONNECTED) { + MS_DEBUG_TAG(dtls, "disconnected"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::CLOSED; + this->listener->OnDtlsTransportClosed(this); + } else { + MS_WARN_TAG(dtls, "connection failed"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + } + + return false; + } else { + return true; + } +} + +inline void DtlsTransport::SendPendingOutgoingDtlsData() { + MS_TRACE(); + + if (BIO_eof(this->sslBioToNetwork)) return; + + int64_t read; + char* data{nullptr}; + + read = BIO_get_mem_data(this->sslBioToNetwork, &data); // NOLINT + + if (read <= 0) return; + + MS_DEBUG_DEV("%ld bytes of DTLS data ready to sent to the peer", read); + + // Notify the listener. + this->listener->OnDtlsTransportSendData(this, reinterpret_cast(data), + static_cast(read)); + + // Clear the BIO buffer. + // NOTE: the (void) avoids the -Wunused-value warning. + (void)BIO_reset(this->sslBioToNetwork); +} + +inline bool DtlsTransport::SetTimeout() { + MS_TRACE(); + + MS_ASSERT(this->state == DtlsState::CONNECTING || this->state == DtlsState::CONNECTED, + "invalid DTLS state"); + + int64_t ret; + uv_timeval_t dtlsTimeout{0, 0}; + uint64_t timeoutMs; + + // NOTE: If ret == 0 then ignore the value in dtlsTimeout. + // NOTE: No DTLSv_1_2_get_timeout() or DTLS_get_timeout() in OpenSSL 1.1.0-dev. + ret = DTLSv1_get_timeout(this->ssl, static_cast(&dtlsTimeout)); // NOLINT + + if (ret == 0) return true; + + timeoutMs = (dtlsTimeout.tv_sec * static_cast(1000)) + (dtlsTimeout.tv_usec / 1000); + + if (timeoutMs == 0) { + return true; + } else if (timeoutMs < 30000) { + MS_DEBUG_DEV("DTLS timer set in %lu ms", timeoutMs); + + // this->timer->Start(timeoutMs); + + return true; + } + // NOTE: Don't start the timer again if the timeout is greater than 30 seconds. + else { + MS_WARN_TAG(dtls, "DTLS timeout too high (%lu ms), resetting DLTS", timeoutMs); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + + return false; + } +} + +inline bool DtlsTransport::ProcessHandshake() { + MS_TRACE(); + + MS_ASSERT(this->handshakeDone, "handshake not done yet"); +// MS_ASSERT(this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, +// "remote fingerprint not set"); + + // Validate the remote fingerprint. + // if (!CheckRemoteFingerprint()) { + // Reset(); + + // // Set state and notify the listener. + // this->state = DtlsState::FAILED; + // this->listener->OnDtlsTransportFailed(this); + + // return false; + // } + + // Get the negotiated SRTP crypto suite. + RTC::CryptoSuite srtpCryptoSuite = GetNegotiatedSrtpCryptoSuite(); + + if (srtpCryptoSuite != RTC::CryptoSuite::NONE) { + // Extract the SRTP keys (will notify the listener with them). + ExtractSrtpKeys(srtpCryptoSuite); + + return true; + } + + // NOTE: We assume that "use_srtp" DTLS extension is required even if + // there is no audio/video. + MS_WARN_2TAGS(dtls, srtp, "SRTP crypto suite not negotiated"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + + return false; +} + +inline bool DtlsTransport::CheckRemoteFingerprint() { + MS_TRACE(); + + MS_ASSERT(this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, + "remote fingerprint not set"); + + X509* certificate; + uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; + unsigned int size{0}; + char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; + const EVP_MD* hashFunction; + int ret; + + certificate = SSL_get_peer_certificate(this->ssl); + + if (!certificate) { + MS_WARN_TAG(dtls, "no certificate was provided by the peer"); + + return false; + } + + switch (this->remoteFingerprint.algorithm) { + case FingerprintAlgorithm::SHA1: + hashFunction = EVP_sha1(); + break; + + case FingerprintAlgorithm::SHA224: + hashFunction = EVP_sha224(); + break; + + case FingerprintAlgorithm::SHA256: + hashFunction = EVP_sha256(); + break; + + case FingerprintAlgorithm::SHA384: + hashFunction = EVP_sha384(); + break; + + case FingerprintAlgorithm::SHA512: + hashFunction = EVP_sha512(); + break; + + default: + MS_ABORT("unknown algorithm"); + } + + // Compare the remote fingerprint with the value given via signaling. + ret = X509_digest(certificate, hashFunction, binaryFingerprint, &size); + + if (ret == 0) { + MS_ERROR("X509_digest() failed"); + + X509_free(certificate); + + return false; + } + + // Convert to hexadecimal format in uppercase with colons. + for (unsigned int i{0}; i < size; ++i) { + std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); + } + hexFingerprint[(size * 3) - 1] = '\0'; + + if (this->remoteFingerprint.value != hexFingerprint) { + MS_WARN_TAG(dtls, + "fingerprint in the remote certificate (%s) does not match the announced one (%s)", + hexFingerprint, this->remoteFingerprint.value.c_str()); + + X509_free(certificate); + + return false; + } + + MS_DEBUG_TAG(dtls, "valid remote fingerprint"); + + // Get the remote certificate in PEM format. + + BIO* bio = BIO_new(BIO_s_mem()); + + // Ensure the underlying BUF_MEM structure is also freed. + // NOTE: Avoid stupid "warning: value computed is not used [-Wunused-value]" since + // BIO_set_close() always returns 1. + (void)BIO_set_close(bio, BIO_CLOSE); + + ret = PEM_write_bio_X509(bio, certificate); + + if (ret != 1) { + LOG_OPENSSL_ERROR("PEM_write_bio_X509() failed"); + + X509_free(certificate); + BIO_free(bio); + + return false; + } + + BUF_MEM* mem; + + BIO_get_mem_ptr(bio, &mem); // NOLINT[cppcoreguidelines-pro-type-cstyle-cast] + + if (!mem || !mem->data || mem->length == 0u) { + LOG_OPENSSL_ERROR("BIO_get_mem_ptr() failed"); + + X509_free(certificate); + BIO_free(bio); + + return false; + } + + this->remoteCert = std::string(mem->data, mem->length); + + X509_free(certificate); + BIO_free(bio); + + return true; +} + +inline void DtlsTransport::ExtractSrtpKeys(RTC::CryptoSuite srtpCryptoSuite) { + MS_TRACE(); + + size_t srtpKeyLength{0}; + size_t srtpSaltLength{0}; + size_t srtpMasterLength{0}; + + switch (srtpCryptoSuite) { + case RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_80: + case RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_32: { + srtpKeyLength = SrtpMasterKeyLength; + srtpSaltLength = SrtpMasterSaltLength; + srtpMasterLength = SrtpMasterLength; + + break; + } + + case RTC::CryptoSuite::AEAD_AES_256_GCM: { + srtpKeyLength = SrtpAesGcm256MasterKeyLength; + srtpSaltLength = SrtpAesGcm256MasterSaltLength; + srtpMasterLength = SrtpAesGcm256MasterLength; + + break; + } + + case RTC::CryptoSuite::AEAD_AES_128_GCM: { + srtpKeyLength = SrtpAesGcm128MasterKeyLength; + srtpSaltLength = SrtpAesGcm128MasterSaltLength; + srtpMasterLength = SrtpAesGcm128MasterLength; + + break; + } + + default: { + MS_ABORT("unknown SRTP crypto suite"); + } + } + + auto* srtpMaterial = new uint8_t[srtpMasterLength * 2]; + uint8_t* srtpLocalKey{nullptr}; + uint8_t* srtpLocalSalt{nullptr}; + uint8_t* srtpRemoteKey{nullptr}; + uint8_t* srtpRemoteSalt{nullptr}; + auto* srtpLocalMasterKey = new uint8_t[srtpMasterLength]; + auto* srtpRemoteMasterKey = new uint8_t[srtpMasterLength]; + int ret; + + ret = SSL_export_keying_material(this->ssl, srtpMaterial, srtpMasterLength * 2, + "EXTRACTOR-dtls_srtp", 19, nullptr, 0, 0); + + MS_ASSERT(ret != 0, "SSL_export_keying_material() failed"); + + switch (this->localRole) { + case Role::SERVER: { + srtpRemoteKey = srtpMaterial; + srtpLocalKey = srtpRemoteKey + srtpKeyLength; + srtpRemoteSalt = srtpLocalKey + srtpKeyLength; + srtpLocalSalt = srtpRemoteSalt + srtpSaltLength; + + break; + } + + case Role::CLIENT: { + srtpLocalKey = srtpMaterial; + srtpRemoteKey = srtpLocalKey + srtpKeyLength; + srtpLocalSalt = srtpRemoteKey + srtpKeyLength; + srtpRemoteSalt = srtpLocalSalt + srtpSaltLength; + + break; + } + + default: { + MS_ABORT("no DTLS role set"); + } + } + + // Create the SRTP local master key. + std::memcpy(srtpLocalMasterKey, srtpLocalKey, srtpKeyLength); + std::memcpy(srtpLocalMasterKey + srtpKeyLength, srtpLocalSalt, srtpSaltLength); + // Create the SRTP remote master key. + std::memcpy(srtpRemoteMasterKey, srtpRemoteKey, srtpKeyLength); + std::memcpy(srtpRemoteMasterKey + srtpKeyLength, srtpRemoteSalt, srtpSaltLength); + + // Set state and notify the listener. + this->state = DtlsState::CONNECTED; + this->listener->OnDtlsTransportConnected(this, srtpCryptoSuite, srtpLocalMasterKey, + srtpMasterLength, srtpRemoteMasterKey, srtpMasterLength, + this->remoteCert); + + delete[] srtpMaterial; + delete[] srtpLocalMasterKey; + delete[] srtpRemoteMasterKey; +} + +inline RTC::CryptoSuite DtlsTransport::GetNegotiatedSrtpCryptoSuite() { + MS_TRACE(); + + RTC::CryptoSuite negotiatedSrtpCryptoSuite = RTC::CryptoSuite::NONE; + + // Ensure that the SRTP crypto suite has been negotiated. + // NOTE: This is a OpenSSL type. + SRTP_PROTECTION_PROFILE* sslSrtpCryptoSuite = SSL_get_selected_srtp_profile(this->ssl); + + if (!sslSrtpCryptoSuite) return negotiatedSrtpCryptoSuite; + + // Get the negotiated SRTP crypto suite. + for (auto& srtpCryptoSuite : DtlsTransport::srtpCryptoSuites) { + SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(srtpCryptoSuite); + + if (std::strcmp(sslSrtpCryptoSuite->name, cryptoSuiteEntry->name) == 0) { + MS_DEBUG_2TAGS(dtls, srtp, "chosen SRTP crypto suite: %s", cryptoSuiteEntry->name); + + negotiatedSrtpCryptoSuite = cryptoSuiteEntry->cryptoSuite; + } + } + + MS_ASSERT(negotiatedSrtpCryptoSuite != RTC::CryptoSuite::NONE, + "chosen SRTP crypto suite is not an available one"); + + return negotiatedSrtpCryptoSuite; +} + +inline void DtlsTransport::OnSslInfo(int where, int ret) { + MS_TRACE(); + + int w = where & -SSL_ST_MASK; + const char* role; + + if ((w & SSL_ST_CONNECT) != 0) + role = "client"; + else if ((w & SSL_ST_ACCEPT) != 0) + role = "server"; + else + role = "undefined"; + + if ((where & SSL_CB_LOOP) != 0) { + MS_DEBUG_TAG(dtls, "[role:%s, action:'%s']", role, SSL_state_string_long(this->ssl)); + } else if ((where & SSL_CB_ALERT) != 0) { + const char* alertType; + + switch (*SSL_alert_type_string(ret)) { + case 'W': + alertType = "warning"; + break; + + case 'F': + alertType = "fatal"; + break; + + default: + alertType = "undefined"; + } + + if ((where & SSL_CB_READ) != 0) { + MS_WARN_TAG(dtls, "received DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } else if ((where & SSL_CB_WRITE) != 0) { + MS_DEBUG_TAG(dtls, "sending DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } else { + MS_DEBUG_TAG(dtls, "DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } + } else if ((where & SSL_CB_EXIT) != 0) { + if (ret == 0) + MS_DEBUG_TAG(dtls, "[role:%s, failed:'%s']", role, SSL_state_string_long(this->ssl)); + else if (ret < 0) + MS_DEBUG_TAG(dtls, "role: %s, waiting:'%s']", role, SSL_state_string_long(this->ssl)); + } else if ((where & SSL_CB_HANDSHAKE_START) != 0) { + MS_DEBUG_TAG(dtls, "DTLS handshake start"); + } else if ((where & SSL_CB_HANDSHAKE_DONE) != 0) { + MS_DEBUG_TAG(dtls, "DTLS handshake done"); + + this->handshakeDoneNow = true; + } + + // NOTE: checking SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN here upon + // receipt of a close alert does not work (the flag is set after this callback). +} + +inline void DtlsTransport::OnTimer() { + MS_TRACE(); + + // Workaround for https://github.com/openssl/openssl/issues/7998. + if (this->handshakeDone) { + MS_DEBUG_DEV("handshake is done so return"); + + return; + } + + DTLSv1_handle_timeout(this->ssl); + + // If required, send DTLS data. + SendPendingOutgoingDtlsData(); + + // Set the DTLS timer again. + SetTimeout(); +} +} // namespace RTC diff --git a/webrtc/rtc_dtls_transport.h b/webrtc/rtc_dtls_transport.h new file mode 100644 index 00000000..628f966b --- /dev/null +++ b/webrtc/rtc_dtls_transport.h @@ -0,0 +1,187 @@ +#ifndef MS_RTC_DTLS_TRANSPORT_HPP +#define MS_RTC_DTLS_TRANSPORT_HPP + +#include +#include +#include + +#include +#include +#include + +namespace RTC { +enum class CryptoSuite { + NONE = 0, + AES_CM_128_HMAC_SHA1_80 = 1, + AES_CM_128_HMAC_SHA1_32, + AEAD_AES_256_GCM, + AEAD_AES_128_GCM +}; +class DtlsTransport { + public: + enum class DtlsState { NEW = 1, CONNECTING, CONNECTED, FAILED, CLOSED }; + + public: + enum class Role { NONE = 0, AUTO = 1, CLIENT, SERVER }; + + public: + enum class FingerprintAlgorithm { NONE = 0, SHA1 = 1, SHA224, SHA256, SHA384, SHA512 }; + + public: + struct Fingerprint { + FingerprintAlgorithm algorithm{FingerprintAlgorithm::NONE}; + std::string value; + }; + + private: + struct SrtpCryptoSuiteMapEntry { + RTC::CryptoSuite cryptoSuite; + const char* name; + }; + + public: + class Listener { + public: + // DTLS is in the process of negotiating a secure connection. Incoming + // media can flow through. + // NOTE: The caller MUST NOT call any method during this callback. + virtual void OnDtlsTransportConnecting(const RTC::DtlsTransport* dtlsTransport) = 0; + // DTLS has completed negotiation of a secure connection (including DTLS-SRTP + // and remote fingerprint verification). Outgoing media can now flow through. + // NOTE: The caller MUST NOT call any method during this callback. + virtual void OnDtlsTransportConnected(const RTC::DtlsTransport* dtlsTransport, + RTC::CryptoSuite srtpCryptoSuite, uint8_t* srtpLocalKey, + size_t srtpLocalKeyLen, uint8_t* srtpRemoteKey, + size_t srtpRemoteKeyLen, std::string& remoteCert) = 0; + // The DTLS connection has been closed as the result of an error (such as a + // DTLS alert or a failure to validate the remote fingerprint). + virtual void OnDtlsTransportFailed(const RTC::DtlsTransport* dtlsTransport) = 0; + // The DTLS connection has been closed due to receipt of a close_notify alert. + virtual void OnDtlsTransportClosed(const RTC::DtlsTransport* dtlsTransport) = 0; + // Need to send DTLS data to the peer. + virtual void OnDtlsTransportSendData(const RTC::DtlsTransport* dtlsTransport, + const uint8_t* data, size_t len) = 0; + // DTLS application data received. + virtual void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport* dtlsTransport, + const uint8_t* data, size_t len) = 0; + }; + + public: + static void ClassInit(); + static void ClassDestroy(); + static Role StringToRole(const std::string& role) { + auto it = DtlsTransport::string2Role.find(role); + + if (it != DtlsTransport::string2Role.end()) + return it->second; + else + return DtlsTransport::Role::NONE; + } + static FingerprintAlgorithm GetFingerprintAlgorithm(const std::string& fingerprint) { + auto it = DtlsTransport::string2FingerprintAlgorithm.find(fingerprint); + + if (it != DtlsTransport::string2FingerprintAlgorithm.end()) + return it->second; + else + return DtlsTransport::FingerprintAlgorithm::NONE; + } + static std::string& GetFingerprintAlgorithmString(FingerprintAlgorithm fingerprint) { + auto it = DtlsTransport::fingerprintAlgorithm2String.find(fingerprint); + + return it->second; + } + static bool IsDtls(const uint8_t* data, size_t len) { + // clang-format off + return ( + // Minimum DTLS record length is 13 bytes. + (len >= 13) && + // DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes + (data[0] > 19 && data[0] < 64) + ); + // clang-format on + } + + private: + static void GenerateCertificateAndPrivateKey(); + static void ReadCertificateAndPrivateKeyFromFiles(); + static void CreateSslCtx(); + static void GenerateFingerprints(); + + private: + static X509* certificate; + static EVP_PKEY* privateKey; + static SSL_CTX* sslCtx; + static uint8_t sslReadBuffer[]; + static std::map string2Role; + static std::map string2FingerprintAlgorithm; + static std::map fingerprintAlgorithm2String; + static std::vector localFingerprints; + static std::vector srtpCryptoSuites; + + public: + explicit DtlsTransport(Listener* listener); + ~DtlsTransport(); + + public: + void Dump() const; + void Run(Role localRole); + std::vector& GetLocalFingerprints() const { + return DtlsTransport::localFingerprints; + } + bool SetRemoteFingerprint(Fingerprint fingerprint); + void ProcessDtlsData(const uint8_t* data, size_t len); + DtlsState GetState() const { return this->state; } + Role GetLocalRole() const { return this->localRole; } + void SendApplicationData(const uint8_t* data, size_t len); + + private: + bool IsRunning() const { + switch (this->state) { + case DtlsState::NEW: + return false; + case DtlsState::CONNECTING: + case DtlsState::CONNECTED: + return true; + case DtlsState::FAILED: + case DtlsState::CLOSED: + return false; + } + + // Make GCC 4.9 happy. + return false; + } + void Reset(); + bool CheckStatus(int returnCode); + void SendPendingOutgoingDtlsData(); + bool SetTimeout(); + bool ProcessHandshake(); + bool CheckRemoteFingerprint(); + void ExtractSrtpKeys(RTC::CryptoSuite srtpCryptoSuite); + RTC::CryptoSuite GetNegotiatedSrtpCryptoSuite(); + + /* Callbacks fired by OpenSSL events. */ + public: + void OnSslInfo(int where, int ret); + + /* Pure virtual methods inherited from Timer::Listener. */ + public: + void OnTimer(); + + private: + // Passed by argument. + Listener* listener{nullptr}; + // Allocated by this. + SSL* ssl{nullptr}; + BIO* sslBioFromNetwork{nullptr}; // The BIO from which ssl reads. + BIO* sslBioToNetwork{nullptr}; // The BIO in which ssl writes. + // Others. + DtlsState state{DtlsState::NEW}; + Role localRole{Role::NONE}; + Fingerprint remoteFingerprint; + bool handshakeDone{false}; + bool handshakeDoneNow{false}; + std::string remoteCert; +}; +} // namespace RTC + +#endif diff --git a/webrtc/srtp_session.cc b/webrtc/srtp_session.cc new file mode 100644 index 00000000..b21a83cd --- /dev/null +++ b/webrtc/srtp_session.cc @@ -0,0 +1,269 @@ +#define MS_CLASS "RTC::SrtpSession" +// #define MS_LOG_DEV_LEVEL 3 + +#include "srtp_session.h" + +#include // std::memset(), std::memcpy() +#include + +#include "logger.h" + +namespace RTC { +/* Static. */ + +static constexpr size_t EncryptBufferSize{65536}; +static uint8_t EncryptBuffer[EncryptBufferSize]; + +/* Class methods. */ + +std::vector DepLibSRTP::errors = { + // From 0 (srtp_err_status_ok) to 24 (srtp_err_status_pfkey_err). + "success (srtp_err_status_ok)", + "unspecified failure (srtp_err_status_fail)", + "unsupported parameter (srtp_err_status_bad_param)", + "couldn't allocate memory (srtp_err_status_alloc_fail)", + "couldn't deallocate memory (srtp_err_status_dealloc_fail)", + "couldn't initialize (srtp_err_status_init_fail)", + "can’t process as much data as requested (srtp_err_status_terminus)", + "authentication failure (srtp_err_status_auth_fail)", + "cipher failure (srtp_err_status_cipher_fail)", + "replay check failed (bad index) (srtp_err_status_replay_fail)", + "replay check failed (index too old) (srtp_err_status_replay_old)", + "algorithm failed test routine (srtp_err_status_algo_fail)", + "unsupported operation (srtp_err_status_no_such_op)", + "no appropriate context found (srtp_err_status_no_ctx)", + "unable to perform desired validation (srtp_err_status_cant_check)", + "can’t use key any more (srtp_err_status_key_expired)", + "error in use of socket (srtp_err_status_socket_err)", + "error in use POSIX signals (srtp_err_status_signal_err)", + "nonce check failed (srtp_err_status_nonce_bad)", + "couldn’t read data (srtp_err_status_read_fail)", + "couldn’t write data (srtp_err_status_write_fail)", + "error parsing data (srtp_err_status_parse_err)", + "error encoding data (srtp_err_status_encode_err)", + "error while using semaphores (srtp_err_status_semaphore_err)", + "error while using pfkey (srtp_err_status_pfkey_err)"}; +// clang-format on + +/* Static methods. */ + +void DepLibSRTP::ClassInit() { + MS_TRACE(); + + MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string()); + + srtp_err_status_t err = srtp_init(); + + if (DepLibSRTP::IsError(err)) + MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err)); +} + +void DepLibSRTP::ClassDestroy() { + MS_TRACE(); + + srtp_shutdown(); +} + +void SrtpSession::ClassInit() { + // Set libsrtp event handler. + srtp_err_status_t err = + srtp_install_event_handler(static_cast(OnSrtpEvent)); + if (DepLibSRTP::IsError(err)) { + MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err)); + std::cout << "srtp_install_event_handler() failed :" << DepLibSRTP::GetErrorString(err); + } +} + +void SrtpSession::OnSrtpEvent(srtp_event_data_t *data) { + MS_TRACE(); + + switch (data->event) { + case event_ssrc_collision: + MS_WARN_TAG(srtp, "SSRC collision occurred"); + break; + + case event_key_soft_limit: + MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon"); + break; + + case event_key_hard_limit: + MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired"); + break; + + case event_packet_index_limit: + MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)"); + break; + } +} + +/* Instance methods. */ + +SrtpSession::SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t *key, size_t keyLen) { + MS_TRACE(); + + srtp_policy_t policy;// NOLINT(cppcoreguidelines-pro-type-member-init) + + // Set all policy fields to 0. + std::memset(&policy, 0, sizeof(srtp_policy_t)); + + switch (cryptoSuite) { + case CryptoSuite::AES_CM_128_HMAC_SHA1_80: { + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp); + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); + + break; + } + + case CryptoSuite::AES_CM_128_HMAC_SHA1_32: { + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); + // NOTE: Must be 80 for RTCP. + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); + + break; + } + + case CryptoSuite::AEAD_AES_256_GCM: { + srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp); + srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp); + + break; + } + + case CryptoSuite::AEAD_AES_128_GCM: { + srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp); + srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp); + + break; + } + + default: { + MS_ABORT("unknown SRTP crypto suite"); + } + } + + MS_ASSERT((int) keyLen == policy.rtp.cipher_key_len, + "given keyLen does not match policy.rtp.cipher_keyLen"); + + switch (type) { + case Type::INBOUND: + policy.ssrc.type = ssrc_any_inbound; + break; + + case Type::OUTBOUND: + policy.ssrc.type = ssrc_any_outbound; + break; + } + + policy.ssrc.value = 0; + policy.key = key; + // Required for sending RTP retransmission without RTX. + policy.allow_repeat_tx = 1; + policy.window_size = 1024; + policy.next = nullptr; + + // Set the SRTP session. + srtp_err_status_t err = srtp_create(&this->session, &policy); + if (DepLibSRTP::IsError(err)) { + is_init = false; + MS_THROW_ERROR("srtp_create() failed: %s", DepLibSRTP::GetErrorString(err)); + } else { + is_init = true; + } +} + +SrtpSession::~SrtpSession() { + MS_TRACE(); + + if (this->session != nullptr) { + srtp_err_status_t err = srtp_dealloc(this->session); + + if (DepLibSRTP::IsError(err)) + MS_ABORT("srtp_dealloc() failed: %s", DepLibSRTP::GetErrorString(err)); + } +} + +bool SrtpSession::EncryptRtp(const uint8_t **data, size_t *len) { + MS_TRACE(); + if (!is_init) { + return false; + } + // Ensure that the resulting SRTP packet fits into the encrypt buffer. + if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) { + MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len); + + return false; + } + std::memcpy(EncryptBuffer, *data, *len); + + srtp_err_status_t err = + srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) { + MS_WARN_TAG(srtp, "srtp_protect() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + // Update the given data pointer. + *data = (const uint8_t *) EncryptBuffer; + + return true; +} + +bool SrtpSession::DecryptSrtp(uint8_t *data, size_t *len) { + MS_TRACE(); + + srtp_err_status_t err = + srtp_unprotect(this->session, static_cast(data), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) { + MS_DEBUG_TAG(srtp, "srtp_unprotect() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + return true; +} + +bool SrtpSession::EncryptRtcp(const uint8_t **data, size_t *len) { + MS_TRACE(); + + // Ensure that the resulting SRTCP packet fits into the encrypt buffer. + if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) { + MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len); + + return false; + } + + std::memcpy(EncryptBuffer, *data, *len); + + srtp_err_status_t err = srtp_protect_rtcp(this->session, static_cast(EncryptBuffer), + reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) { + MS_WARN_TAG(srtp, "srtp_protect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + // Update the given data pointer. + *data = (const uint8_t *) EncryptBuffer; + + return true; +} + +bool SrtpSession::DecryptSrtcp(uint8_t *data, size_t *len) { + MS_TRACE(); + + srtp_err_status_t err = + srtp_unprotect_rtcp(this->session, static_cast(data), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) { + MS_DEBUG_TAG(srtp, "srtp_unprotect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + return true; +} +}// namespace RTC diff --git a/webrtc/srtp_session.h b/webrtc/srtp_session.h new file mode 100644 index 00000000..95da5fbf --- /dev/null +++ b/webrtc/srtp_session.h @@ -0,0 +1,54 @@ +#ifndef MS_RTC_SRTP_SESSION_HPP +#define MS_RTC_SRTP_SESSION_HPP + +#include "rtc_dtls_transport.h" +#include "utils.h" +#include +#include + +namespace RTC { + +class DepLibSRTP { +public: + static void ClassInit(); + static void ClassDestroy(); + static bool IsError(srtp_err_status_t code) { return (code != srtp_err_status_ok); } + static const char *GetErrorString(srtp_err_status_t code) { + // This throws out_of_range if the given index is not in the vector. + return DepLibSRTP::errors.at(code); + } + +private: + static std::vector errors; +}; + +class SrtpSession { +public: +public: + enum class Type { INBOUND = 1, OUTBOUND }; + +public: + static void ClassInit(); + +private: + static void OnSrtpEvent(srtp_event_data_t *data); + +public: + SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t *key, size_t keyLen); + ~SrtpSession(); + +public: + bool EncryptRtp(const uint8_t **data, size_t *len); + bool DecryptSrtp(uint8_t *data, size_t *len); + bool EncryptRtcp(const uint8_t **data, size_t *len); + bool DecryptSrtcp(uint8_t *data, size_t *len); + void RemoveStream(uint32_t ssrc) { srtp_remove_stream(this->session, uint32_t{htonl(ssrc)}); } + +private: + bool is_init = false; + // Allocated by this. + srtp_t session{nullptr}; +}; +}// namespace RTC + +#endif diff --git a/webrtc/stun_packet.cc b/webrtc/stun_packet.cc new file mode 100644 index 00000000..926f863b --- /dev/null +++ b/webrtc/stun_packet.cc @@ -0,0 +1,710 @@ +#define MS_CLASS "RTC::StunPacket" +// #define MS_LOG_DEV + +#include "stun_packet.h" + +#include // std::snprintf() +#include // std::memcmp(), std::memcpy() + +#include "utils.h" + +namespace RTC { + +/* Class variables. */ + +const uint8_t StunPacket::kMagicCookie[] = {0x21, 0x12, 0xA4, 0x42}; + +/* Class methods. */ + +StunPacket* StunPacket::Parse(const uint8_t* data, size_t len) { + if (!StunPacket::IsStun(data, len)) return nullptr; + + /* + The message type field is decomposed further into the following + structure: + + 0 1 + 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ + |M |M |M|M|M|C|M|M|M|C|M|M|M|M| + |11|10|9|8|7|1|6|5|4|0|3|2|1|0| + +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ + + Figure 3: Format of STUN Message Type Field + + Here the bits in the message type field are shown as most significant + (M11) through least significant (M0). M11 through M0 represent a 12- + bit encoding of the method. C1 and C0 represent a 2-bit encoding of + the class. + */ + + // Get type field. + uint16_t msgType = Utils::Byte::Get2Bytes(data, 0); + + // Get length field. + uint16_t msgLength = Utils::Byte::Get2Bytes(data, 2); + + // length field must be total size minus header's 20 bytes, and must be multiple of 4 Bytes. + if ((static_cast(msgLength) != len - 20) || ((msgLength & 0x03) != 0)) { + ELOG_DEBUG( + "length field + 20 does not match total size (or it is not multiple of 4 bytes), " + "packet discarded"); + + return nullptr; + } + + // Get STUN method. + uint16_t msgMethod = (msgType & 0x000f) | ((msgType & 0x00e0) >> 1) | ((msgType & 0x3E00) >> 2); + + // Get STUN class. + uint16_t msgClass = ((data[0] & 0x01) << 1) | ((data[1] & 0x10) >> 4); + + // Create a new StunPacket (data + 8 points to the received TransactionID field). + auto packet = new StunPacket(static_cast(msgClass), static_cast(msgMethod), + data + 8, data, len); + + /* + STUN Attributes + + After the STUN header are zero or more attributes. Each attribute + MUST be TLV encoded, with a 16-bit type, 16-bit length, and value. + Each STUN attribute MUST end on a 32-bit boundary. As mentioned + above, all fields in an attribute are transmitted most significant + bit first. + + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | Type | Length | + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | Value (variable) .... + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + + // Start looking for attributes after STUN header (Byte #20). + size_t pos{20}; + // Flags (positions) for special MESSAGE-INTEGRITY and FINGERPRINT attributes. + bool hasMessageIntegrity{false}; + bool hasFingerprint{false}; + size_t fingerprintAttrPos; // Will point to the beginning of the attribute. + uint32_t fingerprint; // Holds the value of the FINGERPRINT attribute. + + // Ensure there are at least 4 remaining bytes (attribute with 0 length). + while (pos + 4 <= len) { + // Get the attribute type. + auto attrType = static_cast(Utils::Byte::Get2Bytes(data, pos)); + + // Get the attribute length. + uint16_t attrLength = Utils::Byte::Get2Bytes(data, pos + 2); + + // Ensure the attribute length is not greater than the remaining size. + if ((pos + 4 + attrLength) > len) { + ELOG_DEBUG("the attribute length exceeds the remaining size, packet discarded"); + + delete packet; + return nullptr; + } + + // FINGERPRINT must be the last attribute. + if (hasFingerprint) { + ELOG_DEBUG("attribute after FINGERPRINT is not allowed, packet discarded"); + + delete packet; + return nullptr; + } + + // After a MESSAGE-INTEGRITY attribute just FINGERPRINT is allowed. + if (hasMessageIntegrity && attrType != Attribute::FINGERPRINT) { + ELOG_DEBUG( + "attribute after MESSAGE-INTEGRITY other than FINGERPRINT is not allowed, " + "packet discarded"); + + delete packet; + return nullptr; + } + + const uint8_t* attrValuePos = data + pos + 4; + + switch (attrType) { + case Attribute::USERNAME: { + packet->SetUsername(reinterpret_cast(attrValuePos), + static_cast(attrLength)); + + break; + } + + case Attribute::PRIORITY: { + // Ensure attribute length is 4 bytes. + if (attrLength != 4) { + ELOG_DEBUG("attribute PRIORITY must be 4 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetPriority(Utils::Byte::Get4Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::ICE_CONTROLLING: { + // Ensure attribute length is 8 bytes. + if (attrLength != 8) { + ELOG_DEBUG("attribute ICE-CONTROLLING must be 8 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetIceControlling(Utils::Byte::Get8Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::ICE_CONTROLLED: { + // Ensure attribute length is 8 bytes. + if (attrLength != 8) { + ELOG_DEBUG("attribute ICE-CONTROLLED must be 8 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetIceControlled(Utils::Byte::Get8Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::USE_CANDIDATE: { + // Ensure attribute length is 0 bytes. + if (attrLength != 0) { + ELOG_DEBUG("attribute USE-CANDIDATE must be 0 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetUseCandidate(); + + break; + } + + case Attribute::MESSAGE_INTEGRITY: { + // Ensure attribute length is 20 bytes. + if (attrLength != 20) { + ELOG_DEBUG("attribute MESSAGE-INTEGRITY must be 20 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + hasMessageIntegrity = true; + packet->SetMessageIntegrity(attrValuePos); + + break; + } + + case Attribute::FINGERPRINT: { + // Ensure attribute length is 4 bytes. + if (attrLength != 4) { + ELOG_DEBUG("attribute FINGERPRINT must be 4 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + hasFingerprint = true; + fingerprintAttrPos = pos; + fingerprint = Utils::Byte::Get4Bytes(attrValuePos, 0); + packet->SetFingerprint(); + + break; + } + + case Attribute::ERROR_CODE: { + // Ensure attribute length >= 4bytes. + if (attrLength < 4) { + ELOG_DEBUG("attribute ERROR-CODE must be >= 4bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + uint8_t errorClass = Utils::Byte::Get1Byte(attrValuePos, 2); + uint8_t errorNumber = Utils::Byte::Get1Byte(attrValuePos, 3); + auto errorCode = static_cast(errorClass * 100 + errorNumber); + + packet->SetErrorCode(errorCode); + + break; + } + + default:; + } + + // Set next attribute position. + pos = static_cast(Utils::Byte::PadTo4Bytes(static_cast(pos + 4 + attrLength))); + } + + // Ensure current position matches the total length. + if (pos != len) { + ELOG_DEBUG("computed packet size does not match total size, packet discarded"); + + delete packet; + return nullptr; + } + + // If it has FINGERPRINT attribute then verify it. + if (hasFingerprint) { + // Compute the CRC32 of the received packet up to (but excluding) the + // FINGERPRINT attribute and XOR it with 0x5354554e. + uint32_t computedFingerprint = Utils::Crypto::GetCRC32(data, fingerprintAttrPos) ^ 0x5354554e; + + // Compare with the FINGERPRINT value in the packet. + if (fingerprint != computedFingerprint) { + ELOG_DEBUG( + "computed FINGERPRINT value does not match the value in the packet, " + "packet discarded"); + + delete packet; + return nullptr; + } + } + + return packet; +} + +/* Instance methods. */ + +StunPacket::StunPacket(Class klass, Method method, const uint8_t* transactionId, + const uint8_t* data, size_t size) + : klass(klass), + method(method), + transactionId(transactionId), + data(const_cast(data)), + size(size) { + // MS_TRACE(); +} + +StunPacket::~StunPacket() { + // MS_TRACE(); +} + +void StunPacket::Dump() const { + // MS_TRACE(); + + // MS_DUMP(""); + + std::string klass; + switch (this->klass) { + case Class::REQUEST: + klass = "Request"; + break; + case Class::INDICATION: + klass = "Indication"; + break; + case Class::SUCCESS_RESPONSE: + klass = "SuccessResponse"; + break; + case Class::ERROR_RESPONSE: + klass = "ErrorResponse"; + break; + } + if (this->method == Method::BINDING) { + // MS_DUMP(" Binding %s", klass.c_str()); + } else { + // This prints the unknown method number. Example: TURN Allocate => 0x003. + // MS_DUMP(" %s with unknown method %#.3x", klass.c_str(), + // static_cast(this->method)); + } + // MS_DUMP(" size: %zu bytes", this->size); + + static char transactionId[25]; + + for (int i{0}; i < 12; ++i) { + // NOTE: n must be 3 because snprintf adds a \0 after printed chars. + std::snprintf(transactionId + (i * 2), 3, "%.2x", this->transactionId[i]); + } + // MS_DUMP(" transactionId: %s", transactionId); + if (this->errorCode != 0u) + // MS_DUMP(" errorCode: %" PRIu16, this->errorCode); + if (!this->username.empty()) + // MS_DUMP(" username: %s", this->username.c_str()); + if (this->priority != 0u) + // MS_DUMP(" priority: %" PRIu32, this->priority); + if (this->iceControlling != 0u) + // MS_DUMP(" iceControlling: %" PRIu64, this->iceControlling); + if (this->iceControlled != 0u) + // MS_DUMP(" iceControlled: %" PRIu64, this->iceControlled); + if (this->hasUseCandidate) + // MS_DUMP(" useCandidate"); + if (this->xorMappedAddress != nullptr) { + int family; + uint16_t port; + std::string ip; + + Utils::IP::GetAddressInfo(this->xorMappedAddress, family, ip, port); + + // MS_DUMP(" xorMappedAddress: %s : %" PRIu16, ip.c_str(), port); + } + if (this->messageIntegrity != nullptr) { + static char messageIntegrity[41]; + + for (int i{0}; i < 20; ++i) { + std::snprintf(messageIntegrity + (i * 2), 3, "%.2x", this->messageIntegrity[i]); + } + + // MS_DUMP(" messageIntegrity: %s", messageIntegrity); + } + if (this->hasFingerprint) { + } + // MS_DUMP(" has fingerprint"); + + // MS_DUMP(""); +} + +StunPacket::Authentication StunPacket::CheckAuthentication(const std::string& localUsername, + const std::string& localPassword) { + // MS_TRACE(); + + switch (this->klass) { + case Class::REQUEST: + case Class::INDICATION: { + // Both USERNAME and MESSAGE-INTEGRITY must be present. + if (this->messageIntegrity == nullptr || this->username.empty()) + return Authentication::BAD_REQUEST; + + // Check that USERNAME attribute begins with our local username plus ":". + size_t localUsernameLen = localUsername.length(); + + if (this->username.length() <= localUsernameLen || + this->username.at(localUsernameLen) != ':' || + (this->username.compare(0, localUsernameLen, localUsername) != 0)) { + return Authentication::UNAUTHORIZED; + } + + break; + } + // This method cannot check authentication in received responses (as we + // are ICE-Lite and don't generate requests). + case Class::SUCCESS_RESPONSE: + case Class::ERROR_RESPONSE: { + // MS_ERROR("cannot check authentication for a STUN response"); + + return Authentication::BAD_REQUEST; + } + } + + // If there is FINGERPRINT it must be discarded for MESSAGE-INTEGRITY calculation, + // so the header length field must be modified (and later restored). + if (this->hasFingerprint) + // Set the header length field: full size - header length (20) - FINGERPRINT length (8). + Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20 - 8)); + + // Calculate the HMAC-SHA1 of the message according to MESSAGE-INTEGRITY rules. + const uint8_t* computedMessageIntegrity = Utils::Crypto::GetHmacShA1( + localPassword, this->data, (this->messageIntegrity - 4) - this->data); + + Authentication result; + + // Compare the computed HMAC-SHA1 with the MESSAGE-INTEGRITY in the packet. + if (std::memcmp(this->messageIntegrity, computedMessageIntegrity, 20) == 0) + result = Authentication::OK; + else + result = Authentication::UNAUTHORIZED; + + // Restore the header length field. + if (this->hasFingerprint) + Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20)); + + return result; +} + +StunPacket* StunPacket::CreateSuccessResponse() { + // MS_TRACE(); + + // MS_ASSERT( + // this->klass == Class::REQUEST, + // "attempt to create a success response for a non Request STUN packet"); + + return new StunPacket(Class::SUCCESS_RESPONSE, this->method, this->transactionId, nullptr, 0); +} + +StunPacket* StunPacket::CreateErrorResponse(uint16_t errorCode) { + // MS_TRACE(); + + // MS_ASSERT( + // this->klass == Class::REQUEST, + // "attempt to create an error response for a non Request STUN packet"); + + auto response = + new StunPacket(Class::ERROR_RESPONSE, this->method, this->transactionId, nullptr, 0); + + response->SetErrorCode(errorCode); + + return response; +} + +void StunPacket::Authenticate(const std::string& password) { + // Just for Request, Indication and SuccessResponse messages. + if (this->klass == Class::ERROR_RESPONSE) { + // MS_ERROR("cannot set password for ErrorResponse messages"); + + return; + } + + this->password = password; +} + +void StunPacket::Serialize(uint8_t* buffer) { + // MS_TRACE(); + + // Some useful variables. + uint16_t usernamePaddedLen{0}; + uint16_t xorMappedAddressPaddedLen{0}; + bool addXorMappedAddress = + ((this->xorMappedAddress != nullptr) && this->method == StunPacket::Method::BINDING && + this->klass == Class::SUCCESS_RESPONSE); + bool addErrorCode = ((this->errorCode != 0u) && this->klass == Class::ERROR_RESPONSE); + bool addMessageIntegrity = (this->klass != Class::ERROR_RESPONSE && !this->password.empty()); + bool addFingerprint{true}; // Do always. + + // Update data pointer. + this->data = buffer; + + // First calculate the total required size for the entire packet. + this->size = 20; // Header. + + if (!this->username.empty()) { + usernamePaddedLen = Utils::Byte::PadTo4Bytes(static_cast(this->username.length())); + this->size += 4 + usernamePaddedLen; + } + + if (this->priority != 0u) this->size += 4 + 4; + + if (this->iceControlling != 0u) this->size += 4 + 8; + + if (this->iceControlled != 0u) this->size += 4 + 8; + + if (this->hasUseCandidate) this->size += 4; + + if (addXorMappedAddress) { + switch (this->xorMappedAddress->sa_family) { + case AF_INET: { + xorMappedAddressPaddedLen = 8; + this->size += 4 + 8; + + break; + } + + case AF_INET6: { + xorMappedAddressPaddedLen = 20; + this->size += 4 + 20; + + break; + } + + default: { + // MS_ERROR("invalid inet family in XOR-MAPPED-ADDRESS attribute"); + + addXorMappedAddress = false; + } + } + } + + if (addErrorCode) this->size += 4 + 4; + + if (addMessageIntegrity) this->size += 4 + 20; + + if (addFingerprint) this->size += 4 + 4; + + // Merge class and method fields into type. + uint16_t typeField = (static_cast(this->method) & 0x0f80) << 2; + + typeField |= (static_cast(this->method) & 0x0070) << 1; + typeField |= (static_cast(this->method) & 0x000f); + typeField |= (static_cast(this->klass) & 0x02) << 7; + typeField |= (static_cast(this->klass) & 0x01) << 4; + + // Set type field. + Utils::Byte::Set2Bytes(buffer, 0, typeField); + // Set length field. + Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size) - 20); + // Set magic cookie. + std::memcpy(buffer + 4, StunPacket::kMagicCookie, 4); + // Set TransactionId field. + std::memcpy(buffer + 8, this->transactionId, 12); + // Update the transaction ID pointer. + this->transactionId = buffer + 8; + // Add atributes. + size_t pos{20}; + + // Add USERNAME. + if (usernamePaddedLen != 0u) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USERNAME)); + Utils::Byte::Set2Bytes(buffer, pos + 2, static_cast(this->username.length())); + std::memcpy(buffer + pos + 4, this->username.c_str(), this->username.length()); + pos += 4 + usernamePaddedLen; + } + + // Add PRIORITY. + if (this->priority != 0u) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::PRIORITY)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + Utils::Byte::Set4Bytes(buffer, pos + 4, this->priority); + pos += 4 + 4; + } + + // Add ICE-CONTROLLING. + if (this->iceControlling != 0u) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLING)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 8); + Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlling); + pos += 4 + 8; + } + + // Add ICE-CONTROLLED. + if (this->iceControlled != 0u) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLED)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 8); + Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlled); + pos += 4 + 8; + } + + // Add USE-CANDIDATE. + if (this->hasUseCandidate) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USE_CANDIDATE)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 0); + pos += 4; + } + + // Add XOR-MAPPED-ADDRESS + if (addXorMappedAddress) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::XOR_MAPPED_ADDRESS)); + Utils::Byte::Set2Bytes(buffer, pos + 2, xorMappedAddressPaddedLen); + + uint8_t* attrValue = buffer + pos + 4; + + switch (this->xorMappedAddress->sa_family) { + case AF_INET: { + // Set first byte to 0. + attrValue[0] = 0; + // Set inet family. + attrValue[1] = 0x01; + // Set port and XOR it. + std::memcpy(attrValue + 2, + &(reinterpret_cast(this->xorMappedAddress))->sin_port, 2); + attrValue[2] ^= StunPacket::kMagicCookie[0]; + attrValue[3] ^= StunPacket::kMagicCookie[1]; + // Set address and XOR it. + std::memcpy( + attrValue + 4, + &(reinterpret_cast(this->xorMappedAddress))->sin_addr.s_addr, 4); + attrValue[4] ^= StunPacket::kMagicCookie[0]; + attrValue[5] ^= StunPacket::kMagicCookie[1]; + attrValue[6] ^= StunPacket::kMagicCookie[2]; + attrValue[7] ^= StunPacket::kMagicCookie[3]; + + pos += 4 + 8; + + break; + } + + case AF_INET6: { + // Set first byte to 0. + attrValue[0] = 0; + // Set inet family. + attrValue[1] = 0x02; + // Set port and XOR it. + std::memcpy(attrValue + 2, + &(reinterpret_cast(this->xorMappedAddress))->sin6_port, 2); + attrValue[2] ^= StunPacket::kMagicCookie[0]; + attrValue[3] ^= StunPacket::kMagicCookie[1]; + // Set address and XOR it. + std::memcpy( + attrValue + 4, + &(reinterpret_cast(this->xorMappedAddress))->sin6_addr.s6_addr, + 16); + attrValue[4] ^= StunPacket::kMagicCookie[0]; + attrValue[5] ^= StunPacket::kMagicCookie[1]; + attrValue[6] ^= StunPacket::kMagicCookie[2]; + attrValue[7] ^= StunPacket::kMagicCookie[3]; + attrValue[8] ^= this->transactionId[0]; + attrValue[9] ^= this->transactionId[1]; + attrValue[10] ^= this->transactionId[2]; + attrValue[11] ^= this->transactionId[3]; + attrValue[12] ^= this->transactionId[4]; + attrValue[13] ^= this->transactionId[5]; + attrValue[14] ^= this->transactionId[6]; + attrValue[15] ^= this->transactionId[7]; + attrValue[16] ^= this->transactionId[8]; + attrValue[17] ^= this->transactionId[9]; + attrValue[18] ^= this->transactionId[10]; + attrValue[19] ^= this->transactionId[11]; + + pos += 4 + 20; + + break; + } + } + } + + // Add ERROR-CODE. + if (addErrorCode) { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ERROR_CODE)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + + auto codeClass = static_cast(this->errorCode / 100); + uint8_t codeNumber = static_cast(this->errorCode) - (codeClass * 100); + + Utils::Byte::Set2Bytes(buffer, pos + 4, 0); + Utils::Byte::Set1Byte(buffer, pos + 6, codeClass); + Utils::Byte::Set1Byte(buffer, pos + 7, codeNumber); + pos += 4 + 4; + } + + // Add MESSAGE-INTEGRITY. + if (addMessageIntegrity) { + // Ignore FINGERPRINT. + if (addFingerprint) + Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20 - 8)); + + // Calculate the HMAC-SHA1 of the packet according to MESSAGE-INTEGRITY rules. + const uint8_t* computedMessageIntegrity = + Utils::Crypto::GetHmacShA1(this->password, buffer, pos); + + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::MESSAGE_INTEGRITY)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 20); + std::memcpy(buffer + pos + 4, computedMessageIntegrity, 20); + + // Update the pointer. + this->messageIntegrity = buffer + pos + 4; + pos += 4 + 20; + + // Restore length field. + if (addFingerprint) Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20)); + } else { + // Unset the pointer (if it was set). + this->messageIntegrity = nullptr; + } + + // Add FINGERPRINT. + if (addFingerprint) { + // Compute the CRC32 of the packet up to (but excluding) the FINGERPRINT + // attribute and XOR it with 0x5354554e. + uint32_t computedFingerprint = Utils::Crypto::GetCRC32(buffer, pos) ^ 0x5354554e; + + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::FINGERPRINT)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + Utils::Byte::Set4Bytes(buffer, pos + 4, computedFingerprint); + pos += 4 + 4; + + // Set flag. + this->hasFingerprint = true; + } else { + this->hasFingerprint = false; + } + + // MS_ASSERT(pos == this->size, "pos != this->size"); +} +} // namespace RTC diff --git a/webrtc/stun_packet.h b/webrtc/stun_packet.h new file mode 100644 index 00000000..861d11cd --- /dev/null +++ b/webrtc/stun_packet.h @@ -0,0 +1,179 @@ +#ifndef MS_RTC_STUN_PACKET_HPP +#define MS_RTC_STUN_PACKET_HPP + + +#include "logger.h" +#include "utils.h" +#include + +namespace RTC { +class StunPacket { +public: + // STUN message class. + enum class Class : uint16_t { + REQUEST = 0, + INDICATION = 1, + SUCCESS_RESPONSE = 2, + ERROR_RESPONSE = 3 + }; + + // STUN message method. + enum class Method : uint16_t { BINDING = 1 }; + + // Attribute type. + enum class Attribute : uint16_t { + MAPPED_ADDRESS = 0x0001, + USERNAME = 0x0006, + MESSAGE_INTEGRITY = 0x0008, + ERROR_CODE = 0x0009, + UNKNOWN_ATTRIBUTES = 0x000A, + REALM = 0x0014, + NONCE = 0x0015, + XOR_MAPPED_ADDRESS = 0x0020, + PRIORITY = 0x0024, + USE_CANDIDATE = 0x0025, + SOFTWARE = 0x8022, + ALTERNATE_SERVER = 0x8023, + FINGERPRINT = 0x8028, + ICE_CONTROLLED = 0x8029, + ICE_CONTROLLING = 0x802A + }; + + // Authentication result. + enum class Authentication { OK = 0, UNAUTHORIZED = 1, BAD_REQUEST = 2 }; + +public: + static bool IsStun(const uint8_t *data, size_t len); + static StunPacket *Parse(const uint8_t *data, size_t len); + +private: + static const uint8_t kMagicCookie[]; + +public: + StunPacket(Class klass, Method method, const uint8_t *transactionId, const uint8_t *data, + size_t size); + ~StunPacket(); + + void Dump() const; + Class GetClass() const; + Method GetMethod() const; + const uint8_t *GetData() const; + size_t GetSize() const; + void SetUsername(const char *username, size_t len); + void SetPriority(uint32_t priority); + void SetIceControlling(uint64_t iceControlling); + void SetIceControlled(uint64_t iceControlled); + void SetUseCandidate(); + void SetXorMappedAddress(const struct sockaddr *xorMappedAddress); + void SetErrorCode(uint16_t errorCode); + void SetMessageIntegrity(const uint8_t *messageIntegrity); + void SetFingerprint(); + const std::string &GetUsername() const; + uint32_t GetPriority() const; + uint64_t GetIceControlling() const; + uint64_t GetIceControlled() const; + bool HasUseCandidate() const; + uint16_t GetErrorCode() const; + bool HasMessageIntegrity() const; + bool HasFingerprint() const; + Authentication CheckAuthentication(const std::string &localUsername, + const std::string &localPassword); + StunPacket *CreateSuccessResponse(); + StunPacket *CreateErrorResponse(uint16_t errorCode); + void Authenticate(const std::string &password); + void Serialize(uint8_t *buffer); + +private: + // Passed by argument. + Class klass; // 2 bytes. + Method method; // 2 bytes. + const uint8_t *transactionId{nullptr};// 12 bytes. + uint8_t *data{nullptr}; // Pointer to binary data. + size_t size{0}; // The full message size (including header). + // STUN attributes. + std::string username; // Less than 513 bytes. + uint32_t priority{0}; // 4 bytes unsigned integer. + uint64_t iceControlling{0}; // 8 bytes unsigned integer. + uint64_t iceControlled{0}; // 8 bytes unsigned integer. + bool hasUseCandidate{false}; // 0 bytes. + const uint8_t *messageIntegrity{nullptr}; // 20 bytes. + bool hasFingerprint{false}; // 4 bytes. + const struct sockaddr *xorMappedAddress{nullptr};// 8 or 20 bytes. + uint16_t errorCode{0}; // 4 bytes (no reason phrase). + std::string password; +}; + +/* Inline class methods. */ + +inline bool StunPacket::IsStun(const uint8_t *data, size_t len) { + // clang-format off + return ( + // STUN headers are 20 bytes. + (len >= 20) && + // DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes + (data[0] < 3) && + // Magic cookie must match. + (data[4] == StunPacket::kMagicCookie[0]) && (data[5] == StunPacket::kMagicCookie[1]) && + (data[6] == StunPacket::kMagicCookie[2]) && (data[7] == StunPacket::kMagicCookie[3]) + ); + // clang-format on +} + +/* Inline instance methods. */ + +inline StunPacket::Class StunPacket::GetClass() const { return this->klass; } + +inline StunPacket::Method StunPacket::GetMethod() const { return this->method; } + +inline const uint8_t *StunPacket::GetData() const { return this->data; } + +inline size_t StunPacket::GetSize() const { return this->size; } + +inline void StunPacket::SetUsername(const char *username, size_t len) { + this->username.assign(username, len); +} + +inline void StunPacket::SetPriority(const uint32_t priority) { this->priority = priority; } + +inline void StunPacket::SetIceControlling(const uint64_t iceControlling) { + this->iceControlling = iceControlling; +} + +inline void StunPacket::SetIceControlled(const uint64_t iceControlled) { + this->iceControlled = iceControlled; +} + +inline void StunPacket::SetUseCandidate() { this->hasUseCandidate = true; } + +inline void StunPacket::SetXorMappedAddress(const struct sockaddr *xorMappedAddress) { + this->xorMappedAddress = xorMappedAddress; +} + +inline void StunPacket::SetErrorCode(uint16_t errorCode) { this->errorCode = errorCode; } + +inline void StunPacket::SetMessageIntegrity(const uint8_t *messageIntegrity) { + this->messageIntegrity = messageIntegrity; +} + +inline void StunPacket::SetFingerprint() { this->hasFingerprint = true; } + +inline const std::string &StunPacket::GetUsername() const { return this->username; } + +inline uint32_t StunPacket::GetPriority() const { return this->priority; } + +inline uint64_t StunPacket::GetIceControlling() const { return this->iceControlling; } + +inline uint64_t StunPacket::GetIceControlled() const { return this->iceControlled; } + +inline bool StunPacket::HasUseCandidate() const { return this->hasUseCandidate; } + +inline uint16_t StunPacket::GetErrorCode() const { return this->errorCode; } + +inline bool StunPacket::HasMessageIntegrity() const { + return (this->messageIntegrity ? true : false); +} + +inline bool StunPacket::HasFingerprint() const { return this->hasFingerprint; } +}// namespace RTC + +#endif diff --git a/webrtc/utils.cc b/webrtc/utils.cc new file mode 100644 index 00000000..ab9f1fda --- /dev/null +++ b/webrtc/utils.cc @@ -0,0 +1,139 @@ +#define MS_CLASS "Utils::Crypto" +// #define MS_LOG_DEV + +#include "utils.h" + +#include "openssl/sha.h" + +namespace Utils { +/* Static variables. */ + +uint32_t Crypto::seed; +HMAC_CTX *Crypto::hmacSha1Ctx{nullptr}; +uint8_t Crypto::hmacSha1Buffer[20];// SHA-1 result is 20 bytes long. +// clang-format off +const uint32_t Crypto::crc32Table[] = +{ + 0x00000000, 0x77073096, 0xee0e612c, 0x990951ba, 0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3, + 0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988, 0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91, + 0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de, 0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7, + 0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec, 0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5, + 0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172, 0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b, + 0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940, 0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59, + 0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116, 0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f, + 0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924, 0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d, + 0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, 0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433, + 0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818, 0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01, + 0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e, 0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457, + 0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c, 0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65, + 0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2, 0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb, + 0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0, 0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9, + 0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086, 0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f, + 0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4, 0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad, + 0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a, 0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683, + 0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8, 0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1, + 0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe, 0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7, + 0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc, 0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5, + 0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252, 0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b, + 0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60, 0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79, + 0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236, 0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f, + 0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04, 0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d, + 0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a, 0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713, + 0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38, 0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21, + 0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e, 0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777, + 0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c, 0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45, + 0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2, 0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db, + 0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0, 0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9, + 0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6, 0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf, + 0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94, 0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d +}; +// clang-format on + +/* Static methods. */ + +void Crypto::ClassInit() { + // MS_TRACE(); + + // Init the vrypto seed with a random number taken from the address + // of the seed variable itself (which is random). + Crypto::seed = static_cast(reinterpret_cast(std::addressof(Crypto::seed))); + + // Create an OpenSSL HMAC_CTX context for HMAC SHA1 calculation. + // Crypto::hmacSha1Ctx = HMAC_CTX_new(); + if (Crypto::hmacSha1Ctx == nullptr) { + Crypto::hmacSha1Ctx = HMAC_CTX_new(); + } +} + +void Crypto::ClassDestroy() { + // MS_TRACE(); + + if (Crypto::hmacSha1Ctx != nullptr) { + HMAC_CTX_free(Crypto::hmacSha1Ctx); + } +} + +const uint8_t *Crypto::GetHmacShA1(const std::string &key, const uint8_t *data, size_t len) { + // MS_TRACE(); + + size_t ret; + + ret = HMAC_Init_ex(Crypto::hmacSha1Ctx, key.c_str(), key.length(), EVP_sha1(), nullptr); + + // MS_ASSERT(ret == 1, "OpenSSL HMAC_Init_ex() failed with key '%s'", key.c_str()); + + ret = HMAC_Update(Crypto::hmacSha1Ctx, data, static_cast(len)); + /* + MS_ASSERT( + ret == 1, + "OpenSSL HMAC_Update() failed with key '%s' and data length %zu bytes", + key.c_str(), + len); + */ + uint32_t resultLen; + + ret = HMAC_Final(Crypto::hmacSha1Ctx, (uint8_t *) Crypto::hmacSha1Buffer, &resultLen); + + /* + MS_ASSERT( + ret == 1, "OpenSSL HMAC_Final() failed with key '%s' and data length %zu bytes", key.c_str(), + len); MS_ASSERT(resultLen == 20, "OpenSSL HMAC_Final() resultLen is %u instead of 20", resultLen); + */ + return Crypto::hmacSha1Buffer; +} +}// namespace Utils + +namespace Utils { + +static std::string inet_ntoa(struct in_addr in) { + char buf[20]; + unsigned char *p = (unsigned char *) &(in); + snprintf(buf, sizeof(buf), "%u.%u.%u.%u", p[0], p[1], p[2], p[3]); + return buf; +} + +void IP::GetAddressInfo(const struct sockaddr *addr, int &family, std::string &ip, uint16_t &port) { + char ipBuffer[INET6_ADDRSTRLEN + 1]; + + switch (addr->sa_family) { + case AF_INET: { + ip = Utils::inet_ntoa(reinterpret_cast(addr)->sin_addr); + port = static_cast(ntohs(reinterpret_cast(addr)->sin_port)); + break; + } + + case AF_INET6: { + port = static_cast(ntohs(reinterpret_cast(addr)->sin6_port)); + break; + } + + default: { + // MS_ABORT("unknown network family: %d", static_cast(addr->sa_family)); + } + } + + family = addr->sa_family; + ip.assign(ipBuffer); +} + +}// namespace Utils \ No newline at end of file diff --git a/webrtc/utils.h b/webrtc/utils.h new file mode 100644 index 00000000..1cd6ba9d --- /dev/null +++ b/webrtc/utils.h @@ -0,0 +1,318 @@ +#ifndef MS_UTILS_HPP +#define MS_UTILS_HPP + +#if defined(_WIN32) +#include +#include +#include +#pragma comment (lib, "Ws2_32.lib") +#pragma comment(lib,"Iphlpapi.lib") +#else +#include +#include +#include +#include +#include +#include +#include +#endif // defined(_WIN32) + +#include // std::transform(), std::find(), std::min(), std::max() +#include // PRIu64, etc +#include +#include // size_t +#include // uint8_t, etc +#include // std::memcmp(), std::memcpy() +#include +#include +#include +#include +#include + +namespace Utils { +class IP { +public: + static int GetFamily(const char *ip, size_t ipLen); + static int GetFamily(const std::string &ip); + static void GetAddressInfo(const struct sockaddr *addr, int &family, std::string &ip, + uint16_t &port); + static bool CompareAddresses(const struct sockaddr *addr1, const struct sockaddr *addr2); + static struct sockaddr_storage CopyAddress(const struct sockaddr *addr); + static void NormalizeIp(std::string &ip); +}; + +/* Inline static methods. */ + +inline int IP::GetFamily(const std::string &ip) { return GetFamily(ip.c_str(), ip.size()); } + +inline bool IP::CompareAddresses(const struct sockaddr *addr1, const struct sockaddr *addr2) { + // Compare family. + if (addr1->sa_family != addr2->sa_family || + (addr1->sa_family != AF_INET && addr1->sa_family != AF_INET6)) { + return false; + } + + // Compare port. + if (reinterpret_cast(addr1)->sin_port != + reinterpret_cast(addr2)->sin_port) { + return false; + } + + // Compare IP. + switch (addr1->sa_family) { + case AF_INET: { + return (reinterpret_cast(addr1)->sin_addr.s_addr == + reinterpret_cast(addr2)->sin_addr.s_addr); + } + + case AF_INET6: { + return (std::memcmp( + std::addressof(reinterpret_cast(addr1)->sin6_addr), + std::addressof(reinterpret_cast(addr2)->sin6_addr), + 16) == 0 + ? true + : false); + } + + default: { + return false; + } + } +} + +inline struct sockaddr_storage IP::CopyAddress(const struct sockaddr *addr) { + struct sockaddr_storage copiedAddr; + + switch (addr->sa_family) { + case AF_INET: + std::memcpy(std::addressof(copiedAddr), addr, sizeof(struct sockaddr_in)); + break; + + case AF_INET6: + std::memcpy(std::addressof(copiedAddr), addr, sizeof(struct sockaddr_in6)); + break; + } + + return copiedAddr; +} + +class File { +public: + static void CheckFile(const char *file); +}; + +class Byte { +public: + /** + * Getters below get value in Host Byte Order. + * Setters below set value in Network Byte Order. + */ + static uint8_t Get1Byte(const uint8_t *data, size_t i); + static uint16_t Get2Bytes(const uint8_t *data, size_t i); + static uint32_t Get3Bytes(const uint8_t *data, size_t i); + static uint32_t Get4Bytes(const uint8_t *data, size_t i); + static uint64_t Get8Bytes(const uint8_t *data, size_t i); + static void Set1Byte(uint8_t *data, size_t i, uint8_t value); + static void Set2Bytes(uint8_t *data, size_t i, uint16_t value); + static void Set3Bytes(uint8_t *data, size_t i, uint32_t value); + static void Set4Bytes(uint8_t *data, size_t i, uint32_t value); + static void Set8Bytes(uint8_t *data, size_t i, uint64_t value); + static uint16_t PadTo4Bytes(uint16_t size); + static uint32_t PadTo4Bytes(uint32_t size); +}; + +/* Inline static methods. */ + +inline uint8_t Byte::Get1Byte(const uint8_t *data, size_t i) { return data[i]; } + +inline uint16_t Byte::Get2Bytes(const uint8_t *data, size_t i) { + return uint16_t{data[i + 1]} | uint16_t{data[i]} << 8; +} + +inline uint32_t Byte::Get3Bytes(const uint8_t *data, size_t i) { + return uint32_t{data[i + 2]} | uint32_t{data[i + 1]} << 8 | uint32_t{data[i]} << 16; +} + +inline uint32_t Byte::Get4Bytes(const uint8_t *data, size_t i) { + return uint32_t{data[i + 3]} | uint32_t{data[i + 2]} << 8 | uint32_t{data[i + 1]} << 16 | + uint32_t{data[i]} << 24; +} + +inline uint64_t Byte::Get8Bytes(const uint8_t *data, size_t i) { + return uint64_t{Byte::Get4Bytes(data, i)} << 32 | Byte::Get4Bytes(data, i + 4); +} + +inline void Byte::Set1Byte(uint8_t *data, size_t i, uint8_t value) { data[i] = value; } + +inline void Byte::Set2Bytes(uint8_t *data, size_t i, uint16_t value) { + data[i + 1] = static_cast(value); + data[i] = static_cast(value >> 8); +} + +inline void Byte::Set3Bytes(uint8_t *data, size_t i, uint32_t value) { + data[i + 2] = static_cast(value); + data[i + 1] = static_cast(value >> 8); + data[i] = static_cast(value >> 16); +} + +inline void Byte::Set4Bytes(uint8_t *data, size_t i, uint32_t value) { + data[i + 3] = static_cast(value); + data[i + 2] = static_cast(value >> 8); + data[i + 1] = static_cast(value >> 16); + data[i] = static_cast(value >> 24); +} + +inline void Byte::Set8Bytes(uint8_t *data, size_t i, uint64_t value) { + data[i + 7] = static_cast(value); + data[i + 6] = static_cast(value >> 8); + data[i + 5] = static_cast(value >> 16); + data[i + 4] = static_cast(value >> 24); + data[i + 3] = static_cast(value >> 32); + data[i + 2] = static_cast(value >> 40); + data[i + 1] = static_cast(value >> 48); + data[i] = static_cast(value >> 56); +} + +inline uint16_t Byte::PadTo4Bytes(uint16_t size) { + // If size is not multiple of 32 bits then pad it. + if (size & 0x03) + return (size & 0xFFFC) + 4; + else + return size; +} + +inline uint32_t Byte::PadTo4Bytes(uint32_t size) { + // If size is not multiple of 32 bits then pad it. + if (size & 0x03) + return (size & 0xFFFFFFFC) + 4; + else + return size; +} + +class Bits { +public: + static size_t CountSetBits(const uint16_t mask); +}; + +/* Inline static methods. */ + +class Crypto { +public: + static void ClassInit(); + static void ClassDestroy(); + static uint32_t GetRandomUInt(uint32_t min, uint32_t max); + static const std::string GetRandomString(size_t len); + static uint32_t GetCRC32(const uint8_t *data, size_t size); + static const uint8_t *GetHmacShA1(const std::string &key, const uint8_t *data, size_t len); + +private: + static uint32_t seed; + static HMAC_CTX *hmacSha1Ctx; + static uint8_t hmacSha1Buffer[]; + static const uint32_t crc32Table[256]; +}; + +/* Inline static methods. */ + +inline uint32_t Crypto::GetRandomUInt(uint32_t min, uint32_t max) { + // NOTE: This is the original, but produces very small values. + // Crypto::seed = (214013 * Crypto::seed) + 2531011; + // return (((Crypto::seed>>16)&0x7FFF) % (max - min + 1)) + min; + + // This seems to produce better results. + Crypto::seed = uint32_t{((214013 * Crypto::seed) + 2531011)}; + + return (((Crypto::seed >> 4) & 0x7FFF7FFF) % (max - min + 1)) + min; +} + +inline const std::string Crypto::GetRandomString(size_t len) { + static char buffer[64]; + static const char chars[] = {'0', '1', '2', '3', '4', '5', '6', '7', '8', '9', 'a', 'b', + 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', + 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z'}; + + if (len > 64) len = 64; + + for (size_t i{0}; i < len; ++i) { + buffer[i] = chars[GetRandomUInt(0, sizeof(chars) - 1)]; + } + + return std::string(buffer, len); +} + +inline uint32_t Crypto::GetCRC32(const uint8_t *data, size_t size) { + uint32_t crc{0xFFFFFFFF}; + const uint8_t *p = data; + + while (size--) { + crc = Crypto::crc32Table[(crc ^ *p++) & 0xFF] ^ (crc >> 8); + } + + return crc ^ ~0U; +} + +class String { +public: + static void ToLowerCase(std::string &str); +}; + +inline void String::ToLowerCase(std::string &str) { + std::transform(str.begin(), str.end(), str.begin(), ::tolower); +} + +class Time { + // Seconds from Jan 1, 1900 to Jan 1, 1970. + static constexpr uint32_t UnixNtpOffset{0x83AA7E80}; + // NTP fractional unit. + static constexpr uint64_t NtpFractionalUnit{1LL << 32}; + +public: + struct Ntp { + uint32_t seconds; + uint32_t fractions; + }; + + static Time::Ntp TimeMs2Ntp(uint64_t ms); + static uint64_t Ntp2TimeMs(Time::Ntp ntp); + static bool IsNewerTimestamp(uint32_t timestamp, uint32_t prevTimestamp); + static uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2); +}; + +inline Time::Ntp Time::TimeMs2Ntp(uint64_t ms) { + Time::Ntp ntp;// NOLINT(cppcoreguidelines-pro-type-member-init) + + ntp.seconds = uint32_t(ms / 1000); + ntp.fractions = + static_cast((static_cast(ms % 1000) / 1000) * NtpFractionalUnit); + + return ntp; +} + +inline uint64_t Time::Ntp2TimeMs(Time::Ntp ntp) { + // clang-format off + return ( + static_cast(ntp.seconds) * 1000 + + static_cast(std::round((static_cast(ntp.fractions) * 1000) / NtpFractionalUnit)) + ); + // clang-format on +} + +inline bool Time::IsNewerTimestamp(uint32_t timestamp, uint32_t prevTimestamp) { + // Distinguish between elements that are exactly 0x80000000 apart. + // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true, + // IsNewer(t2,t1)=false + // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false. + if (static_cast(timestamp - prevTimestamp) == 0x80000000) + return timestamp > prevTimestamp; + + return timestamp != prevTimestamp && + static_cast(timestamp - prevTimestamp) < 0x80000000; +} + +inline uint32_t Time::LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) { + return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2; +} + +}// namespace Utils + +#endif diff --git a/webrtc/webrtc_transport.cc b/webrtc/webrtc_transport.cc new file mode 100644 index 00000000..a475f2ac --- /dev/null +++ b/webrtc/webrtc_transport.cc @@ -0,0 +1,215 @@ +#include "webrtc_transport.h" +#include +#include "Rtcp/Rtcp.h" + +WebRtcTransport::WebRtcTransport() { + static onceToken token([](){ + Utils::Crypto::ClassInit(); + RTC::DtlsTransport::ClassInit(); + RTC::DepLibSRTP::ClassInit(); + RTC::SrtpSession::ClassInit(); + }); + + ice_server_ = std::make_shared(Utils::Crypto::GetRandomString(4), Utils::Crypto::GetRandomString(24)); + ice_server_->SetIceServerCompletedCB([this]() { + this->OnIceServerCompleted(); + }); + ice_server_->SetSendCB([this](char *buf, size_t len, struct sockaddr_in *remote_address) { + this->WritePacket(buf, len, remote_address); + }); + + // todo dtls服务器或客户端模式 + dtls_transport_ = std::make_shared(true); + dtls_transport_->SetHandshakeCompletedCB([this](std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite) { + this->OnDtlsCompleted(client_key, server_key, srtp_crypto_suite); + }); + dtls_transport_->SetOutPutCB([this](char *buf, size_t len) { this->WritePacket(buf, len); }); +} + +WebRtcTransport::~WebRtcTransport() {} + +std::string WebRtcTransport::GetLocalSdp() { + char sdp[1024 * 10] = {0}; + auto ssrc = getSSRC(); + auto ip = getIP(); + auto pt = getPayloadType(); + auto port = getPort(); + sprintf(sdp, + "v=0\r\n" + "o=- 1495799811084970 1495799811084970 IN IP4 %s\r\n" + "s=Streaming Test\r\n" + "t=0 0\r\n" + "a=group:BUNDLE video\r\n" + "a=msid-semantic: WMS janus\r\n" + "m=video %u RTP/SAVPF %u\r\n" + "c=IN IP4 %s\r\n" + "a=mid:video\r\n" + "a=sendonly\r\n" + "a=rtcp-mux\r\n" + "a=ice-ufrag:%s\r\n" + "a=ice-pwd:%s\r\n" + "a=ice-options:trickle\r\n" + "a=fingerprint:sha-256 %s\r\n" + "a=setup:actpass\r\n" + "a=connection:new\r\n" + "a=rtpmap:%u H264/90000\r\n" + "a=ssrc:%u cname:janusvideo\r\n" + "a=ssrc:%u msid:janus janusv0\r\n" + "a=ssrc:%u mslabel:janus\r\n" + "a=ssrc:%u label:janusv0\r\n" + "a=candidate:%s 1 udp %u %s %u typ %s\r\n", + ip.c_str(), port, pt, ip.c_str(), + ice_server_->GetUsernameFragment().c_str(),ice_server_->GetPassword().c_str(), + dtls_transport_->GetMyFingerprint().c_str(), pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); + return sdp; +} + +void WebRtcTransport::OnIceServerCompleted() { + InfoL; + dtls_transport_->Start(); + onIceConnected(); +} + +void WebRtcTransport::OnDtlsCompleted(std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite) { + InfoL << client_key << " " << server_key << " " << (int)srtp_crypto_suite; + srtp_session_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtp_crypto_suite, (uint8_t *) client_key.c_str(), client_key.size()); + onDtlsCompleted(); +} + +bool is_dtls(char *buf) { + return ((*buf > 19) && (*buf < 64)); +} + +bool is_rtp(char *buf) { + RtpHeader *header = (RtpHeader *) buf; + return ((header->pt < 64) || (header->pt >= 96)); +} + +bool is_rtcp(char *buf) { + RtpHeader *header = (RtpHeader *) buf; + return ((header->pt >= 64) && (header->pt < 96)); +} + +void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, struct sockaddr_in *remote_address) { + if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { + InfoL << "stun:" << hexdump(buf, len); + RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len); + if (packet == nullptr) { + WarnL << "parse stun error" << std::endl; + return; + } + ice_server_->ProcessStunPacket(packet, remote_address); + return; + } + if (DtlsTransport::IsDtlsPacket(buf, len)) { + InfoL << "dtls:" << hexdump(buf, len); + dtls_transport_->InputData(buf, len); + return; + } + if (is_rtp(buf)) { + RtpHeader *header = (RtpHeader *) buf; + InfoL << "rtp:" << header->dumpString(len); + return; + } + if (is_rtcp(buf)) { + RtcpHeader *header = (RtcpHeader *) buf; +// InfoL << "rtcp:" << header->dumpString(); + return; + } +} + +void WebRtcTransport::WritePacket(char *buf, size_t len, struct sockaddr_in *remote_address) { + onWrite(buf, len, remote_address ? remote_address : (ice_server_ ? ice_server_->GetSelectAddr() : nullptr)); +} + +void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { + const uint8_t *p = (uint8_t *) buf; + bool ret = false; + if (srtp_session_) { + ret = srtp_session_->EncryptRtp(&p, &len); + } + if (ret) { + onWrite((char *) p, len, ice_server_->GetSelectAddr()); + } +} + +/////////////////////////////////////////////////////////////////////////////////// + +WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) { + _socket = Socket::createSocket(poller, false); + //随机端口，绑定全部网卡 + _socket->bindUdpSock(0); + _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len){ + OnInputDataPacket(buf->data(), buf->size(), (struct sockaddr_in*)addr); + }); +} + +void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { + assert(src); + _src = src; +} + +void WebRtcTransportImp::onDtlsCompleted() { + _reader = _src->getRing()->attach(_socket->getPoller(), true); + weak_ptr weak_self = shared_from_this(); + _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt){ + auto strongSelf = weak_self.lock(); + if (!strongSelf) { + return; + } + pkt->for_each([&](const RtpPacket::Ptr &rtp) { + if(rtp->type == TrackVideo) { + //目前只支持视频 + strongSelf->WritRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, + rtp->size() - RtpPacket::kRtpTcpHeaderSize); + } + }); + }); +} + +void WebRtcTransportImp::onIceConnected(){ + +} + +void WebRtcTransportImp::onWrite(const char *buf, size_t len, struct sockaddr_in *dst) { + auto ptr = BufferRaw::create(); + ptr->assign(buf, len); +// InfoL << len << " " << SockUtil::inet_ntoa(dst->sin_addr) << " " << ntohs(dst->sin_port); + _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr)); +} + +uint32_t WebRtcTransportImp::getSSRC() const { + return _src->getSsrc(TrackVideo); +} + +int WebRtcTransportImp::getPayloadType() const{ + auto sdp = SdpParser(_src->getSdp()); + auto track = sdp.getTrack(TrackVideo); + assert(track); + return track ? track->_pt : 0; +} + +uint16_t WebRtcTransportImp::getPort() const { + //todo udp端口号应该与外网映射端口相同 + return _socket->get_local_port(); +} + +std::string WebRtcTransportImp::getIP() const { + //todo 替换为外网ip + return SockUtil::get_local_ip(); +} + +/////////////////////////////////////////////////////////////////// + +INSTANCE_IMP(WebRtcManager) + +WebRtcManager::WebRtcManager() { + +} + +WebRtcManager::~WebRtcManager() { + +} + + + diff --git a/webrtc/webrtc_transport.h b/webrtc/webrtc_transport.h new file mode 100644 index 00000000..2289e5f1 --- /dev/null +++ b/webrtc/webrtc_transport.h @@ -0,0 +1,112 @@ +#pragma once + +#include +#include + +#include "dtls_transport.h" +#include "ice_server.h" +#include "srtp_session.h" +#include "stun_packet.h" + +class WebRtcTransport { +public: + using Ptr = std::shared_ptr; + WebRtcTransport(); + virtual ~WebRtcTransport(); + + /// 获取本地sdp + /// \return + std::string GetLocalSdp(); + + /// 收到udp数据 + /// \param buf + /// \param len + /// \param remote_address + void OnInputDataPacket(char *buf, size_t len, struct sockaddr_in *remote_address); + + /// 发送rtp + /// \param buf + /// \param len + void WritRtpPacket(char *buf, size_t len); + +protected: + /// 输出udp数据 + /// \param buf + /// \param len + /// \param dst + virtual void onWrite(const char *buf, size_t len, struct sockaddr_in *dst) = 0; + virtual uint32_t getSSRC() const = 0; + virtual uint16_t getPort() const = 0; + virtual std::string getIP() const = 0; + virtual int getPayloadType() const = 0; + virtual void onIceConnected() = 0; + virtual void onDtlsCompleted() = 0; + +private: + void OnIceServerCompleted(); + void OnDtlsCompleted(std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite); + void WritePacket(char *buf, size_t len, struct sockaddr_in *remote_address = nullptr); + +private: + IceServer::Ptr ice_server_; + DtlsTransport::Ptr dtls_transport_; + std::shared_ptr srtp_session_; +}; + +#include "Poller/EventPoller.h" +#include "Network/Socket.h" +#include "Rtsp/RtspMediaSource.h" +using namespace toolkit; +using namespace mediakit; + +class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ +public: + using Ptr = std::shared_ptr; + + WebRtcTransportImp(const EventPoller::Ptr &poller); + ~WebRtcTransportImp() override = default; + + void attach(const RtspMediaSource::Ptr &src); + +protected: + void onWrite(const char *buf, size_t len, struct sockaddr_in *dst) override; + int getPayloadType() const ; + uint32_t getSSRC() const override; + uint16_t getPort() const override; + std::string getIP() const override; + void onIceConnected() override; + void onDtlsCompleted() override; + +private: + Socket::Ptr _socket; + RtspMediaSource::Ptr _src; + RtspMediaSource::RingType::RingReader::Ptr _reader; +}; + +class WebRtcManager : public std::enable_shared_from_this { +public: + ~WebRtcManager(); + static WebRtcManager& Instance(); + +private: + WebRtcManager(); + +}; + + + + + + + + + + + + + + + + + + From 581ebfad71b51b7c76bd9f93b160215bee366dc7 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 25 Mar 2021 16:03:28 +0800 Subject: [PATCH 002/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E6=B5=8B=E8=AF=95web?= =?UTF-8?q?=E6=96=87=E4=BB=B6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- www/webrtc/index.html | 34 + www/webrtc/js/adapter.js | 2475 ++++++++++++++++++++++++++++++++++++++ www/webrtc/js/common.js | 12 + www/webrtc/js/main.js | 134 +++ 4 files changed, 2655 insertions(+) create mode 100644 www/webrtc/index.html create mode 100644 www/webrtc/js/adapter.js create mode 100644 www/webrtc/js/common.js create mode 100644 www/webrtc/js/main.js diff --git a/www/webrtc/index.html b/www/webrtc/index.html new file mode 100644 index 00000000..a9603957 --- /dev/null +++ b/www/webrtc/index.html @@ -0,0 +1,34 @@ + + + + + PeerConnection PRANSWER Demo + + + + + +
+

ip_address

+ +
+ + + + + + + + diff --git a/www/webrtc/js/adapter.js b/www/webrtc/js/adapter.js new file mode 100644 index 00000000..2562bfd7 --- /dev/null +++ b/www/webrtc/js/adapter.js @@ -0,0 +1,2475 @@ +(function(f){if(typeof exports==="object"&&typeof module!=="undefined"){module.exports=f()}else if(typeof define==="function"&&define.amd){define([],f)}else{var g;if(typeof window!=="undefined"){g=window}else if(typeof global!=="undefined"){g=global}else if(typeof self!=="undefined"){g=self}else{g=this}g.adapter = f()}})(function(){var define,module,exports;return (function e(t,n,r){function s(o,u){if(!n[o]){if(!t[o]){var a=typeof require=="function"&&require;if(!u&&a)return a(o,!0);if(i)return i(o,!0);var f=new Error("Cannot find module '"+o+"'");throw f.code="MODULE_NOT_FOUND",f}var l=n[o]={exports:{}};t[o][0].call(l.exports,function(e){var n=t[o][1][e];return s(n?n:e)},l,l.exports,e,t,n,r)}return n[o].exports}var i=typeof require=="function"&&require;for(var o=0;o 0 && typeof selector === 'function') { + return origGetStats(selector, successCallback); + } + + var fixChromeStats_ = function(response) { + var standardReport = {}; + var reports = response.result(); + reports.forEach(function(report) { + var standardStats = { + id: report.id, + timestamp: report.timestamp, + type: report.type + }; + report.names().forEach(function(name) { + standardStats[name] = report.stat(name); + }); + standardReport[standardStats.id] = standardStats; + }); + + return standardReport; + }; + + if (arguments.length >= 2) { + var successCallbackWrapper_ = function(response) { + args[1](fixChromeStats_(response)); + }; + + return origGetStats.apply(this, [successCallbackWrapper_, + arguments[0]]); + } + + // promise-support + return new Promise(function(resolve, reject) { + if (args.length === 1 && typeof selector === 'object') { + origGetStats.apply(self, + [function(response) { + resolve.apply(null, [fixChromeStats_(response)]); + }, reject]); + } else { + origGetStats.apply(self, [resolve, reject]); + } + }); + }; + + return pc; + }; + window.RTCPeerConnection.prototype = webkitRTCPeerConnection.prototype; + + // wrap static methods. Currently just generateCertificate. + if (webkitRTCPeerConnection.generateCertificate) { + Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', { + get: function() { + return webkitRTCPeerConnection.generateCertificate; + } + }); + } + + // add promise support + ['createOffer', 'createAnswer'].forEach(function(method) { + var nativeMethod = webkitRTCPeerConnection.prototype[method]; + webkitRTCPeerConnection.prototype[method] = function() { + var self = this; + if (arguments.length < 1 || (arguments.length === 1 && + typeof(arguments[0]) === 'object')) { + var opts = arguments.length === 1 ? arguments[0] : undefined; + return new Promise(function(resolve, reject) { + nativeMethod.apply(self, [resolve, reject, opts]); + }); + } + return nativeMethod.apply(this, arguments); + }; + }); + + ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] + .forEach(function(method) { + var nativeMethod = webkitRTCPeerConnection.prototype[method]; + webkitRTCPeerConnection.prototype[method] = function() { + var args = arguments; + var self = this; + args[0] = new ((method === 'addIceCandidate')? + RTCIceCandidate : RTCSessionDescription)(args[0]); + return new Promise(function(resolve, reject) { + nativeMethod.apply(self, [args[0], + function() { + resolve(); + if (args.length >= 2) { + args[1].apply(null, []); + } + }, + function(err) { + reject(err); + if (args.length >= 3) { + args[2].apply(null, [err]); + } + }] + ); + }); + }; + }); + }, + + // Attach a media stream to an element. + attachMediaStream: function(element, stream) { + logging('DEPRECATED, attachMediaStream will soon be removed.'); + if (browserDetails.version >= 43) { + element.srcObject = stream; + } else if (typeof element.src !== 'undefined') { + element.src = URL.createObjectURL(stream); + } else { + logging('Error attaching stream to element.'); + } + }, + + reattachMediaStream: function(to, from) { + logging('DEPRECATED, reattachMediaStream will soon be removed.'); + if (browserDetails.version >= 43) { + to.srcObject = from.srcObject; + } else { + to.src = from.src; + } + } +}; + + +// Expose public methods. +module.exports = { + shimOnTrack: chromeShim.shimOnTrack, + shimSourceObject: chromeShim.shimSourceObject, + shimPeerConnection: chromeShim.shimPeerConnection, + shimGetUserMedia: require('./getusermedia'), + attachMediaStream: chromeShim.attachMediaStream, + reattachMediaStream: chromeShim.reattachMediaStream +}; + +},{"../utils.js":9,"./getusermedia":3}],3:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; +var logging = require('../utils.js').log; + +// Expose public methods. +module.exports = function() { + var constraintsToChrome_ = function(c) { + if (typeof c !== 'object' || c.mandatory || c.optional) { + return c; + } + var cc = {}; + Object.keys(c).forEach(function(key) { + if (key === 'require' || key === 'advanced' || key === 'mediaSource') { + return; + } + var r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]}; + if (r.exact !== undefined && typeof r.exact === 'number') { + r.min = r.max = r.exact; + } + var oldname_ = function(prefix, name) { + if (prefix) { + return prefix + name.charAt(0).toUpperCase() + name.slice(1); + } + return (name === 'deviceId') ? 'sourceId' : name; + }; + if (r.ideal !== undefined) { + cc.optional = cc.optional || []; + var oc = {}; + if (typeof r.ideal === 'number') { + oc[oldname_('min', key)] = r.ideal; + cc.optional.push(oc); + oc = {}; + oc[oldname_('max', key)] = r.ideal; + cc.optional.push(oc); + } else { + oc[oldname_('', key)] = r.ideal; + cc.optional.push(oc); + } + } + if (r.exact !== undefined && typeof r.exact !== 'number') { + cc.mandatory = cc.mandatory || {}; + cc.mandatory[oldname_('', key)] = r.exact; + } else { + ['min', 'max'].forEach(function(mix) { + if (r[mix] !== undefined) { + cc.mandatory = cc.mandatory || {}; + cc.mandatory[oldname_(mix, key)] = r[mix]; + } + }); + } + }); + if (c.advanced) { + cc.optional = (cc.optional || []).concat(c.advanced); + } + return cc; + }; + + var getUserMedia_ = function(constraints, onSuccess, onError) { + constraints = JSON.parse(JSON.stringify(constraints)); + if (constraints.audio) { + constraints.audio = constraintsToChrome_(constraints.audio); + } + if (constraints.video) { + constraints.video = constraintsToChrome_(constraints.video); + } + logging('chrome: ' + JSON.stringify(constraints)); + return navigator.webkitGetUserMedia(constraints, onSuccess, onError); + }; + navigator.getUserMedia = getUserMedia_; + + // Returns the result of getUserMedia as a Promise. + var getUserMediaPromise_ = function(constraints) { + return new Promise(function(resolve, reject) { + navigator.getUserMedia(constraints, resolve, reject); + }); + }; + + if (!navigator.mediaDevices) { + navigator.mediaDevices = { + getUserMedia: getUserMediaPromise_, + enumerateDevices: function() { + return new Promise(function(resolve) { + var kinds = {audio: 'audioinput', video: 'videoinput'}; + return MediaStreamTrack.getSources(function(devices) { + resolve(devices.map(function(device) { + return {label: device.label, + kind: kinds[device.kind], + deviceId: device.id, + groupId: ''}; + })); + }); + }); + } + }; + } + + // A shim for getUserMedia method on the mediaDevices object. + // TODO(KaptenJansson) remove once implemented in Chrome stable. + if (!navigator.mediaDevices.getUserMedia) { + navigator.mediaDevices.getUserMedia = function(constraints) { + return getUserMediaPromise_(constraints); + }; + } else { + // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia + // function which returns a Promise, it does not accept spec-style + // constraints. + var origGetUserMedia = navigator.mediaDevices.getUserMedia. + bind(navigator.mediaDevices); + navigator.mediaDevices.getUserMedia = function(c) { + if (c) { + logging('spec: ' + JSON.stringify(c)); // whitespace for alignment + c.audio = constraintsToChrome_(c.audio); + c.video = constraintsToChrome_(c.video); + logging('chrome: ' + JSON.stringify(c)); + } + return origGetUserMedia(c); + }.bind(this); + } + + // Dummy devicechange event methods. + // TODO(KaptenJansson) remove once implemented in Chrome stable. + if (typeof navigator.mediaDevices.addEventListener === 'undefined') { + navigator.mediaDevices.addEventListener = function() { + logging('Dummy mediaDevices.addEventListener called.'); + }; + } + if (typeof navigator.mediaDevices.removeEventListener === 'undefined') { + navigator.mediaDevices.removeEventListener = function() { + logging('Dummy mediaDevices.removeEventListener called.'); + }; + } +}; + +},{"../utils.js":9}],4:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; + +// SDP helpers. +var SDPUtils = {}; + +// Generate an alphanumeric identifier for cname or mids. +// TODO: use UUIDs instead? https://gist.github.com/jed/982883 +SDPUtils.generateIdentifier = function() { + return Math.random().toString(36).substr(2, 10); +}; + +// The RTCP CNAME used by all peerconnections from the same JS. +SDPUtils.localCName = SDPUtils.generateIdentifier(); + +// Splits SDP into lines, dealing with both CRLF and LF. +SDPUtils.splitLines = function(blob) { + return blob.trim().split('\n').map(function(line) { + return line.trim(); + }); +}; +// Splits SDP into sessionpart and mediasections. Ensures CRLF. +SDPUtils.splitSections = function(blob) { + var parts = blob.split('\nm='); + return parts.map(function(part, index) { + return (index > 0 ? 'm=' + part : part).trim() + '\r\n'; + }); +}; + +// Returns lines that start with a certain prefix. +SDPUtils.matchPrefix = function(blob, prefix) { + return SDPUtils.splitLines(blob).filter(function(line) { + return line.indexOf(prefix) === 0; + }); +}; + +// Parses an ICE candidate line. Sample input: +// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8 +// rport 55996" +SDPUtils.parseCandidate = function(line) { + var parts; + // Parse both variants. + if (line.indexOf('a=candidate:') === 0) { + parts = line.substring(12).split(' '); + } else { + parts = line.substring(10).split(' '); + } + + var candidate = { + foundation: parts[0], + component: parts[1], + protocol: parts[2].toLowerCase(), + priority: parseInt(parts[3], 10), + ip: parts[4], + port: parseInt(parts[5], 10), + // skip parts[6] == 'typ' + type: parts[7] + }; + + for (var i = 8; i < parts.length; i += 2) { + switch (parts[i]) { + case 'raddr': + candidate.relatedAddress = parts[i + 1]; + break; + case 'rport': + candidate.relatedPort = parseInt(parts[i + 1], 10); + break; + case 'tcptype': + candidate.tcpType = parts[i + 1]; + break; + default: // Unknown extensions are silently ignored. + break; + } + } + return candidate; +}; + +// Translates a candidate object into SDP candidate attribute. +SDPUtils.writeCandidate = function(candidate) { + var sdp = []; + sdp.push(candidate.foundation); + sdp.push(candidate.component); + sdp.push(candidate.protocol.toUpperCase()); + sdp.push(candidate.priority); + sdp.push(candidate.ip); + sdp.push(candidate.port); + + var type = candidate.type; + sdp.push('typ'); + sdp.push(type); + if (type !== 'host' && candidate.relatedAddress && + candidate.relatedPort) { + sdp.push('raddr'); + sdp.push(candidate.relatedAddress); // was: relAddr + sdp.push('rport'); + sdp.push(candidate.relatedPort); // was: relPort + } + if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') { + sdp.push('tcptype'); + sdp.push(candidate.tcpType); + } + return 'candidate:' + sdp.join(' '); +}; + +// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input: +// a=rtpmap:111 opus/48000/2 +SDPUtils.parseRtpMap = function(line) { + var parts = line.substr(9).split(' '); + var parsed = { + payloadType: parseInt(parts.shift(), 10) // was: id + }; + + parts = parts[0].split('/'); + + parsed.name = parts[0]; + parsed.clockRate = parseInt(parts[1], 10); // was: clockrate + // was: channels + parsed.numChannels = parts.length === 3 ? parseInt(parts[2], 10) : 1; + return parsed; +}; + +// Generate an a=rtpmap line from RTCRtpCodecCapability or +// RTCRtpCodecParameters. +SDPUtils.writeRtpMap = function(codec) { + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate + + (codec.numChannels !== 1 ? '/' + codec.numChannels : '') + '\r\n'; +}; + +// Parses an a=extmap line (headerextension from RFC 5285). Sample input: +// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset +SDPUtils.parseExtmap = function(line) { + var parts = line.substr(9).split(' '); + return { + id: parseInt(parts[0], 10), + uri: parts[1] + }; +}; + +// Generates a=extmap line from RTCRtpHeaderExtensionParameters or +// RTCRtpHeaderExtension. +SDPUtils.writeExtmap = function(headerExtension) { + return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) + + ' ' + headerExtension.uri + '\r\n'; +}; + +// Parses an ftmp line, returns dictionary. Sample input: +// a=fmtp:96 vbr=on;cng=on +// Also deals with vbr=on; cng=on +SDPUtils.parseFmtp = function(line) { + var parsed = {}; + var kv; + var parts = line.substr(line.indexOf(' ') + 1).split(';'); + for (var j = 0; j < parts.length; j++) { + kv = parts[j].trim().split('='); + parsed[kv[0].trim()] = kv[1]; + } + return parsed; +}; + +// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters. +SDPUtils.writeFmtp = function(codec) { + var line = ''; + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + if (codec.parameters && Object.keys(codec.parameters).length) { + var params = []; + Object.keys(codec.parameters).forEach(function(param) { + params.push(param + '=' + codec.parameters[param]); + }); + line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\r\n'; + } + return line; +}; + +// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input: +// a=rtcp-fb:98 nack rpsi +SDPUtils.parseRtcpFb = function(line) { + var parts = line.substr(line.indexOf(' ') + 1).split(' '); + return { + type: parts.shift(), + parameter: parts.join(' ') + }; +}; +// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters. +SDPUtils.writeRtcpFb = function(codec) { + var lines = ''; + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + if (codec.rtcpFeedback && codec.rtcpFeedback.length) { + // FIXME: special handling for trr-int? + codec.rtcpFeedback.forEach(function(fb) { + lines += 'a=rtcp-fb:' + pt + ' ' + fb.type + ' ' + fb.parameter + + '\r\n'; + }); + } + return lines; +}; + +// Parses an RFC 5576 ssrc media attribute. Sample input: +// a=ssrc:3735928559 cname:something +SDPUtils.parseSsrcMedia = function(line) { + var sp = line.indexOf(' '); + var parts = { + ssrc: parseInt(line.substr(7, sp - 7), 10) + }; + var colon = line.indexOf(':', sp); + if (colon > -1) { + parts.attribute = line.substr(sp + 1, colon - sp - 1); + parts.value = line.substr(colon + 1); + } else { + parts.attribute = line.substr(sp + 1); + } + return parts; +}; + +// Extracts DTLS parameters from SDP media section or sessionpart. +// FIXME: for consistency with other functions this should only +// get the fingerprint line as input. See also getIceParameters. +SDPUtils.getDtlsParameters = function(mediaSection, sessionpart) { + var lines = SDPUtils.splitLines(mediaSection); + // Search in session part, too. + lines = lines.concat(SDPUtils.splitLines(sessionpart)); + var fpLine = lines.filter(function(line) { + return line.indexOf('a=fingerprint:') === 0; + })[0].substr(14); + // Note: a=setup line is ignored since we use the 'auto' role. + var dtlsParameters = { + role: 'auto', + fingerprints: [{ + algorithm: fpLine.split(' ')[0], + value: fpLine.split(' ')[1] + }] + }; + return dtlsParameters; +}; + +// Serializes DTLS parameters to SDP. +SDPUtils.writeDtlsParameters = function(params, setupType) { + var sdp = 'a=setup:' + setupType + '\r\n'; + params.fingerprints.forEach(function(fp) { + sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\r\n'; + }); + return sdp; +}; +// Parses ICE information from SDP media section or sessionpart. +// FIXME: for consistency with other functions this should only +// get the ice-ufrag and ice-pwd lines as input. +SDPUtils.getIceParameters = function(mediaSection, sessionpart) { + var lines = SDPUtils.splitLines(mediaSection); + // Search in session part, too. + lines = lines.concat(SDPUtils.splitLines(sessionpart)); + var iceParameters = { + usernameFragment: lines.filter(function(line) { + return line.indexOf('a=ice-ufrag:') === 0; + })[0].substr(12), + password: lines.filter(function(line) { + return line.indexOf('a=ice-pwd:') === 0; + })[0].substr(10) + }; + return iceParameters; +}; + +// Serializes ICE parameters to SDP. +SDPUtils.writeIceParameters = function(params) { + return 'a=ice-ufrag:' + params.usernameFragment + '\r\n' + + 'a=ice-pwd:' + params.password + '\r\n'; +}; + +// Parses the SDP media section and returns RTCRtpParameters. +SDPUtils.parseRtpParameters = function(mediaSection) { + var description = { + codecs: [], + headerExtensions: [], + fecMechanisms: [], + rtcp: [] + }; + var lines = SDPUtils.splitLines(mediaSection); + var mline = lines[0].split(' '); + for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..] + var pt = mline[i]; + var rtpmapline = SDPUtils.matchPrefix( + mediaSection, 'a=rtpmap:' + pt + ' ')[0]; + if (rtpmapline) { + var codec = SDPUtils.parseRtpMap(rtpmapline); + var fmtps = SDPUtils.matchPrefix( + mediaSection, 'a=fmtp:' + pt + ' '); + // Only the first a=fmtp: is considered. + codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {}; + codec.rtcpFeedback = SDPUtils.matchPrefix( + mediaSection, 'a=rtcp-fb:' + pt + ' ') + .map(SDPUtils.parseRtcpFb); + description.codecs.push(codec); + // parse FEC mechanisms from rtpmap lines. + switch (codec.name.toUpperCase()) { + case 'RED': + case 'ULPFEC': + description.fecMechanisms.push(codec.name.toUpperCase()); + break; + default: // only RED and ULPFEC are recognized as FEC mechanisms. + break; + } + } + } + SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) { + description.headerExtensions.push(SDPUtils.parseExtmap(line)); + }); + // FIXME: parse rtcp. + return description; +}; + +// Generates parts of the SDP media section describing the capabilities / +// parameters. +SDPUtils.writeRtpDescription = function(kind, caps) { + var sdp = ''; + + // Build the mline. + sdp += 'm=' + kind + ' '; + sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs. + sdp += ' UDP/TLS/RTP/SAVPF '; + sdp += caps.codecs.map(function(codec) { + if (codec.preferredPayloadType !== undefined) { + return codec.preferredPayloadType; + } + return codec.payloadType; + }).join(' ') + '\r\n'; + + sdp += 'c=IN IP4 0.0.0.0\r\n'; + sdp += 'a=rtcp:9 IN IP4 0.0.0.0\r\n'; + + // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb. + caps.codecs.forEach(function(codec) { + sdp += SDPUtils.writeRtpMap(codec); + sdp += SDPUtils.writeFmtp(codec); + sdp += SDPUtils.writeRtcpFb(codec); + }); + // FIXME: add headerExtensions, fecMechanismş and rtcp. + sdp += 'a=rtcp-mux\r\n'; + return sdp; +}; + +// Parses the SDP media section and returns an array of +// RTCRtpEncodingParameters. +SDPUtils.parseRtpEncodingParameters = function(mediaSection) { + var encodingParameters = []; + var description = SDPUtils.parseRtpParameters(mediaSection); + var hasRed = description.fecMechanisms.indexOf('RED') !== -1; + var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1; + + // filter a=ssrc:... cname:, ignore PlanB-msid + var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') + .map(function(line) { + return SDPUtils.parseSsrcMedia(line); + }) + .filter(function(parts) { + return parts.attribute === 'cname'; + }); + var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc; + var secondarySsrc; + + var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID') + .map(function(line) { + var parts = line.split(' '); + parts.shift(); + return parts.map(function(part) { + return parseInt(part, 10); + }); + }); + if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) { + secondarySsrc = flows[0][1]; + } + + description.codecs.forEach(function(codec) { + if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) { + var encParam = { + ssrc: primarySsrc, + codecPayloadType: parseInt(codec.parameters.apt, 10), + rtx: { + ssrc: secondarySsrc + } + }; + encodingParameters.push(encParam); + if (hasRed) { + encParam = JSON.parse(JSON.stringify(encParam)); + encParam.fec = { + ssrc: secondarySsrc, + mechanism: hasUlpfec ? 'red+ulpfec' : 'red' + }; + encodingParameters.push(encParam); + } + } + }); + if (encodingParameters.length === 0 && primarySsrc) { + encodingParameters.push({ + ssrc: primarySsrc + }); + } + + // we support both b=AS and b=TIAS but interpret AS as TIAS. + var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b='); + if (bandwidth.length) { + if (bandwidth[0].indexOf('b=TIAS:') === 0) { + bandwidth = parseInt(bandwidth[0].substr(7), 10); + } else if (bandwidth[0].indexOf('b=AS:') === 0) { + bandwidth = parseInt(bandwidth[0].substr(5), 10); + } + encodingParameters.forEach(function(params) { + params.maxBitrate = bandwidth; + }); + } + return encodingParameters; +}; + +SDPUtils.writeSessionBoilerplate = function() { + // FIXME: sess-id should be an NTP timestamp. + return 'v=0\r\n' + + 'o=thisisadapterortc 8169639915646943137 2 IN IP4 127.0.0.1\r\n' + + 's=-\r\n' + + 't=0 0\r\n'; +}; + +SDPUtils.writeMediaSection = function(transceiver, caps, type, stream) { + var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps); + + // Map ICE parameters (ufrag, pwd) to SDP. + sdp += SDPUtils.writeIceParameters( + transceiver.iceGatherer.getLocalParameters()); + + // Map DTLS parameters to SDP. + sdp += SDPUtils.writeDtlsParameters( + transceiver.dtlsTransport.getLocalParameters(), + type === 'offer' ? 'actpass' : 'active'); + + sdp += 'a=mid:' + transceiver.mid + '\r\n'; + + if (transceiver.rtpSender && transceiver.rtpReceiver) { + sdp += 'a=sendrecv\r\n'; + } else if (transceiver.rtpSender) { + sdp += 'a=sendonly\r\n'; + } else if (transceiver.rtpReceiver) { + sdp += 'a=recvonly\r\n'; + } else { + sdp += 'a=inactive\r\n'; + } + + // FIXME: for RTX there might be multiple SSRCs. Not implemented in Edge yet. + if (transceiver.rtpSender) { + var msid = 'msid:' + stream.id + ' ' + + transceiver.rtpSender.track.id + '\r\n'; + sdp += 'a=' + msid; + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' ' + msid; + } + // FIXME: this should be written by writeRtpDescription. + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' cname:' + SDPUtils.localCName + '\r\n'; + return sdp; +}; + +// Gets the direction from the mediaSection or the sessionpart. +SDPUtils.getDirection = function(mediaSection, sessionpart) { + // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv. + var lines = SDPUtils.splitLines(mediaSection); + for (var i = 0; i < lines.length; i++) { + switch (lines[i]) { + case 'a=sendrecv': + case 'a=sendonly': + case 'a=recvonly': + case 'a=inactive': + return lines[i].substr(2); + default: + // FIXME: What should happen here? + } + } + if (sessionpart) { + return SDPUtils.getDirection(sessionpart); + } + return 'sendrecv'; +}; + +// Expose public methods. +module.exports = SDPUtils; + +},{}],5:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; + +var SDPUtils = require('./edge_sdp'); +var logging = require('../utils').log; + +var edgeShim = { + shimPeerConnection: function() { + if (window.RTCIceGatherer) { + // ORTC defines an RTCIceCandidate object but no constructor. + // Not implemented in Edge. + if (!window.RTCIceCandidate) { + window.RTCIceCandidate = function(args) { + return args; + }; + } + // ORTC does not have a session description object but + // other browsers (i.e. Chrome) that will support both PC and ORTC + // in the future might have this defined already. + if (!window.RTCSessionDescription) { + window.RTCSessionDescription = function(args) { + return args; + }; + } + } + + window.RTCPeerConnection = function(config) { + var self = this; + + var _eventTarget = document.createDocumentFragment(); + ['addEventListener', 'removeEventListener', 'dispatchEvent'] + .forEach(function(method) { + self[method] = _eventTarget[method].bind(_eventTarget); + }); + + this.onicecandidate = null; + this.onaddstream = null; + this.ontrack = null; + this.onremovestream = null; + this.onsignalingstatechange = null; + this.oniceconnectionstatechange = null; + this.onnegotiationneeded = null; + this.ondatachannel = null; + + this.localStreams = []; + this.remoteStreams = []; + this.getLocalStreams = function() { + return self.localStreams; + }; + this.getRemoteStreams = function() { + return self.remoteStreams; + }; + + this.localDescription = new RTCSessionDescription({ + type: '', + sdp: '' + }); + this.remoteDescription = new RTCSessionDescription({ + type: '', + sdp: '' + }); + this.signalingState = 'stable'; + this.iceConnectionState = 'new'; + this.iceGatheringState = 'new'; + + this.iceOptions = { + gatherPolicy: 'all', + iceServers: [] + }; + if (config && config.iceTransportPolicy) { + switch (config.iceTransportPolicy) { + case 'all': + case 'relay': + this.iceOptions.gatherPolicy = config.iceTransportPolicy; + break; + case 'none': + // FIXME: remove once implementation and spec have added this. + throw new TypeError('iceTransportPolicy "none" not supported'); + default: + // don't set iceTransportPolicy. + break; + } + } + if (config && config.iceServers) { + // Edge does not like + // 1) stun: + // 2) turn: that does not have all of turn:host:port?transport=udp + this.iceOptions.iceServers = config.iceServers.filter(function(server) { + if (server && server.urls) { + server.urls = server.urls.filter(function(url) { + return url.indexOf('turn:') === 0 && + url.indexOf('transport=udp') !== -1; + })[0]; + return !!server.urls; + } + return false; + }); + } + + // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ... + // everything that is needed to describe a SDP m-line. + this.transceivers = []; + + // since the iceGatherer is currently created in createOffer but we + // must not emit candidates until after setLocalDescription we buffer + // them in this array. + this._localIceCandidatesBuffer = []; + }; + + window.RTCPeerConnection.prototype._emitBufferedCandidates = function() { + var self = this; + var sections = SDPUtils.splitSections(self.localDescription.sdp); + // FIXME: need to apply ice candidates in a way which is async but + // in-order + this._localIceCandidatesBuffer.forEach(function(event) { + var end = !event.candidate || Object.keys(event.candidate).length === 0; + if (end) { + for (var j = 1; j < sections.length; j++) { + if (sections[j].indexOf('\r\na=end-of-candidates\r\n') === -1) { + sections[j] += 'a=end-of-candidates\r\n'; + } + } + } else if (event.candidate.candidate.indexOf('typ endOfCandidates') + === -1) { + sections[event.candidate.sdpMLineIndex + 1] += + 'a=' + event.candidate.candidate + '\r\n'; + } + self.localDescription.sdp = sections.join(''); + self.dispatchEvent(event); + if (self.onicecandidate !== null) { + self.onicecandidate(event); + } + if (!event.candidate && self.iceGatheringState !== 'complete') { + var complete = self.transceivers.every(function(transceiver) { + return transceiver.iceGatherer && + transceiver.iceGatherer.state === 'completed'; + }); + if (complete) { + self.iceGatheringState = 'complete'; + } + } + }); + this._localIceCandidatesBuffer = []; + }; + + window.RTCPeerConnection.prototype.addStream = function(stream) { + // Clone is necessary for local demos mostly, attaching directly + // to two different senders does not work (build 10547). + this.localStreams.push(stream.clone()); + this._maybeFireNegotiationNeeded(); + }; + + window.RTCPeerConnection.prototype.removeStream = function(stream) { + var idx = this.localStreams.indexOf(stream); + if (idx > -1) { + this.localStreams.splice(idx, 1); + this._maybeFireNegotiationNeeded(); + } + }; + + // Determines the intersection of local and remote capabilities. + window.RTCPeerConnection.prototype._getCommonCapabilities = + function(localCapabilities, remoteCapabilities) { + var commonCapabilities = { + codecs: [], + headerExtensions: [], + fecMechanisms: [] + }; + localCapabilities.codecs.forEach(function(lCodec) { + for (var i = 0; i < remoteCapabilities.codecs.length; i++) { + var rCodec = remoteCapabilities.codecs[i]; + if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() && + lCodec.clockRate === rCodec.clockRate && + lCodec.numChannels === rCodec.numChannels) { + // push rCodec so we reply with offerer payload type + commonCapabilities.codecs.push(rCodec); + + // FIXME: also need to determine intersection between + // .rtcpFeedback and .parameters + break; + } + } + }); + + localCapabilities.headerExtensions + .forEach(function(lHeaderExtension) { + for (var i = 0; i < remoteCapabilities.headerExtensions.length; + i++) { + var rHeaderExtension = remoteCapabilities.headerExtensions[i]; + if (lHeaderExtension.uri === rHeaderExtension.uri) { + commonCapabilities.headerExtensions.push(rHeaderExtension); + break; + } + } + }); + + // FIXME: fecMechanisms + return commonCapabilities; + }; + + // Create ICE gatherer, ICE transport and DTLS transport. + window.RTCPeerConnection.prototype._createIceAndDtlsTransports = + function(mid, sdpMLineIndex) { + var self = this; + var iceGatherer = new RTCIceGatherer(self.iceOptions); + var iceTransport = new RTCIceTransport(iceGatherer); + iceGatherer.onlocalcandidate = function(evt) { + var event = new Event('icecandidate'); + event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex}; + + var cand = evt.candidate; + var end = !cand || Object.keys(cand).length === 0; + // Edge emits an empty object for RTCIceCandidateComplete‥ + if (end) { + // polyfill since RTCIceGatherer.state is not implemented in + // Edge 10547 yet. + if (iceGatherer.state === undefined) { + iceGatherer.state = 'completed'; + } + + // Emit a candidate with type endOfCandidates to make the samples + // work. Edge requires addIceCandidate with this empty candidate + // to start checking. The real solution is to signal + // end-of-candidates to the other side when getting the null + // candidate but some apps (like the samples) don't do that. + event.candidate.candidate = + 'candidate:1 1 udp 1 0.0.0.0 9 typ endOfCandidates'; + } else { + // RTCIceCandidate doesn't have a component, needs to be added + cand.component = iceTransport.component === 'RTCP' ? 2 : 1; + event.candidate.candidate = SDPUtils.writeCandidate(cand); + } + + var complete = self.transceivers.every(function(transceiver) { + return transceiver.iceGatherer && + transceiver.iceGatherer.state === 'completed'; + }); + + // Emit candidate if localDescription is set. + // Also emits null candidate when all gatherers are complete. + switch (self.iceGatheringState) { + case 'new': + self._localIceCandidatesBuffer.push(event); + if (end && complete) { + self._localIceCandidatesBuffer.push( + new Event('icecandidate')); + } + break; + case 'gathering': + self._emitBufferedCandidates(); + self.dispatchEvent(event); + if (self.onicecandidate !== null) { + self.onicecandidate(event); + } + if (complete) { + self.dispatchEvent(new Event('icecandidate')); + if (self.onicecandidate !== null) { + self.onicecandidate(new Event('icecandidate')); + } + self.iceGatheringState = 'complete'; + } + break; + case 'complete': + // should not happen... currently! + break; + default: // no-op. + break; + } + }; + iceTransport.onicestatechange = function() { + self._updateConnectionState(); + }; + + var dtlsTransport = new RTCDtlsTransport(iceTransport); + dtlsTransport.ondtlsstatechange = function() { + self._updateConnectionState(); + }; + dtlsTransport.onerror = function() { + // onerror does not set state to failed by itself. + dtlsTransport.state = 'failed'; + self._updateConnectionState(); + }; + + return { + iceGatherer: iceGatherer, + iceTransport: iceTransport, + dtlsTransport: dtlsTransport + }; + }; + + // Start the RTP Sender and Receiver for a transceiver. + window.RTCPeerConnection.prototype._transceive = function(transceiver, + send, recv) { + var params = this._getCommonCapabilities(transceiver.localCapabilities, + transceiver.remoteCapabilities); + if (send && transceiver.rtpSender) { + params.encodings = transceiver.sendEncodingParameters; + params.rtcp = { + cname: SDPUtils.localCName + }; + if (transceiver.recvEncodingParameters.length) { + params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc; + } + transceiver.rtpSender.send(params); + } + if (recv && transceiver.rtpReceiver) { + params.encodings = transceiver.recvEncodingParameters; + params.rtcp = { + cname: transceiver.cname + }; + if (transceiver.sendEncodingParameters.length) { + params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc; + } + transceiver.rtpReceiver.receive(params); + } + }; + + window.RTCPeerConnection.prototype.setLocalDescription = + function(description) { + var self = this; + var sections; + var sessionpart; + if (description.type === 'offer') { + // FIXME: What was the purpose of this empty if statement? + // if (!this._pendingOffer) { + // } else { + if (this._pendingOffer) { + // VERY limited support for SDP munging. Limited to: + // * changing the order of codecs + sections = SDPUtils.splitSections(description.sdp); + sessionpart = sections.shift(); + sections.forEach(function(mediaSection, sdpMLineIndex) { + var caps = SDPUtils.parseRtpParameters(mediaSection); + self._pendingOffer[sdpMLineIndex].localCapabilities = caps; + }); + this.transceivers = this._pendingOffer; + delete this._pendingOffer; + } + } else if (description.type === 'answer') { + sections = SDPUtils.splitSections(self.remoteDescription.sdp); + sessionpart = sections.shift(); + sections.forEach(function(mediaSection, sdpMLineIndex) { + var transceiver = self.transceivers[sdpMLineIndex]; + var iceGatherer = transceiver.iceGatherer; + var iceTransport = transceiver.iceTransport; + var dtlsTransport = transceiver.dtlsTransport; + var localCapabilities = transceiver.localCapabilities; + var remoteCapabilities = transceiver.remoteCapabilities; + var rejected = mediaSection.split('\n', 1)[0] + .split(' ', 2)[1] === '0'; + + if (!rejected) { + var remoteIceParameters = SDPUtils.getIceParameters( + mediaSection, sessionpart); + iceTransport.start(iceGatherer, remoteIceParameters, + 'controlled'); + + var remoteDtlsParameters = SDPUtils.getDtlsParameters( + mediaSection, sessionpart); + dtlsTransport.start(remoteDtlsParameters); + + // Calculate intersection of capabilities. + var params = self._getCommonCapabilities(localCapabilities, + remoteCapabilities); + + // Start the RTCRtpSender. The RTCRtpReceiver for this + // transceiver has already been started in setRemoteDescription. + self._transceive(transceiver, + params.codecs.length > 0, + false); + } + }); + } + + this.localDescription = { + type: description.type, + sdp: description.sdp + }; + switch (description.type) { + case 'offer': + this._updateSignalingState('have-local-offer'); + break; + case 'answer': + this._updateSignalingState('stable'); + break; + default: + throw new TypeError('unsupported type "' + description.type + + '"'); + } + + // If a success callback was provided, emit ICE candidates after it + // has been executed. Otherwise, emit callback after the Promise is + // resolved. + var hasCallback = arguments.length > 1 && + typeof arguments[1] === 'function'; + if (hasCallback) { + var cb = arguments[1]; + window.setTimeout(function() { + cb(); + if (self.iceGatheringState === 'new') { + self.iceGatheringState = 'gathering'; + } + self._emitBufferedCandidates(); + }, 0); + } + var p = Promise.resolve(); + p.then(function() { + if (!hasCallback) { + if (self.iceGatheringState === 'new') { + self.iceGatheringState = 'gathering'; + } + // Usually candidates will be emitted earlier. + window.setTimeout(self._emitBufferedCandidates.bind(self), 500); + } + }); + return p; + }; + + window.RTCPeerConnection.prototype.setRemoteDescription = + function(description) { + var self = this; + var stream = new MediaStream(); + var receiverList = []; + var sections = SDPUtils.splitSections(description.sdp); + var sessionpart = sections.shift(); + sections.forEach(function(mediaSection, sdpMLineIndex) { + var lines = SDPUtils.splitLines(mediaSection); + var mline = lines[0].substr(2).split(' '); + var kind = mline[0]; + var rejected = mline[1] === '0'; + var direction = SDPUtils.getDirection(mediaSection, sessionpart); + + var transceiver; + var iceGatherer; + var iceTransport; + var dtlsTransport; + var rtpSender; + var rtpReceiver; + var sendEncodingParameters; + var recvEncodingParameters; + var localCapabilities; + + var track; + // FIXME: ensure the mediaSection has rtcp-mux set. + var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection); + var remoteIceParameters; + var remoteDtlsParameters; + if (!rejected) { + remoteIceParameters = SDPUtils.getIceParameters(mediaSection, + sessionpart); + remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection, + sessionpart); + } + recvEncodingParameters = + SDPUtils.parseRtpEncodingParameters(mediaSection); + + var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:'); + if (mid.length) { + mid = mid[0].substr(6); + } else { + mid = SDPUtils.generateIdentifier(); + } + + var cname; + // Gets the first SSRC. Note that with RTX there might be multiple + // SSRCs. + var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') + .map(function(line) { + return SDPUtils.parseSsrcMedia(line); + }) + .filter(function(obj) { + return obj.attribute === 'cname'; + })[0]; + if (remoteSsrc) { + cname = remoteSsrc.value; + } + + var isComplete = SDPUtils.matchPrefix(mediaSection, + 'a=end-of-candidates').length > 0; + var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:') + .map(function(cand) { + return SDPUtils.parseCandidate(cand); + }) + .filter(function(cand) { + return cand.component === '1'; + }); + if (description.type === 'offer' && !rejected) { + var transports = self._createIceAndDtlsTransports(mid, + sdpMLineIndex); + if (isComplete) { + transports.iceTransport.setRemoteCandidates(cands); + } + + localCapabilities = RTCRtpReceiver.getCapabilities(kind); + sendEncodingParameters = [{ + ssrc: (2 * sdpMLineIndex + 2) * 1001 + }]; + + rtpReceiver = new RTCRtpReceiver(transports.dtlsTransport, kind); + + track = rtpReceiver.track; + receiverList.push([track, rtpReceiver]); + // FIXME: not correct when there are multiple streams but that is + // not currently supported in this shim. + stream.addTrack(track); + + // FIXME: look at direction. + if (self.localStreams.length > 0 && + self.localStreams[0].getTracks().length >= sdpMLineIndex) { + // FIXME: actually more complicated, needs to match types etc + var localtrack = self.localStreams[0] + .getTracks()[sdpMLineIndex]; + rtpSender = new RTCRtpSender(localtrack, + transports.dtlsTransport); + } + + self.transceivers[sdpMLineIndex] = { + iceGatherer: transports.iceGatherer, + iceTransport: transports.iceTransport, + dtlsTransport: transports.dtlsTransport, + localCapabilities: localCapabilities, + remoteCapabilities: remoteCapabilities, + rtpSender: rtpSender, + rtpReceiver: rtpReceiver, + kind: kind, + mid: mid, + cname: cname, + sendEncodingParameters: sendEncodingParameters, + recvEncodingParameters: recvEncodingParameters + }; + // Start the RTCRtpReceiver now. The RTPSender is started in + // setLocalDescription. + self._transceive(self.transceivers[sdpMLineIndex], + false, + direction === 'sendrecv' || direction === 'sendonly'); + } else if (description.type === 'answer' && !rejected) { + transceiver = self.transceivers[sdpMLineIndex]; + iceGatherer = transceiver.iceGatherer; + iceTransport = transceiver.iceTransport; + dtlsTransport = transceiver.dtlsTransport; + rtpSender = transceiver.rtpSender; + rtpReceiver = transceiver.rtpReceiver; + sendEncodingParameters = transceiver.sendEncodingParameters; + localCapabilities = transceiver.localCapabilities; + + self.transceivers[sdpMLineIndex].recvEncodingParameters = + recvEncodingParameters; + self.transceivers[sdpMLineIndex].remoteCapabilities = + remoteCapabilities; + self.transceivers[sdpMLineIndex].cname = cname; + + if (isComplete) { + iceTransport.setRemoteCandidates(cands); + } + iceTransport.start(iceGatherer, remoteIceParameters, + 'controlling'); + dtlsTransport.start(remoteDtlsParameters); + + self._transceive(transceiver, + direction === 'sendrecv' || direction === 'recvonly', + direction === 'sendrecv' || direction === 'sendonly'); + + if (rtpReceiver && + (direction === 'sendrecv' || direction === 'sendonly')) { + track = rtpReceiver.track; + receiverList.push([track, rtpReceiver]); + stream.addTrack(track); + } else { + // FIXME: actually the receiver should be created later. + delete transceiver.rtpReceiver; + } + } + }); + + this.remoteDescription = { + type: description.type, + sdp: description.sdp + }; + switch (description.type) { + case 'offer': + this._updateSignalingState('have-remote-offer'); + break; + case 'answer': + this._updateSignalingState('stable'); + break; + default: + throw new TypeError('unsupported type "' + description.type + + '"'); + } + if (stream.getTracks().length) { + self.remoteStreams.push(stream); + window.setTimeout(function() { + var event = new Event('addstream'); + event.stream = stream; + self.dispatchEvent(event); + if (self.onaddstream !== null) { + window.setTimeout(function() { + self.onaddstream(event); + }, 0); + } + + receiverList.forEach(function(item) { + var track = item[0]; + var receiver = item[1]; + var trackEvent = new Event('track'); + trackEvent.track = track; + trackEvent.receiver = receiver; + trackEvent.streams = [stream]; + self.dispatchEvent(event); + if (self.ontrack !== null) { + window.setTimeout(function() { + self.ontrack(trackEvent); + }, 0); + } + }); + }, 0); + } + if (arguments.length > 1 && typeof arguments[1] === 'function') { + window.setTimeout(arguments[1], 0); + } + return Promise.resolve(); + }; + + window.RTCPeerConnection.prototype.close = function() { + this.transceivers.forEach(function(transceiver) { + /* not yet + if (transceiver.iceGatherer) { + transceiver.iceGatherer.close(); + } + */ + if (transceiver.iceTransport) { + transceiver.iceTransport.stop(); + } + if (transceiver.dtlsTransport) { + transceiver.dtlsTransport.stop(); + } + if (transceiver.rtpSender) { + transceiver.rtpSender.stop(); + } + if (transceiver.rtpReceiver) { + transceiver.rtpReceiver.stop(); + } + }); + // FIXME: clean up tracks, local streams, remote streams, etc + this._updateSignalingState('closed'); + }; + + // Update the signaling state. + window.RTCPeerConnection.prototype._updateSignalingState = + function(newState) { + this.signalingState = newState; + var event = new Event('signalingstatechange'); + this.dispatchEvent(event); + if (this.onsignalingstatechange !== null) { + this.onsignalingstatechange(event); + } + }; + + // Determine whether to fire the negotiationneeded event. + window.RTCPeerConnection.prototype._maybeFireNegotiationNeeded = + function() { + // Fire away (for now). + var event = new Event('negotiationneeded'); + this.dispatchEvent(event); + if (this.onnegotiationneeded !== null) { + this.onnegotiationneeded(event); + } + }; + + // Update the connection state. + window.RTCPeerConnection.prototype._updateConnectionState = function() { + var self = this; + var newState; + var states = { + 'new': 0, + closed: 0, + connecting: 0, + checking: 0, + connected: 0, + completed: 0, + failed: 0 + }; + this.transceivers.forEach(function(transceiver) { + states[transceiver.iceTransport.state]++; + states[transceiver.dtlsTransport.state]++; + }); + // ICETransport.completed and connected are the same for this purpose. + states.connected += states.completed; + + newState = 'new'; + if (states.failed > 0) { + newState = 'failed'; + } else if (states.connecting > 0 || states.checking > 0) { + newState = 'connecting'; + } else if (states.disconnected > 0) { + newState = 'disconnected'; + } else if (states.new > 0) { + newState = 'new'; + } else if (states.connected > 0 || states.completed > 0) { + newState = 'connected'; + } + + if (newState !== self.iceConnectionState) { + self.iceConnectionState = newState; + var event = new Event('iceconnectionstatechange'); + this.dispatchEvent(event); + if (this.oniceconnectionstatechange !== null) { + this.oniceconnectionstatechange(event); + } + } + }; + + window.RTCPeerConnection.prototype.createOffer = function() { + var self = this; + if (this._pendingOffer) { + throw new Error('createOffer called while there is a pending offer.'); + } + var offerOptions; + if (arguments.length === 1 && typeof arguments[0] !== 'function') { + offerOptions = arguments[0]; + } else if (arguments.length === 3) { + offerOptions = arguments[2]; + } + + var tracks = []; + var numAudioTracks = 0; + var numVideoTracks = 0; + // Default to sendrecv. + if (this.localStreams.length) { + numAudioTracks = this.localStreams[0].getAudioTracks().length; + numVideoTracks = this.localStreams[0].getVideoTracks().length; + } + // Determine number of audio and video tracks we need to send/recv. + if (offerOptions) { + // Reject Chrome legacy constraints. + if (offerOptions.mandatory || offerOptions.optional) { + throw new TypeError( + 'Legacy mandatory/optional constraints not supported.'); + } + if (offerOptions.offerToReceiveAudio !== undefined) { + numAudioTracks = offerOptions.offerToReceiveAudio; + } + if (offerOptions.offerToReceiveVideo !== undefined) { + numVideoTracks = offerOptions.offerToReceiveVideo; + } + } + if (this.localStreams.length) { + // Push local streams. + this.localStreams[0].getTracks().forEach(function(track) { + tracks.push({ + kind: track.kind, + track: track, + wantReceive: track.kind === 'audio' ? + numAudioTracks > 0 : numVideoTracks > 0 + }); + if (track.kind === 'audio') { + numAudioTracks--; + } else if (track.kind === 'video') { + numVideoTracks--; + } + }); + } + // Create M-lines for recvonly streams. + while (numAudioTracks > 0 || numVideoTracks > 0) { + if (numAudioTracks > 0) { + tracks.push({ + kind: 'audio', + wantReceive: true + }); + numAudioTracks--; + } + if (numVideoTracks > 0) { + tracks.push({ + kind: 'video', + wantReceive: true + }); + numVideoTracks--; + } + } + + var sdp = SDPUtils.writeSessionBoilerplate(); + var transceivers = []; + tracks.forEach(function(mline, sdpMLineIndex) { + // For each track, create an ice gatherer, ice transport, + // dtls transport, potentially rtpsender and rtpreceiver. + var track = mline.track; + var kind = mline.kind; + var mid = SDPUtils.generateIdentifier(); + + var transports = self._createIceAndDtlsTransports(mid, sdpMLineIndex); + + var localCapabilities = RTCRtpSender.getCapabilities(kind); + var rtpSender; + var rtpReceiver; + + // generate an ssrc now, to be used later in rtpSender.send + var sendEncodingParameters = [{ + ssrc: (2 * sdpMLineIndex + 1) * 1001 + }]; + if (track) { + rtpSender = new RTCRtpSender(track, transports.dtlsTransport); + } + + if (mline.wantReceive) { + rtpReceiver = new RTCRtpReceiver(transports.dtlsTransport, kind); + } + + transceivers[sdpMLineIndex] = { + iceGatherer: transports.iceGatherer, + iceTransport: transports.iceTransport, + dtlsTransport: transports.dtlsTransport, + localCapabilities: localCapabilities, + remoteCapabilities: null, + rtpSender: rtpSender, + rtpReceiver: rtpReceiver, + kind: kind, + mid: mid, + sendEncodingParameters: sendEncodingParameters, + recvEncodingParameters: null + }; + var transceiver = transceivers[sdpMLineIndex]; + sdp += SDPUtils.writeMediaSection(transceiver, + transceiver.localCapabilities, 'offer', self.localStreams[0]); + }); + + this._pendingOffer = transceivers; + var desc = new RTCSessionDescription({ + type: 'offer', + sdp: sdp + }); + if (arguments.length && typeof arguments[0] === 'function') { + window.setTimeout(arguments[0], 0, desc); + } + return Promise.resolve(desc); + }; + + window.RTCPeerConnection.prototype.createAnswer = function() { + var self = this; + + var sdp = SDPUtils.writeSessionBoilerplate(); + this.transceivers.forEach(function(transceiver) { + // Calculate intersection of capabilities. + var commonCapabilities = self._getCommonCapabilities( + transceiver.localCapabilities, + transceiver.remoteCapabilities); + + sdp += SDPUtils.writeMediaSection(transceiver, commonCapabilities, + 'answer', self.localStreams[0]); + }); + + var desc = new RTCSessionDescription({ + type: 'answer', + sdp: sdp + }); + if (arguments.length && typeof arguments[0] === 'function') { + window.setTimeout(arguments[0], 0, desc); + } + return Promise.resolve(desc); + }; + + window.RTCPeerConnection.prototype.addIceCandidate = function(candidate) { + var mLineIndex = candidate.sdpMLineIndex; + if (candidate.sdpMid) { + for (var i = 0; i < this.transceivers.length; i++) { + if (this.transceivers[i].mid === candidate.sdpMid) { + mLineIndex = i; + break; + } + } + } + var transceiver = this.transceivers[mLineIndex]; + if (transceiver) { + var cand = Object.keys(candidate.candidate).length > 0 ? + SDPUtils.parseCandidate(candidate.candidate) : {}; + // Ignore Chrome's invalid candidates since Edge does not like them. + if (cand.protocol === 'tcp' && cand.port === 0) { + return; + } + // Ignore RTCP candidates, we assume RTCP-MUX. + if (cand.component !== '1') { + return; + } + // A dirty hack to make samples work. + if (cand.type === 'endOfCandidates') { + cand = {}; + } + transceiver.iceTransport.addRemoteCandidate(cand); + + // update the remoteDescription. + var sections = SDPUtils.splitSections(this.remoteDescription.sdp); + sections[mLineIndex + 1] += (cand.type ? candidate.candidate.trim() + : 'a=end-of-candidates') + '\r\n'; + this.remoteDescription.sdp = sections.join(''); + } + if (arguments.length > 1 && typeof arguments[1] === 'function') { + window.setTimeout(arguments[1], 0); + } + return Promise.resolve(); + }; + + window.RTCPeerConnection.prototype.getStats = function() { + var promises = []; + this.transceivers.forEach(function(transceiver) { + ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport', + 'dtlsTransport'].forEach(function(method) { + if (transceiver[method]) { + promises.push(transceiver[method].getStats()); + } + }); + }); + var cb = arguments.length > 1 && typeof arguments[1] === 'function' && + arguments[1]; + return new Promise(function(resolve) { + var results = {}; + Promise.all(promises).then(function(res) { + res.forEach(function(result) { + Object.keys(result).forEach(function(id) { + results[id] = result[id]; + }); + }); + if (cb) { + window.setTimeout(cb, 0, results); + } + resolve(results); + }); + }); + }; + }, + + // Attach a media stream to an element. + attachMediaStream: function(element, stream) { + logging('DEPRECATED, attachMediaStream will soon be removed.'); + element.srcObject = stream; + }, + + reattachMediaStream: function(to, from) { + logging('DEPRECATED, reattachMediaStream will soon be removed.'); + to.srcObject = from.srcObject; + } +}; + +// Expose public methods. +module.exports = { + shimPeerConnection: edgeShim.shimPeerConnection, + attachMediaStream: edgeShim.attachMediaStream, + reattachMediaStream: edgeShim.reattachMediaStream +}; + +},{"../utils":9,"./edge_sdp":4}],6:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; + +var logging = require('../utils').log; +var browserDetails = require('../utils').browserDetails; + +var firefoxShim = { + shimOnTrack: function() { + if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in + window.RTCPeerConnection.prototype)) { + Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', { + get: function() { + return this._ontrack; + }, + set: function(f) { + if (this._ontrack) { + this.removeEventListener('track', this._ontrack); + this.removeEventListener('addstream', this._ontrackpoly); + } + this.addEventListener('track', this._ontrack = f); + this.addEventListener('addstream', this._ontrackpoly = function(e) { + e.stream.getTracks().forEach(function(track) { + var event = new Event('track'); + event.track = track; + event.receiver = {track: track}; + event.streams = [e.stream]; + this.dispatchEvent(event); + }.bind(this)); + }.bind(this)); + } + }); + } + }, + + shimSourceObject: function() { + // Firefox has supported mozSrcObject since FF22, unprefixed in 42. + if (typeof window === 'object') { + if (window.HTMLMediaElement && + !('srcObject' in window.HTMLMediaElement.prototype)) { + // Shim the srcObject property, once, when HTMLMediaElement is found. + Object.defineProperty(window.HTMLMediaElement.prototype, 'srcObject', { + get: function() { + return this.mozSrcObject; + }, + set: function(stream) { + this.mozSrcObject = stream; + } + }); + } + } + }, + + shimPeerConnection: function() { + // The RTCPeerConnection object. + if (!window.RTCPeerConnection) { + window.RTCPeerConnection = function(pcConfig, pcConstraints) { + if (browserDetails.version < 38) { + // .urls is not supported in FF < 38. + // create RTCIceServers with a single url. + if (pcConfig && pcConfig.iceServers) { + var newIceServers = []; + for (var i = 0; i < pcConfig.iceServers.length; i++) { + var server = pcConfig.iceServers[i]; + if (server.hasOwnProperty('urls')) { + for (var j = 0; j < server.urls.length; j++) { + var newServer = { + url: server.urls[j] + }; + if (server.urls[j].indexOf('turn') === 0) { + newServer.username = server.username; + newServer.credential = server.credential; + } + newIceServers.push(newServer); + } + } else { + newIceServers.push(pcConfig.iceServers[i]); + } + } + pcConfig.iceServers = newIceServers; + } + } + return new mozRTCPeerConnection(pcConfig, pcConstraints); + }; + window.RTCPeerConnection.prototype = mozRTCPeerConnection.prototype; + + // wrap static methods. Currently just generateCertificate. + if (mozRTCPeerConnection.generateCertificate) { + Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', { + get: function() { + return mozRTCPeerConnection.generateCertificate; + } + }); + } + + window.RTCSessionDescription = mozRTCSessionDescription; + window.RTCIceCandidate = mozRTCIceCandidate; + } + + // shim away need for obsolete RTCIceCandidate/RTCSessionDescription. + ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] + .forEach(function(method) { + var nativeMethod = RTCPeerConnection.prototype[method]; + RTCPeerConnection.prototype[method] = function() { + arguments[0] = new ((method === 'addIceCandidate')? + RTCIceCandidate : RTCSessionDescription)(arguments[0]); + return nativeMethod.apply(this, arguments); + }; + }); + }, + + shimGetUserMedia: function() { + // getUserMedia constraints shim. + var getUserMedia_ = function(constraints, onSuccess, onError) { + var constraintsToFF37_ = function(c) { + if (typeof c !== 'object' || c.require) { + return c; + } + var require = []; + Object.keys(c).forEach(function(key) { + if (key === 'require' || key === 'advanced' || + key === 'mediaSource') { + return; + } + var r = c[key] = (typeof c[key] === 'object') ? + c[key] : {ideal: c[key]}; + if (r.min !== undefined || + r.max !== undefined || r.exact !== undefined) { + require.push(key); + } + if (r.exact !== undefined) { + if (typeof r.exact === 'number') { + r. min = r.max = r.exact; + } else { + c[key] = r.exact; + } + delete r.exact; + } + if (r.ideal !== undefined) { + c.advanced = c.advanced || []; + var oc = {}; + if (typeof r.ideal === 'number') { + oc[key] = {min: r.ideal, max: r.ideal}; + } else { + oc[key] = r.ideal; + } + c.advanced.push(oc); + delete r.ideal; + if (!Object.keys(r).length) { + delete c[key]; + } + } + }); + if (require.length) { + c.require = require; + } + return c; + }; + constraints = JSON.parse(JSON.stringify(constraints)); + if (browserDetails.version < 38) { + logging('spec: ' + JSON.stringify(constraints)); + if (constraints.audio) { + constraints.audio = constraintsToFF37_(constraints.audio); + } + if (constraints.video) { + constraints.video = constraintsToFF37_(constraints.video); + } + logging('ff37: ' + JSON.stringify(constraints)); + } + return navigator.mozGetUserMedia(constraints, onSuccess, onError); + }; + + navigator.getUserMedia = getUserMedia_; + + // Returns the result of getUserMedia as a Promise. + var getUserMediaPromise_ = function(constraints) { + return new Promise(function(resolve, reject) { + navigator.getUserMedia(constraints, resolve, reject); + }); + }; + + // Shim for mediaDevices on older versions. + if (!navigator.mediaDevices) { + navigator.mediaDevices = {getUserMedia: getUserMediaPromise_, + addEventListener: function() { }, + removeEventListener: function() { } + }; + } + navigator.mediaDevices.enumerateDevices = + navigator.mediaDevices.enumerateDevices || function() { + return new Promise(function(resolve) { + var infos = [ + {kind: 'audioinput', deviceId: 'default', label: '', groupId: ''}, + {kind: 'videoinput', deviceId: 'default', label: '', groupId: ''} + ]; + resolve(infos); + }); + }; + + if (browserDetails.version < 41) { + // Work around http://bugzil.la/1169665 + var orgEnumerateDevices = + navigator.mediaDevices.enumerateDevices.bind(navigator.mediaDevices); + navigator.mediaDevices.enumerateDevices = function() { + return orgEnumerateDevices().then(undefined, function(e) { + if (e.name === 'NotFoundError') { + return []; + } + throw e; + }); + }; + } + }, + + // Attach a media stream to an element. + attachMediaStream: function(element, stream) { + logging('DEPRECATED, attachMediaStream will soon be removed.'); + element.srcObject = stream; + }, + + reattachMediaStream: function(to, from) { + logging('DEPRECATED, reattachMediaStream will soon be removed.'); + to.srcObject = from.srcObject; + } +}; + +// Expose public methods. +module.exports = { + shimOnTrack: firefoxShim.shimOnTrack, + shimSourceObject: firefoxShim.shimSourceObject, + shimPeerConnection: firefoxShim.shimPeerConnection, + shimGetUserMedia: require('./getusermedia'), + attachMediaStream: firefoxShim.attachMediaStream, + reattachMediaStream: firefoxShim.reattachMediaStream +}; + +},{"../utils":9,"./getusermedia":7}],7:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; + +var logging = require('../utils').log; +var browserDetails = require('../utils').browserDetails; + +// Expose public methods. +module.exports = function() { + // getUserMedia constraints shim. + var getUserMedia_ = function(constraints, onSuccess, onError) { + var constraintsToFF37_ = function(c) { + if (typeof c !== 'object' || c.require) { + return c; + } + var require = []; + Object.keys(c).forEach(function(key) { + if (key === 'require' || key === 'advanced' || key === 'mediaSource') { + return; + } + var r = c[key] = (typeof c[key] === 'object') ? + c[key] : {ideal: c[key]}; + if (r.min !== undefined || + r.max !== undefined || r.exact !== undefined) { + require.push(key); + } + if (r.exact !== undefined) { + if (typeof r.exact === 'number') { + r. min = r.max = r.exact; + } else { + c[key] = r.exact; + } + delete r.exact; + } + if (r.ideal !== undefined) { + c.advanced = c.advanced || []; + var oc = {}; + if (typeof r.ideal === 'number') { + oc[key] = {min: r.ideal, max: r.ideal}; + } else { + oc[key] = r.ideal; + } + c.advanced.push(oc); + delete r.ideal; + if (!Object.keys(r).length) { + delete c[key]; + } + } + }); + if (require.length) { + c.require = require; + } + return c; + }; + constraints = JSON.parse(JSON.stringify(constraints)); + if (browserDetails.version < 38) { + logging('spec: ' + JSON.stringify(constraints)); + if (constraints.audio) { + constraints.audio = constraintsToFF37_(constraints.audio); + } + if (constraints.video) { + constraints.video = constraintsToFF37_(constraints.video); + } + logging('ff37: ' + JSON.stringify(constraints)); + } + return navigator.mozGetUserMedia(constraints, onSuccess, onError); + }; + + navigator.getUserMedia = getUserMedia_; + + // Returns the result of getUserMedia as a Promise. + var getUserMediaPromise_ = function(constraints) { + return new Promise(function(resolve, reject) { + navigator.getUserMedia(constraints, resolve, reject); + }); + }; + + // Shim for mediaDevices on older versions. + if (!navigator.mediaDevices) { + navigator.mediaDevices = {getUserMedia: getUserMediaPromise_, + addEventListener: function() { }, + removeEventListener: function() { } + }; + } + navigator.mediaDevices.enumerateDevices = + navigator.mediaDevices.enumerateDevices || function() { + return new Promise(function(resolve) { + var infos = [ + {kind: 'audioinput', deviceId: 'default', label: '', groupId: ''}, + {kind: 'videoinput', deviceId: 'default', label: '', groupId: ''} + ]; + resolve(infos); + }); + }; + + if (browserDetails.version < 41) { + // Work around http://bugzil.la/1169665 + var orgEnumerateDevices = + navigator.mediaDevices.enumerateDevices.bind(navigator.mediaDevices); + navigator.mediaDevices.enumerateDevices = function() { + return orgEnumerateDevices().then(undefined, function(e) { + if (e.name === 'NotFoundError') { + return []; + } + throw e; + }); + }; + } +}; + +},{"../utils":9}],8:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ +'use strict'; +var safariShim = { + // TODO: DrAlex, should be here, double check against LayoutTests + // shimOnTrack: function() { }, + + // TODO: DrAlex + // attachMediaStream: function(element, stream) { }, + // reattachMediaStream: function(to, from) { }, + + // TODO: once the back-end for the mac port is done, add. + // TODO: check for webkitGTK+ + // shimPeerConnection: function() { }, + + shimGetUserMedia: function() { + navigator.getUserMedia = navigator.webkitGetUserMedia; + } +}; + +// Expose public methods. +module.exports = { + shimGetUserMedia: safariShim.shimGetUserMedia + // TODO + // shimOnTrack: safariShim.shimOnTrack, + // shimPeerConnection: safariShim.shimPeerConnection, + // attachMediaStream: safariShim.attachMediaStream, + // reattachMediaStream: safariShim.reattachMediaStream +}; + +},{}],9:[function(require,module,exports){ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + /* eslint-env node */ +'use strict'; + +var logDisabled_ = false; + +// Utility methods. +var utils = { + disableLog: function(bool) { + if (typeof bool !== 'boolean') { + return new Error('Argument type: ' + typeof bool + + '. Please use a boolean.'); + } + logDisabled_ = bool; + return (bool) ? 'adapter.js logging disabled' : + 'adapter.js logging enabled'; + }, + + log: function() { + if (typeof window === 'object') { + if (logDisabled_) { + return; + } + if (typeof console !== 'undefined' && typeof console.log === 'function') { + console.log.apply(console, arguments); + } + } + }, + + /** + * Extract browser version out of the provided user agent string. + * + * @param {!string} uastring userAgent string. + * @param {!string} expr Regular expression used as match criteria. + * @param {!number} pos position in the version string to be returned. + * @return {!number} browser version. + */ + extractVersion: function(uastring, expr, pos) { + var match = uastring.match(expr); + return match && match.length >= pos && parseInt(match[pos], 10); + }, + + /** + * Browser detector. + * + * @return {object} result containing browser, version and minVersion + * properties. + */ + detectBrowser: function() { + // Returned result object. + var result = {}; + result.browser = null; + result.version = null; + result.minVersion = null; + + // Fail early if it's not a browser + if (typeof window === 'undefined' || !window.navigator) { + result.browser = 'Not a browser.'; + return result; + } + + // Firefox. + if (navigator.mozGetUserMedia) { + result.browser = 'firefox'; + result.version = this.extractVersion(navigator.userAgent, + /Firefox\/([0-9]+)\./, 1); + result.minVersion = 31; + + // all webkit-based browsers + } else if (navigator.webkitGetUserMedia) { + // Chrome, Chromium, Webview, Opera, all use the chrome shim for now + if (window.webkitRTCPeerConnection) { + result.browser = 'chrome'; + result.version = this.extractVersion(navigator.userAgent, + /Chrom(e|ium)\/([0-9]+)\./, 2); + result.minVersion = 38; + + // Safari or unknown webkit-based + // for the time being Safari has support for MediaStreams but not webRTC + } else { + // Safari UA substrings of interest for reference: + // - webkit version: AppleWebKit/602.1.25 (also used in Op,Cr) + // - safari UI version: Version/9.0.3 (unique to Safari) + // - safari UI webkit version: Safari/601.4.4 (also used in Op,Cr) + // + // if the webkit version and safari UI webkit versions are equals, + // ... this is a stable version. + // + // only the internal webkit version is important today to know if + // media streams are supported + // + if (navigator.userAgent.match(/Version\/(\d+).(\d+)/)) { + result.browser = 'safari'; + result.version = this.extractVersion(navigator.userAgent, + /AppleWebKit\/([0-9]+)\./, 1); + result.minVersion = 602; + + // unknown webkit-based browser + } else { + result.browser = 'Unsupported webkit-based browser ' + + 'with GUM support but no WebRTC support.'; + return result; + } + } + + // Edge. + } else if (navigator.mediaDevices && + navigator.userAgent.match(/Edge\/(\d+).(\d+)$/)) { + result.browser = 'edge'; + result.version = this.extractVersion(navigator.userAgent, + /Edge\/(\d+).(\d+)$/, 2); + result.minVersion = 10547; + + // Default fallthrough: not supported. + } else { + result.browser = 'Not a supported browser.'; + return result; + } + + // Warn if version is less than minVersion. + if (result.version < result.minVersion) { + utils.log('Browser: ' + result.browser + ' Version: ' + result.version + + ' < minimum supported version: ' + result.minVersion + + '\n some things might not work!'); + } + + return result; + } +}; + +// Export. +module.exports = { + log: utils.log, + disableLog: utils.disableLog, + browserDetails: utils.detectBrowser(), + extractVersion: utils.extractVersion +}; + +},{}]},{},[1])(1) +}); \ No newline at end of file diff --git a/www/webrtc/js/common.js b/www/webrtc/js/common.js new file mode 100644 index 00000000..d3de5988 --- /dev/null +++ b/www/webrtc/js/common.js @@ -0,0 +1,12 @@ +function trace(text) { + // This function is used for logging. + if (text[text.length - 1] === '\n') { + text = text.substring(0, text.length - 1); + } + if (window.performance) { + var now = (window.performance.now() / 1000).toFixed(3); + console.log(now + ': ' + text); + } else { + console.log(text); + } +} diff --git a/www/webrtc/js/main.js b/www/webrtc/js/main.js new file mode 100644 index 00000000..20c6b2da --- /dev/null +++ b/www/webrtc/js/main.js @@ -0,0 +1,134 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + +'use strict'; + + +var vid2 = document.getElementById('vid2'); +var btn1 = document.getElementById('btn1'); +var btn3 = document.getElementById('btn3'); +var input1 = document.getElementById('input1'); + +btn1.addEventListener('click', start); + +btn3.addEventListener('click', stop); + +btn1.disabled = false; +btn3.disabled = true; + + +var pc2 = null; +var xmlhttp = null; + +function start() { + btn1.disabled = true; + btn3.disabled = false; + trace('Starting Call'); + + var servers = null; + var addr = input1.value; + pc2 = new RTCPeerConnection(servers); + trace('Created remote peer connection object pc2'); + pc2.onicecandidate = iceCallback; + pc2.onaddstream = gotRemoteStream; + + if (window.XMLHttpRequest) + { + // IE7+, Firefox, Chrome, Opera, Safari �����ִ�д��� + xmlhttp=new XMLHttpRequest(); + } + else + { + // IE6, IE5 �����ִ�д��� + xmlhttp=new ActiveXObject("Microsoft.XMLHTTP"); + } + + xmlhttp.onreadystatechange=function() + { + if (xmlhttp.readyState==4 && xmlhttp.status==200) + { + var res = xmlhttp.responseText; + gotoffer(res); + } + } + xmlhttp.open("GET",addr,true); + xmlhttp.send(); + +} + +function onCreateSessionDescriptionError(error) { + trace('Failed to create session description: ' + error.toString()); + stop(); +} + +function onCreateAnswerError(error) { + trace('Failed to set createAnswer: ' + error.toString()); + stop(); +} + +function onSetLocalDescriptionError(error) { + trace('Failed to set setLocalDescription: ' + error.toString()); + stop(); +} + +function onSetLocalDescriptionSuccess() { + trace('localDescription success.'); +} + +function gotoffer(offer) { + + trace('Offer from server \n' + offer); + //??????offer sdp????????RTCSessionDescription???? + var desc = new RTCSessionDescription(); + desc.sdp = offer; + desc.type = 'offer'; + pc2.setRemoteDescription(desc); + // Since the 'remote' side has no media stream we need + // to pass in the right constraints in order for it to + // accept the incoming offer of audio and video. + pc2.createAnswer().then( + gotDescription2, + onCreateSessionDescriptionError + ); +} + +function gotDescription2(desc) { + // Provisional answer, set a=inactive & set sdp type to pranswer. + /*desc.sdp = desc.sdp.replace(/a=recvonly/g, 'a=inactive'); + desc.type = 'pranswer';*/ + + pc2.setLocalDescription(desc).then( + onSetLocalDescriptionSuccess, + onSetLocalDescriptionError + ); + trace('Pranswer from pc2 \n' + desc.sdp); + + //conn.send(JSON.stringify(desc)); + // send desc.sdp to server +} + +function stop() { + trace('Ending Call' + '\n\n'); + pc2.close(); + pc2 = null; +} + +function gotRemoteStream(e) { + vid2.srcObject = e.stream; + trace('Received remote stream'); +} + +function iceCallback(event) { + if (event.candidate) { + trace('Remote ICE candidate: \n ' + event.candidate.candidate); + //conn.send(JSON.stringify(event.candidate)); + } + else { + // All ICE candidates have been sent + } +} From 704421b728f417c189eea3513af8c0fd38a3e0df Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Fri, 26 Mar 2021 11:07:03 +0800 Subject: [PATCH 003/218] =?UTF-8?q?=E5=AE=8C=E5=96=84?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/CMakeLists.txt | 1 - webrtc/dtls_transport.cc | 75 - webrtc/dtls_transport.h | 58 - webrtc/ice_server.cc | 661 ++++++--- webrtc/ice_server.h | 138 +- webrtc/logger.h | 2 +- webrtc/rtc_dtls_transport.cc | 2539 ++++++++++++++++++---------------- webrtc/rtc_dtls_transport.h | 341 +++-- webrtc/srtp_session.cc | 487 +++---- webrtc/srtp_session.h | 89 +- webrtc/stun_packet.cc | 161 ++- webrtc/utils.cc | 139 -- webrtc/utils.h | 202 --- webrtc/webrtc_transport.cc | 133 +- webrtc/webrtc_transport.h | 55 +- www/webrtc/index.html | 2 +- 16 files changed, 2618 insertions(+), 2465 deletions(-) delete mode 100644 webrtc/dtls_transport.cc delete mode 100644 webrtc/dtls_transport.h delete mode 100644 webrtc/utils.cc diff --git a/server/CMakeLists.txt b/server/CMakeLists.txt index c0b3a009..6e0af5e1 100644 --- a/server/CMakeLists.txt +++ b/server/CMakeLists.txt @@ -49,4 +49,3 @@ else() endif() target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) -message(${LINK_LIB_LIST}) diff --git a/webrtc/dtls_transport.cc b/webrtc/dtls_transport.cc deleted file mode 100644 index 69e1f402..00000000 --- a/webrtc/dtls_transport.cc +++ /dev/null @@ -1,75 +0,0 @@ -// -// Created by xueyuegui on 19-12-7. -// - -#include "dtls_transport.h" - -#include - -DtlsTransport::DtlsTransport(bool is_server) : is_server_(is_server) { - dtls_transport_.reset(new RTC::DtlsTransport(this)); -} - -DtlsTransport::~DtlsTransport() {} - -void DtlsTransport::Start() { - if (is_server_) { - dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); - } else { - dtls_transport_->Run(RTC::DtlsTransport::Role::CLIENT); - } -} - -void DtlsTransport::Close() {} - -void DtlsTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {} - -void DtlsTransport::OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, - RTC::CryptoSuite srtp_crypto_suite, - uint8_t *srtpLocalKey, size_t srtpLocalKeyLen, - uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, - std::string &remoteCert) { - std::string client_key; - std::string server_key; - server_key.assign((char *) srtpLocalKey, srtpLocalKeyLen); - client_key.assign((char *) srtpRemoteKey, srtpRemoteKeyLen); - if (is_server_) { - // If we are server, we swap the keys - client_key.swap(server_key); - } - if (handshake_completed_callback_) { - handshake_completed_callback_(client_key, server_key, srtp_crypto_suite); - } -} - -void DtlsTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) { - if (handshake_failed_callback_) { - handshake_failed_callback_(); - } -} - -void DtlsTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {} - -void DtlsTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, - const uint8_t *data, size_t len) { - if (output_callback_) { - output_callback_((char *) data, len); - } -} - -void DtlsTransport::OutputData(char *buf, size_t len) { - if (output_callback_) { - output_callback_(buf, len); - } -} - -void DtlsTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, - const uint8_t *data, size_t len) {} - -bool DtlsTransport::IsDtlsPacket(const char *buf, size_t len) { - return RTC::DtlsTransport::IsDtls((uint8_t *) buf, len); -} - -void DtlsTransport::InputData(char *buf, size_t len) { - dtls_transport_->ProcessDtlsData((uint8_t *) buf, len); -} diff --git a/webrtc/dtls_transport.h b/webrtc/dtls_transport.h deleted file mode 100644 index 41f4c65c..00000000 --- a/webrtc/dtls_transport.h +++ /dev/null @@ -1,58 +0,0 @@ -// -// Created by xueyuegui on 19-12-7. -// - -#ifndef MYWEBRTC_MYDTLSTRANSPORT_H -#define MYWEBRTC_MYDTLSTRANSPORT_H - -#include -#include - -#include "rtc_dtls_transport.h" - -class DtlsTransport : RTC::DtlsTransport::Listener { -public: - typedef std::shared_ptr Ptr; - - DtlsTransport(bool bServer); - ~DtlsTransport(); - - void Start(); - void Close(); - void InputData(char *buf, size_t len); - void OutputData(char *buf, size_t len); - static bool IsDtlsPacket(const char *buf, size_t len); - std::string GetMyFingerprint() { - auto finger_prints = dtls_transport_->GetLocalFingerprints(); - for (size_t i = 0; i < finger_prints.size(); i++) { - if (finger_prints[i].algorithm == RTC::DtlsTransport::FingerprintAlgorithm::SHA256) { - return finger_prints[i].value; - } - } - return ""; - }; - - void SetHandshakeCompletedCB(std::function cb) { - handshake_completed_callback_ = std::move(cb); - } - void SetHandshakeFailedCB(std::function cb) { handshake_failed_callback_ = std::move(cb); } - void SetOutPutCB(std::function cb) { output_callback_ = std::move(cb); } - - /* Pure virtual methods inherited from RTC::DtlsTransport::Listener. */ -public: - void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override; - void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, RTC::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey, size_t srtpLocalKeyLen, uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, std::string &remoteCert) override; - void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override; - void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override; - void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data,size_t len) override; - void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; - -private: - bool is_server_ = false; - std::function handshake_failed_callback_; - std::shared_ptr dtls_transport_; - std::function output_callback_; - std::function handshake_completed_callback_; -}; - -#endif// MYWEBRTC_MYDTLSTRANSPORT_H diff --git a/webrtc/ice_server.cc b/webrtc/ice_server.cc index cf7fb24b..7dc64ec5 100644 --- a/webrtc/ice_server.cc +++ b/webrtc/ice_server.cc @@ -1,201 +1,512 @@ +#define MS_CLASS "RTC::IceServer" +// #define MS_LOG_DEV_LEVEL 3 + +#include #include "ice_server.h" -#include +namespace RTC +{ + /* Static. */ -static constexpr size_t StunSerializeBufferSize{65536}; -static uint8_t StunSerializeBuffer[StunSerializeBufferSize]; + static constexpr size_t StunSerializeBufferSize{ 65536 }; + static uint8_t StunSerializeBuffer[StunSerializeBufferSize]; -IceServer::IceServer() {} + /* Instance methods. */ -IceServer::~IceServer() {} + IceServer::IceServer(Listener* listener, const std::string& usernameFragment, const std::string& password) + : listener(listener), usernameFragment(usernameFragment), password(password) + { + MS_TRACE(); + } -IceServer::IceServer(const std::string &username_fragment, const std::string &password) - : username_fragment_(username_fragment), password_(password) {} + void IceServer::ProcessStunPacket(RTC::StunPacket* packet, RTC::TransportTuple* tuple) + { + MS_TRACE(); -void IceServer::ProcessStunPacket(RTC::StunPacket *packet, sockaddr_in *remote_address) { - // Must be a Binding method. - if (packet->GetMethod() != RTC::StunPacket::Method::BINDING) { - if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) { - ELOG_WARN("unknown method %#.3x in STUN Request => 400", - static_cast(packet->GetMethod())); - ELOG_WARN("unknown method %#.3x in STUN Request => 400", - static_cast(packet->GetMethod())); - // Reply 400. - RTC::StunPacket *response = packet->CreateErrorResponse(400); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - } else { - ELOG_WARN("ignoring STUN Indication or Response with unknown method %#.3x", - static_cast(packet->GetMethod())); - } - return; - } + // Must be a Binding method. + if (packet->GetMethod() != RTC::StunPacket::Method::BINDING) + { + if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) + { + MS_WARN_TAG( + ice, + "unknown method %#.3x in STUN Request => 400", + static_cast(packet->GetMethod())); - // Must use FINGERPRINT (optional for ICE STUN indications). - if (!packet->HasFingerprint() && packet->GetClass() != RTC::StunPacket::Class::INDICATION) { - if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) { - ELOG_WARN("STUN Binding Request without FINGERPRINT => 400"); - // Reply 400. - RTC::StunPacket *response = packet->CreateErrorResponse(400); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - } else { - ELOG_WARN("ignoring STUN Binding Response without FINGERPRINT"); - } - return; - } + // Reply 400. + RTC::StunPacket* response = packet->CreateErrorResponse(400); - switch (packet->GetClass()) { - case RTC::StunPacket::Class::REQUEST: { - // USERNAME, MESSAGE-INTEGRITY and PRIORITY are required. - if (!packet->HasMessageIntegrity() || (packet->GetPriority() == 0u) || - packet->GetUsername().empty()) { - ELOG_WARN("mising required attributes in STUN Binding Request => 400"); + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); - // Reply 400. - RTC::StunPacket *response = packet->CreateErrorResponse(400); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - return; - } + delete response; + } + else + { + MS_WARN_TAG( + ice, + "ignoring STUN Indication or Response with unknown method %#.3x", + static_cast(packet->GetMethod())); + } - // Check authentication. - switch (packet->CheckAuthentication(this->username_fragment_, this->password_)) { - case RTC::StunPacket::Authentication::OK: { - if (!this->old_password_.empty()) { - ELOG_DEBUG("kNew ICE credentials applied"); - this->old_username_fragment_.clear(); - this->old_password_.clear(); - } - break; - } + return; + } - case RTC::StunPacket::Authentication::UNAUTHORIZED: { - // We may have changed our username_fragment_ and password_, so check - // the old ones. - // clang-format off - if (!this->old_username_fragment_.empty() && - !this->old_password_.empty() && - packet->CheckAuthentication(this->old_username_fragment_, this->old_password_) == - RTC::StunPacket::Authentication::OK) { - ELOG_DEBUG("using old ICE credentials"); - break; - } - ELOG_WARN("wrong authentication in STUN Binding Request => 401"); - // Reply 401. - RTC::StunPacket *response = packet->CreateErrorResponse(401); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - return; - } + // Must use FINGERPRINT (optional for ICE STUN indications). + if (!packet->HasFingerprint() && packet->GetClass() != RTC::StunPacket::Class::INDICATION) + { + if (packet->GetClass() == RTC::StunPacket::Class::REQUEST) + { + MS_WARN_TAG(ice, "STUN Binding Request without FINGERPRINT => 400"); - case RTC::StunPacket::Authentication::BAD_REQUEST: { - ELOG_WARN("cannot check authentication in STUN Binding Request => 400"); - // Reply 400. - RTC::StunPacket *response = packet->CreateErrorResponse(400); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - return; - } - } + // Reply 400. + RTC::StunPacket* response = packet->CreateErrorResponse(400); + + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); + + delete response; + } + else + { + MS_WARN_TAG(ice, "ignoring STUN Binding Response without FINGERPRINT"); + } + + return; + } + + switch (packet->GetClass()) + { + case RTC::StunPacket::Class::REQUEST: + { + // USERNAME, MESSAGE-INTEGRITY and PRIORITY are required. + if (!packet->HasMessageIntegrity() || (packet->GetPriority() == 0u) || packet->GetUsername().empty()) + { + MS_WARN_TAG(ice, "mising required attributes in STUN Binding Request => 400"); + + // Reply 400. + RTC::StunPacket* response = packet->CreateErrorResponse(400); + + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); + + delete response; + + return; + } + + // Check authentication. + switch (packet->CheckAuthentication(this->usernameFragment, this->password)) + { + case RTC::StunPacket::Authentication::OK: + { + if (!this->oldPassword.empty()) + { + MS_DEBUG_TAG(ice, "new ICE credentials applied"); + + this->oldUsernameFragment.clear(); + this->oldPassword.clear(); + } + + break; + } + + case RTC::StunPacket::Authentication::UNAUTHORIZED: + { + // We may have changed our usernameFragment and password, so check + // the old ones. + // clang-format off + if ( + !this->oldUsernameFragment.empty() && + !this->oldPassword.empty() && + packet->CheckAuthentication(this->oldUsernameFragment, this->oldPassword) == RTC::StunPacket::Authentication::OK + ) + // clang-format on + { + MS_DEBUG_TAG(ice, "using old ICE credentials"); + + break; + } + + MS_WARN_TAG(ice, "wrong authentication in STUN Binding Request => 401"); + + // Reply 401. + RTC::StunPacket* response = packet->CreateErrorResponse(401); + + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); + + delete response; + + return; + } + + case RTC::StunPacket::Authentication::BAD_REQUEST: + { + MS_WARN_TAG(ice, "cannot check authentication in STUN Binding Request => 400"); + + // Reply 400. + RTC::StunPacket* response = packet->CreateErrorResponse(400); + + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); + + delete response; + + return; + } + } #if 0 - // NOTE: Should be rejected with 487, but this makes Chrome happy: - // https://bugs.chromium.org/p/webrtc/issues/detail?id=7478 - // The remote peer must be ICE controlling. - if (packet->GetIceControlled()) { - MS_WARN_TAG(ice, "peer indicates ICE-CONTROLLED in STUN Binding Request => 487"); - // Reply 487 (Role Conflict). - RTC::StunPacket *response = packet->CreateErrorResponse(487); - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - return; - } + // The remote peer must be ICE controlling. + if (packet->GetIceControlled()) + { + MS_WARN_TAG(ice, "peer indicates ICE-CONTROLLED in STUN Binding Request => 487"); + + // Reply 487 (Role Conflict). + RTC::StunPacket* response = packet->CreateErrorResponse(487); + + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); + + delete response; + + return; + } + #endif - ELOG_DEBUG("processing STUN Binding Request [Priority:%d, UseCandidate:%s]", - static_cast(packet->GetPriority()), - (packet->HasUseCandidate() ? "true" : "false")); - // Create a success response. - RTC::StunPacket *response = packet->CreateSuccessResponse(); - // Add XOR-MAPPED-ADDRESS. - // response->SetXorMappedAddress(tuple->GetRemoteAddress()); - response->SetXorMappedAddress((struct sockaddr *) remote_address); - // Authenticate the response. - if (this->old_password_.empty()) { - response->Authenticate(this->password_); - } else { - response->Authenticate(this->old_password_); - } + MS_DEBUG_DEV( + "processing STUN Binding Request [Priority:%" PRIu32 ", UseCandidate:%s]", + static_cast(packet->GetPriority()), + packet->HasUseCandidate() ? "true" : "false"); - // Send back. - response->Serialize(StunSerializeBuffer); - if (send_callback_) { - send_callback_((char *) StunSerializeBuffer, response->GetSize(), remote_address); - } - delete response; - // Handle the tuple. - HandleTuple(remote_address, packet->HasUseCandidate()); - break; - } + // Create a success response. + RTC::StunPacket* response = packet->CreateSuccessResponse(); - case RTC::StunPacket::Class::INDICATION: { - ELOG_DEBUG("STUN Binding Indication processed"); - break; - } + // Add XOR-MAPPED-ADDRESS. + response->SetXorMappedAddress(tuple); - case RTC::StunPacket::Class::SUCCESS_RESPONSE: { - ELOG_DEBUG("STUN Binding Success Response processed"); - break; - } + // Authenticate the response. + if (this->oldPassword.empty()) + response->Authenticate(this->password); + else + response->Authenticate(this->oldPassword); - case RTC::StunPacket::Class::ERROR_RESPONSE: { - ELOG_DEBUG("STUN Binding Error Response processed"); - break; - } - } -} -void IceServer::HandleTuple(sockaddr_in *remote_address, bool has_use_candidate) { - remote_address_ = *remote_address; - if (has_use_candidate) { - this->state = IceState::kCompleted; - } - if (ice_server_completed_callback_) { - ice_server_completed_callback_(); - ice_server_completed_callback_ = nullptr; - } -} + // Send back. + response->Serialize(StunSerializeBuffer); + this->listener->OnIceServerSendStunPacket(this, response, tuple); -const std::string &IceServer::GetUsernameFragment() const { return this->username_fragment_; } + delete response; -const std::string &IceServer::GetPassword() const { return this->password_; } + // Handle the tuple. + HandleTuple(tuple, packet->HasUseCandidate()); -inline void IceServer::SetUsernameFragment(const std::string &username_fragment) { - this->old_username_fragment_ = this->username_fragment_; - this->username_fragment_ = username_fragment; -} + break; + } -inline void IceServer::SetPassword(const std::string &password) { - this->old_password_ = this->password_; - this->password_ = password; -} + case RTC::StunPacket::Class::INDICATION: + { + MS_DEBUG_TAG(ice, "STUN Binding Indication processed"); -inline IceServer::IceState IceServer::GetState() const { return this->state; } \ No newline at end of file + break; + } + + case RTC::StunPacket::Class::SUCCESS_RESPONSE: + { + MS_DEBUG_TAG(ice, "STUN Binding Success Response processed"); + + break; + } + + case RTC::StunPacket::Class::ERROR_RESPONSE: + { + MS_DEBUG_TAG(ice, "STUN Binding Error Response processed"); + + break; + } + } + } + + bool IceServer::IsValidTuple(const RTC::TransportTuple* tuple) const + { + MS_TRACE(); + + return HasTuple(tuple) != nullptr; + } + + void IceServer::RemoveTuple(RTC::TransportTuple* tuple) + { + MS_TRACE(); + + RTC::TransportTuple* removedTuple{ nullptr }; + + // Find the removed tuple. + auto it = this->tuples.begin(); + + for (; it != this->tuples.end(); ++it) + { + RTC::TransportTuple* storedTuple = std::addressof(*it); + + if (memcmp(storedTuple, tuple, sizeof (RTC::TransportTuple)) == 0) + { + removedTuple = storedTuple; + + break; + } + } + + // If not found, ignore. + if (!removedTuple) + return; + + // Remove from the list of tuples. + this->tuples.erase(it); + + // If this is not the selected tuple, stop here. + if (removedTuple != this->selectedTuple) + return; + + // Otherwise this was the selected tuple. + this->selectedTuple = nullptr; + + // Mark the first tuple as selected tuple (if any). + if (this->tuples.begin() != this->tuples.end()) + { + SetSelectedTuple(std::addressof(*this->tuples.begin())); + } + // Or just emit 'disconnected'. + else + { + // Update state. + this->state = IceState::DISCONNECTED; + // Notify the listener. + this->listener->OnIceServerDisconnected(this); + } + } + + void IceServer::ForceSelectedTuple(const RTC::TransportTuple* tuple) + { + MS_TRACE(); + + MS_ASSERT( + this->selectedTuple, "cannot force the selected tuple if there was not a selected tuple"); + + auto* storedTuple = HasTuple(tuple); + + MS_ASSERT( + storedTuple, + "cannot force the selected tuple if the given tuple was not already a valid tuple"); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + } + + void IceServer::HandleTuple(RTC::TransportTuple* tuple, bool hasUseCandidate) + { + MS_TRACE(); + + switch (this->state) + { + case IceState::NEW: + { + // There should be no tuples. + MS_ASSERT( + this->tuples.empty(), "state is 'new' but there are %zu tuples", this->tuples.size()); + + // There shouldn't be a selected tuple. + MS_ASSERT(!this->selectedTuple, "state is 'new' but there is selected tuple"); + + if (!hasUseCandidate) + { + MS_DEBUG_TAG(ice, "transition from state 'new' to 'connected'"); + + // Store the tuple. + auto* storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + // Update state. + this->state = IceState::CONNECTED; + // Notify the listener. + this->listener->OnIceServerConnected(this); + } + else + { + MS_DEBUG_TAG(ice, "transition from state 'new' to 'completed'"); + + // Store the tuple. + auto* storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + // Update state. + this->state = IceState::COMPLETED; + // Notify the listener. + this->listener->OnIceServerCompleted(this); + } + + break; + } + + case IceState::DISCONNECTED: + { + // There should be no tuples. + MS_ASSERT( + this->tuples.empty(), + "state is 'disconnected' but there are %zu tuples", + this->tuples.size()); + + // There shouldn't be a selected tuple. + MS_ASSERT(!this->selectedTuple, "state is 'disconnected' but there is selected tuple"); + + if (!hasUseCandidate) + { + MS_DEBUG_TAG(ice, "transition from state 'disconnected' to 'connected'"); + + // Store the tuple. + auto* storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + // Update state. + this->state = IceState::CONNECTED; + // Notify the listener. + this->listener->OnIceServerConnected(this); + } + else + { + MS_DEBUG_TAG(ice, "transition from state 'disconnected' to 'completed'"); + + // Store the tuple. + auto* storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + // Update state. + this->state = IceState::COMPLETED; + // Notify the listener. + this->listener->OnIceServerCompleted(this); + } + + break; + } + + case IceState::CONNECTED: + { + // There should be some tuples. + MS_ASSERT(!this->tuples.empty(), "state is 'connected' but there are no tuples"); + + // There should be a selected tuple. + MS_ASSERT(this->selectedTuple, "state is 'connected' but there is not selected tuple"); + + if (!hasUseCandidate) + { + // If a new tuple store it. + if (!HasTuple(tuple)) + AddTuple(tuple); + } + else + { + MS_DEBUG_TAG(ice, "transition from state 'connected' to 'completed'"); + + auto* storedTuple = HasTuple(tuple); + + // If a new tuple store it. + if (!storedTuple) + storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + // Update state. + this->state = IceState::COMPLETED; + // Notify the listener. + this->listener->OnIceServerCompleted(this); + } + + break; + } + + case IceState::COMPLETED: + { + // There should be some tuples. + MS_ASSERT(!this->tuples.empty(), "state is 'completed' but there are no tuples"); + + // There should be a selected tuple. + MS_ASSERT(this->selectedTuple, "state is 'completed' but there is not selected tuple"); + + if (!hasUseCandidate) + { + // If a new tuple store it. + if (!HasTuple(tuple)) + AddTuple(tuple); + } + else + { + auto* storedTuple = HasTuple(tuple); + + // If a new tuple store it. + if (!storedTuple) + storedTuple = AddTuple(tuple); + + // Mark it as selected tuple. + SetSelectedTuple(storedTuple); + } + + break; + } + } + } + + inline RTC::TransportTuple* IceServer::AddTuple(RTC::TransportTuple* tuple) + { + MS_TRACE(); + + // Add the new tuple at the beginning of the list. + this->tuples.push_front(*tuple); + + auto* storedTuple = std::addressof(*this->tuples.begin()); + + // Return the address of the inserted tuple. + return storedTuple; + } + + inline RTC::TransportTuple* IceServer::HasTuple(const RTC::TransportTuple* tuple) const + { + MS_TRACE(); + + // If there is no selected tuple yet then we know that the tuples list + // is empty. + if (!this->selectedTuple) + return nullptr; + + // Check the current selected tuple. + if (memcmp(selectedTuple, tuple, sizeof (RTC::TransportTuple)) == 0) + return this->selectedTuple; + + // Otherwise check other stored tuples. + for (const auto& it : this->tuples) + { + auto* storedTuple = const_cast(std::addressof(it)); + + if (memcmp(storedTuple, tuple, sizeof (RTC::TransportTuple)) == 0) + return storedTuple; + } + + return nullptr; + } + + inline void IceServer::SetSelectedTuple(RTC::TransportTuple* storedTuple) + { + MS_TRACE(); + + // If already the selected tuple do nothing. + if (storedTuple == this->selectedTuple) + return; + + this->selectedTuple = storedTuple; + + // Notify the listener. + this->listener->OnIceServerSelectedTuple(this, this->selectedTuple); + } +} // namespace RTC diff --git a/webrtc/ice_server.h b/webrtc/ice_server.h index d33f26d2..437d9d9f 100644 --- a/webrtc/ice_server.h +++ b/webrtc/ice_server.h @@ -1,40 +1,112 @@ -#pragma once +#ifndef MS_RTC_ICE_SERVER_HPP +#define MS_RTC_ICE_SERVER_HPP +#include "stun_packet.h" +#include "logger.h" +#include +#include #include #include -#include "logger.h" -#include "stun_packet.h" +namespace RTC +{ + using TransportTuple = struct sockaddr; + class IceServer + { + public: + enum class IceState + { + NEW = 1, + CONNECTED, + COMPLETED, + DISCONNECTED + }; -typedef std::function UdpSendCallback; + public: + class Listener + { + public: + virtual ~Listener() = default; -class IceServer { -public: - enum class IceState { kNew = 1, kConnect, kCompleted, kDisconnected }; - typedef std::shared_ptr Ptr; - IceServer(); - IceServer(const std::string &username_fragment, const std::string &password); - const std::string &GetUsernameFragment() const; - const std::string &GetPassword() const; - void SetUsernameFragment(const std::string &username_fragment); - void SetPassword(const std::string &password); - IceState GetState() const; - void ProcessStunPacket(RTC::StunPacket *packet, struct sockaddr_in *remote_address); - void HandleTuple(struct sockaddr_in *remote_address, bool has_use_candidate); - ~IceServer(); - void SetSendCB(UdpSendCallback send_cb) { send_callback_ = send_cb; } - void SetIceServerCompletedCB(std::function cb) { ice_server_completed_callback_ = cb; }; - struct sockaddr_in *GetSelectAddr() { - return &remote_address_; - } + public: + /** + * These callbacks are guaranteed to be called before ProcessStunPacket() + * returns, so the given pointers are still usable. + */ + virtual void OnIceServerSendStunPacket( + const RTC::IceServer* iceServer, const RTC::StunPacket* packet, RTC::TransportTuple* tuple) = 0; + virtual void OnIceServerSelectedTuple( + const RTC::IceServer* iceServer, RTC::TransportTuple* tuple) = 0; + virtual void OnIceServerConnected(const RTC::IceServer* iceServer) = 0; + virtual void OnIceServerCompleted(const RTC::IceServer* iceServer) = 0; + virtual void OnIceServerDisconnected(const RTC::IceServer* iceServer) = 0; + }; -private: - UdpSendCallback send_callback_; - std::function ice_server_completed_callback_; - std::string username_fragment_; - std::string password_; - std::string old_username_fragment_; - std::string old_password_; - IceState state{IceState::kNew}; - struct sockaddr_in remote_address_; -}; + public: + IceServer(Listener* listener, const std::string& usernameFragment, const std::string& password); + + public: + void ProcessStunPacket(RTC::StunPacket* packet, RTC::TransportTuple* tuple); + const std::string& GetUsernameFragment() const + { + return this->usernameFragment; + } + const std::string& GetPassword() const + { + return this->password; + } + IceState GetState() const + { + return this->state; + } + RTC::TransportTuple* GetSelectedTuple() const + { + return this->selectedTuple; + } + void SetUsernameFragment(const std::string& usernameFragment) + { + this->oldUsernameFragment = this->usernameFragment; + this->usernameFragment = usernameFragment; + } + void SetPassword(const std::string& password) + { + this->oldPassword = this->password; + this->password = password; + } + bool IsValidTuple(const RTC::TransportTuple* tuple) const; + void RemoveTuple(RTC::TransportTuple* tuple); + // This should be just called in 'connected' or completed' state + // and the given tuple must be an already valid tuple. + void ForceSelectedTuple(const RTC::TransportTuple* tuple); + + private: + void HandleTuple(RTC::TransportTuple* tuple, bool hasUseCandidate); + /** + * Store the given tuple and return its stored address. + */ + RTC::TransportTuple* AddTuple(RTC::TransportTuple* tuple); + /** + * If the given tuple exists return its stored address, nullptr otherwise. + */ + RTC::TransportTuple* HasTuple(const RTC::TransportTuple* tuple) const; + /** + * Set the given tuple as the selected tuple. + * NOTE: The given tuple MUST be already stored within the list. + */ + void SetSelectedTuple(RTC::TransportTuple* storedTuple); + + private: + // Passed by argument. + Listener* listener{ nullptr }; + // Others. + std::string usernameFragment; + std::string password; + std::string oldUsernameFragment; + std::string oldPassword; + IceState state{ IceState::NEW }; + std::list tuples; + RTC::TransportTuple* selectedTuple{ nullptr }; + }; +} // namespace RTC + +#endif diff --git a/webrtc/logger.h b/webrtc/logger.h index a165536b..3da3984f 100644 --- a/webrtc/logger.h +++ b/webrtc/logger.h @@ -12,7 +12,7 @@ #define MS_DEBUG_2TAGS(tag1, tag2,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) #define MS_WARN_2TAGS(tag1, tag2,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__) #define MS_DEBUG_TAG(tag,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) -#define MS_ASSERT(con, log) assert(con) +#define MS_ASSERT(con, fmt, ...) do{if(!(con)) { printf("assert failed:%s" fmt "\n", #con, ##__VA_ARGS__);} assert(con); } while(false); #define MS_ABORT(fmt, ...) do{ printf("abort:" fmt "\n", ##__VA_ARGS__); abort(); } while(false); #define MS_WARN_TAG(tag,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__) #define MS_DEBUG_DEV(fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) \ No newline at end of file diff --git a/webrtc/rtc_dtls_transport.cc b/webrtc/rtc_dtls_transport.cc index 1087af4a..e66f6469 100644 --- a/webrtc/rtc_dtls_transport.cc +++ b/webrtc/rtc_dtls_transport.cc @@ -2,62 +2,56 @@ // #define MS_LOG_DEV_LEVEL 3 #include "rtc_dtls_transport.h" - +#include "logger.h" #include #include #include #include #include +#include // std::sprintf(), std::fopen() +#include // std::memcpy(), std::strcmp() -#include // std::sprintf(), std::fopen() -#include // std::memcpy(), std::strcmp() - -#include "logger.h" - -typedef struct { - long tv_sec; - long tv_usec; -} uv_timeval_t; - -#define LOG_OPENSSL_ERROR(desc) \ - do { \ - if (ERR_peek_error() == 0) \ - MS_ERROR("OpenSSL error [desc:'%s']", desc); \ - else { \ - int64_t err; \ - while ((err = ERR_get_error()) != 0) { \ - MS_ERROR("OpenSSL error [desc:'%s', error:'%s']", desc, ERR_error_string(err, nullptr)); \ - } \ - ERR_clear_error(); \ - } \ - } while (false) +#define LOG_OPENSSL_ERROR(desc) \ + do \ + { \ + if (ERR_peek_error() == 0) \ + MS_ERROR("OpenSSL error [desc:'%s']", desc); \ + else \ + { \ + int64_t err; \ + while ((err = ERR_get_error()) != 0) \ + { \ + MS_ERROR("OpenSSL error [desc:'%s', error:'%s']", desc, ERR_error_string(err, nullptr)); \ + } \ + ERR_clear_error(); \ + } \ + } while (false) /* Static methods for OpenSSL callbacks. */ -inline static int onSslCertificateVerify(int /*preverifyOk*/, X509_STORE_CTX* /*ctx*/) { - MS_TRACE(); +inline static int onSslCertificateVerify(int /*preverifyOk*/, X509_STORE_CTX* /*ctx*/) +{ + MS_TRACE(); - // Always valid since DTLS certificates are self-signed. - return 1; + // Always valid since DTLS certificates are self-signed. + return 1; } -inline static void onSslInfo(const SSL* ssl, int where, int ret) { - static_cast(SSL_get_ex_data(ssl, 0))->OnSslInfo(where, ret); +inline static unsigned int onSslDtlsTimer(SSL* /*ssl*/, unsigned int timerUs) +{ + if (timerUs == 0) + return 100000; + else if (timerUs >= 4000000) + return 4000000; + else + return 2 * timerUs; } -inline static unsigned int onSslDtlsTimer(SSL* /*ssl*/, unsigned int timerUs) { - if (timerUs == 0) - return 100000; - else if (timerUs >= 4000000) - return 4000000; - else - return 2 * timerUs; -} +namespace RTC +{ + /* Static. */ -namespace RTC { -/* Static. */ - -// clang-format off + // clang-format off static constexpr int DtlsMtu{ 1350 }; static constexpr int SslReadBufferSize{ 65536 }; // AES-HMAC: http://tools.ietf.org/html/rfc3711 @@ -71,15 +65,15 @@ namespace RTC { static constexpr size_t SrtpAesGcm128MasterKeyLength{ 16 }; static constexpr size_t SrtpAesGcm128MasterSaltLength{ 12 }; static constexpr size_t SrtpAesGcm128MasterLength{ SrtpAesGcm128MasterKeyLength + SrtpAesGcm128MasterSaltLength }; -// clang-format on + // clang-format on -/* Class variables. */ + /* Class variables. */ -X509* DtlsTransport::certificate{nullptr}; -EVP_PKEY* DtlsTransport::privateKey{nullptr}; -SSL_CTX* DtlsTransport::sslCtx{nullptr}; -uint8_t DtlsTransport::sslReadBuffer[SslReadBufferSize]; -// clang-format off + X509* DtlsTransport::certificate{ nullptr }; + EVP_PKEY* DtlsTransport::privateKey{ nullptr }; + SSL_CTX* DtlsTransport::sslCtx{ nullptr }; + uint8_t DtlsTransport::sslReadBuffer[SslReadBufferSize]; + // clang-format off std::map DtlsTransport::string2FingerprintAlgorithm = { { "sha-1", DtlsTransport::FingerprintAlgorithm::SHA1 }, @@ -105,1219 +99,1376 @@ uint8_t DtlsTransport::sslReadBuffer[SslReadBufferSize]; std::vector DtlsTransport::localFingerprints; std::vector DtlsTransport::srtpCryptoSuites = { - { RTC::CryptoSuite::AEAD_AES_256_GCM, "SRTP_AEAD_AES_256_GCM" }, - { RTC::CryptoSuite::AEAD_AES_128_GCM, "SRTP_AEAD_AES_128_GCM" }, - { RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_80, "SRTP_AES128_CM_SHA1_80" }, - { RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_32, "SRTP_AES128_CM_SHA1_32" } + { RTC::SrtpSession::CryptoSuite::AEAD_AES_256_GCM, "SRTP_AEAD_AES_256_GCM" }, + { RTC::SrtpSession::CryptoSuite::AEAD_AES_128_GCM, "SRTP_AEAD_AES_128_GCM" }, + { RTC::SrtpSession::CryptoSuite::AES_CM_128_HMAC_SHA1_80, "SRTP_AES128_CM_SHA1_80" }, + { RTC::SrtpSession::CryptoSuite::AES_CM_128_HMAC_SHA1_32, "SRTP_AES128_CM_SHA1_32" } }; -// clang-format on + // clang-format on -/* Class methods. */ + /* Class methods. */ -void DtlsTransport::ClassInit() { - MS_TRACE(); + void DtlsTransport::ClassInit() + { + MS_TRACE(); + + // Generate a X509 certificate and private key (unless PEM files are provided). + if (true /* + Settings::configuration.dtlsCertificateFile.empty() || + Settings::configuration.dtlsPrivateKeyFile.empty()*/) + { + GenerateCertificateAndPrivateKey(); + } + else + { + ReadCertificateAndPrivateKeyFromFiles(); + } + + // Create a global SSL_CTX. + CreateSslCtx(); + + // Generate certificate fingerprints. + GenerateFingerprints(); + } + + void DtlsTransport::ClassDestroy() + { + MS_TRACE(); + + if (DtlsTransport::privateKey) + EVP_PKEY_free(DtlsTransport::privateKey); + if (DtlsTransport::certificate) + X509_free(DtlsTransport::certificate); + if (DtlsTransport::sslCtx) + SSL_CTX_free(DtlsTransport::sslCtx); + } + + void DtlsTransport::GenerateCertificateAndPrivateKey() + { + MS_TRACE(); + + int ret{ 0 }; + EC_KEY* ecKey{ nullptr }; + X509_NAME* certName{ nullptr }; + std::string subject = + std::string("mediasoup") + std::to_string(rand() % 999999 + 100000); + + // Create key with curve. + ecKey = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1); + + if (!ecKey) + { + LOG_OPENSSL_ERROR("EC_KEY_new_by_curve_name() failed"); + + goto error; + } + + EC_KEY_set_asn1_flag(ecKey, OPENSSL_EC_NAMED_CURVE); + + // NOTE: This can take some time. + ret = EC_KEY_generate_key(ecKey); + + if (ret == 0) + { + LOG_OPENSSL_ERROR("EC_KEY_generate_key() failed"); + + goto error; + } + + // Create a private key object. + DtlsTransport::privateKey = EVP_PKEY_new(); + + if (!DtlsTransport::privateKey) + { + LOG_OPENSSL_ERROR("EVP_PKEY_new() failed"); + + goto error; + } + + // NOLINTNEXTLINE(cppcoreguidelines-pro-type-cstyle-cast) + ret = EVP_PKEY_assign_EC_KEY(DtlsTransport::privateKey, ecKey); + + if (ret == 0) + { + LOG_OPENSSL_ERROR("EVP_PKEY_assign_EC_KEY() failed"); + + goto error; + } + + // The EC key now belongs to the private key, so don't clean it up separately. + ecKey = nullptr; + + // Create the X509 certificate. + DtlsTransport::certificate = X509_new(); + + if (!DtlsTransport::certificate) + { + LOG_OPENSSL_ERROR("X509_new() failed"); + + goto error; + } + + // Set version 3 (note that 0 means version 1). + X509_set_version(DtlsTransport::certificate, 2); + + // Set serial number (avoid default 0). + ASN1_INTEGER_set( + X509_get_serialNumber(DtlsTransport::certificate), + static_cast(rand() % 999999 + 100000)); + + // Set valid period. + X509_gmtime_adj(X509_get_notBefore(DtlsTransport::certificate), -315360000); // -10 years. + X509_gmtime_adj(X509_get_notAfter(DtlsTransport::certificate), 315360000); // 10 years. + + // Set the public key for the certificate using the key. + ret = X509_set_pubkey(DtlsTransport::certificate, DtlsTransport::privateKey); + + if (ret == 0) + { + LOG_OPENSSL_ERROR("X509_set_pubkey() failed"); + + goto error; + } + + // Set certificate fields. + certName = X509_get_subject_name(DtlsTransport::certificate); + + if (!certName) + { + LOG_OPENSSL_ERROR("X509_get_subject_name() failed"); + + goto error; + } + + X509_NAME_add_entry_by_txt( + certName, "O", MBSTRING_ASC, reinterpret_cast(subject.c_str()), -1, -1, 0); + X509_NAME_add_entry_by_txt( + certName, "CN", MBSTRING_ASC, reinterpret_cast(subject.c_str()), -1, -1, 0); + + // It is self-signed so set the issuer name to be the same as the subject. + ret = X509_set_issuer_name(DtlsTransport::certificate, certName); + + if (ret == 0) + { + LOG_OPENSSL_ERROR("X509_set_issuer_name() failed"); + + goto error; + } + + // Sign the certificate with its own private key. + ret = X509_sign(DtlsTransport::certificate, DtlsTransport::privateKey, EVP_sha1()); + + if (ret == 0) + { + LOG_OPENSSL_ERROR("X509_sign() failed"); + + goto error; + } + + return; + + error: + + if (ecKey) + EC_KEY_free(ecKey); + + if (DtlsTransport::privateKey) + EVP_PKEY_free(DtlsTransport::privateKey); // NOTE: This also frees the EC key. + + if (DtlsTransport::certificate) + X509_free(DtlsTransport::certificate); + + MS_THROW_ERROR("DTLS certificate and private key generation failed"); + } + + void DtlsTransport::ReadCertificateAndPrivateKeyFromFiles() + { #if 0 - // Generate a X509 certificate and private key (unless PEM files are provided). - if (Settings::configuration.dtlsCertificateFile.empty() || - Settings::configuration.dtlsPrivateKeyFile.empty()) { - GenerateCertificateAndPrivateKey(); - } else { - ReadCertificateAndPrivateKeyFromFiles(); - } -#else - GenerateCertificateAndPrivateKey(); + MS_TRACE(); + + FILE* file{ nullptr }; + + file = fopen(Settings::configuration.dtlsCertificateFile.c_str(), "r"); + + if (!file) + { + MS_ERROR("error reading DTLS certificate file: %s", std::strerror(errno)); + + goto error; + } + + DtlsTransport::certificate = PEM_read_X509(file, nullptr, nullptr, nullptr); + + if (!DtlsTransport::certificate) + { + LOG_OPENSSL_ERROR("PEM_read_X509() failed"); + + goto error; + } + + fclose(file); + + file = fopen(Settings::configuration.dtlsPrivateKeyFile.c_str(), "r"); + + if (!file) + { + MS_ERROR("error reading DTLS private key file: %s", std::strerror(errno)); + + goto error; + } + + DtlsTransport::privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr); + + if (!DtlsTransport::privateKey) + { + LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed"); + + goto error; + } + + fclose(file); + + return; + + error: + + MS_THROW_ERROR("error reading DTLS certificate and private key PEM files"); #endif + } - // Create a global SSL_CTX. - CreateSslCtx(); + void DtlsTransport::CreateSslCtx() + { + MS_TRACE(); - // Generate certificate fingerprints. - GenerateFingerprints(); -} + std::string dtlsSrtpCryptoSuites; + int ret; -void DtlsTransport::ClassDestroy() { - MS_TRACE(); + /* Set the global DTLS context. */ - if (DtlsTransport::privateKey) EVP_PKEY_free(DtlsTransport::privateKey); - if (DtlsTransport::certificate) X509_free(DtlsTransport::certificate); - if (DtlsTransport::sslCtx) SSL_CTX_free(DtlsTransport::sslCtx); -} + // Both DTLS 1.0 and 1.2 (requires OpenSSL >= 1.1.0). + DtlsTransport::sslCtx = SSL_CTX_new(DTLS_method()); -void DtlsTransport::GenerateCertificateAndPrivateKey() { - MS_TRACE(); + if (!DtlsTransport::sslCtx) + { + LOG_OPENSSL_ERROR("SSL_CTX_new() failed"); - int ret{0}; - EC_KEY* ecKey{nullptr}; - X509_NAME* certName{nullptr}; - std::string subject = std::string("mediasoup") + std::to_string(rand() % 999999 + 100000); - // std::string("mediasoup") + std::to_string(Utils::Crypto::GetRandomUInt(100000, 999999)); + goto error; + } - // Create key with curve. - ecKey = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1); + ret = SSL_CTX_use_certificate(DtlsTransport::sslCtx, DtlsTransport::certificate); - if (!ecKey) { - LOG_OPENSSL_ERROR("EC_KEY_new_by_curve_name() failed"); + if (ret == 0) + { + LOG_OPENSSL_ERROR("SSL_CTX_use_certificate() failed"); - goto error; - } + goto error; + } - EC_KEY_set_asn1_flag(ecKey, OPENSSL_EC_NAMED_CURVE); + ret = SSL_CTX_use_PrivateKey(DtlsTransport::sslCtx, DtlsTransport::privateKey); - // NOTE: This can take some time. - ret = EC_KEY_generate_key(ecKey); + if (ret == 0) + { + LOG_OPENSSL_ERROR("SSL_CTX_use_PrivateKey() failed"); - if (ret == 0) { - LOG_OPENSSL_ERROR("EC_KEY_generate_key() failed"); + goto error; + } - goto error; - } + ret = SSL_CTX_check_private_key(DtlsTransport::sslCtx); - // Create a private key object. - DtlsTransport::privateKey = EVP_PKEY_new(); + if (ret == 0) + { + LOG_OPENSSL_ERROR("SSL_CTX_check_private_key() failed"); - if (!DtlsTransport::privateKey) { - LOG_OPENSSL_ERROR("EVP_PKEY_new() failed"); + goto error; + } - goto error; - } + // Set options. + SSL_CTX_set_options( + DtlsTransport::sslCtx, + SSL_OP_CIPHER_SERVER_PREFERENCE | SSL_OP_NO_TICKET | SSL_OP_SINGLE_ECDH_USE | + SSL_OP_NO_QUERY_MTU); - // NOLINTNEXTLINE(cppcoreguidelines-pro-type-cstyle-cast) - ret = EVP_PKEY_assign_EC_KEY(DtlsTransport::privateKey, ecKey); + // Don't use sessions cache. + SSL_CTX_set_session_cache_mode(DtlsTransport::sslCtx, SSL_SESS_CACHE_OFF); - if (ret == 0) { - LOG_OPENSSL_ERROR("EVP_PKEY_assign_EC_KEY() failed"); + // Read always as much into the buffer as possible. + // NOTE: This is the default for DTLS, but a bug in non latest OpenSSL + // versions makes this call required. + SSL_CTX_set_read_ahead(DtlsTransport::sslCtx, 1); - goto error; - } + SSL_CTX_set_verify_depth(DtlsTransport::sslCtx, 4); - // The EC key now belongs to the private key, so don't clean it up separately. - ecKey = nullptr; + // Require certificate from peer. + SSL_CTX_set_verify( + DtlsTransport::sslCtx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, onSslCertificateVerify); - // Create the X509 certificate. - DtlsTransport::certificate = X509_new(); + // Set SSL info callback. + SSL_CTX_set_info_callback(DtlsTransport::sslCtx, [](const SSL* ssl, int where, int ret){ + static_cast(SSL_get_ex_data(ssl, 0))->OnSslInfo(where, ret); + }); + // Set ciphers. + ret = SSL_CTX_set_cipher_list( + DtlsTransport::sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); - if (!DtlsTransport::certificate) { - LOG_OPENSSL_ERROR("X509_new() failed"); + if (ret == 0) + { + LOG_OPENSSL_ERROR("SSL_CTX_set_cipher_list() failed"); - goto error; - } + goto error; + } - // Set version 3 (note that 0 means version 1). - X509_set_version(DtlsTransport::certificate, 2); + // Enable ECDH ciphers. + // DOC: http://en.wikibooks.org/wiki/OpenSSL/Diffie-Hellman_parameters + // NOTE: https://code.google.com/p/chromium/issues/detail?id=406458 + // NOTE: https://bugs.ruby-lang.org/issues/12324 - // Set serial number (avoid default 0). - // ASN1_INTEGER_set(X509_get_serialNumber(DtlsTransport::certificate), - // static_cast(Utils::Crypto::GetRandomUInt(1000000, 9999999))); - ASN1_INTEGER_set(X509_get_serialNumber(DtlsTransport::certificate), - static_cast(rand() % 999999 + 100000)); + // For OpenSSL >= 1.0.2. + SSL_CTX_set_ecdh_auto(DtlsTransport::sslCtx, 1); - // Set valid period. - X509_gmtime_adj(X509_get_notBefore(DtlsTransport::certificate), -315360000); // -10 years. - X509_gmtime_adj(X509_get_notAfter(DtlsTransport::certificate), 315360000); // 10 years. + // Set the "use_srtp" DTLS extension. + for (auto it = DtlsTransport::srtpCryptoSuites.begin(); + it != DtlsTransport::srtpCryptoSuites.end(); + ++it) + { + if (it != DtlsTransport::srtpCryptoSuites.begin()) + dtlsSrtpCryptoSuites += ":"; - // Set the public key for the certificate using the key. - ret = X509_set_pubkey(DtlsTransport::certificate, DtlsTransport::privateKey); + SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(*it); + dtlsSrtpCryptoSuites += cryptoSuiteEntry->name; + } - if (ret == 0) { - LOG_OPENSSL_ERROR("X509_set_pubkey() failed"); + MS_DEBUG_2TAGS(dtls, srtp, "setting SRTP cryptoSuites for DTLS: %s", dtlsSrtpCryptoSuites.c_str()); - goto error; - } + // NOTE: This function returns 0 on success. + ret = SSL_CTX_set_tlsext_use_srtp(DtlsTransport::sslCtx, dtlsSrtpCryptoSuites.c_str()); - // Set certificate fields. - certName = X509_get_subject_name(DtlsTransport::certificate); + if (ret != 0) + { + MS_ERROR( + "SSL_CTX_set_tlsext_use_srtp() failed when entering '%s'", dtlsSrtpCryptoSuites.c_str()); + LOG_OPENSSL_ERROR("SSL_CTX_set_tlsext_use_srtp() failed"); - if (!certName) { - LOG_OPENSSL_ERROR("X509_get_subject_name() failed"); + goto error; + } - goto error; - } + return; - X509_NAME_add_entry_by_txt(certName, "O", MBSTRING_ASC, - reinterpret_cast(subject.c_str()), -1, -1, 0); - X509_NAME_add_entry_by_txt(certName, "CN", MBSTRING_ASC, - reinterpret_cast(subject.c_str()), -1, -1, 0); + error: - // It is self-signed so set the issuer name to be the same as the subject. - ret = X509_set_issuer_name(DtlsTransport::certificate, certName); + if (DtlsTransport::sslCtx) + { + SSL_CTX_free(DtlsTransport::sslCtx); + DtlsTransport::sslCtx = nullptr; + } - if (ret == 0) { - LOG_OPENSSL_ERROR("X509_set_issuer_name() failed"); + MS_THROW_ERROR("SSL context creation failed"); + } - goto error; - } + void DtlsTransport::GenerateFingerprints() + { + MS_TRACE(); - // Sign the certificate with its own private key. - ret = X509_sign(DtlsTransport::certificate, DtlsTransport::privateKey, EVP_sha1()); + for (auto& kv : DtlsTransport::string2FingerprintAlgorithm) + { + const std::string& algorithmString = kv.first; + FingerprintAlgorithm algorithm = kv.second; + uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; + unsigned int size{ 0 }; + char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; + const EVP_MD* hashFunction; + int ret; - if (ret == 0) { - LOG_OPENSSL_ERROR("X509_sign() failed"); + switch (algorithm) + { + case FingerprintAlgorithm::SHA1: + hashFunction = EVP_sha1(); + break; - goto error; - } + case FingerprintAlgorithm::SHA224: + hashFunction = EVP_sha224(); + break; - return; + case FingerprintAlgorithm::SHA256: + hashFunction = EVP_sha256(); + break; -error: + case FingerprintAlgorithm::SHA384: + hashFunction = EVP_sha384(); + break; - if (ecKey) EC_KEY_free(ecKey); + case FingerprintAlgorithm::SHA512: + hashFunction = EVP_sha512(); + break; - if (DtlsTransport::privateKey) - EVP_PKEY_free(DtlsTransport::privateKey); // NOTE: This also frees the EC key. + default: + MS_THROW_ERROR("unknown algorithm"); + } - if (DtlsTransport::certificate) X509_free(DtlsTransport::certificate); + ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size); - MS_THROW_ERROR("DTLS certificate and private key generation failed"); -} + if (ret == 0) + { + MS_ERROR("X509_digest() failed"); + MS_THROW_ERROR("Fingerprints generation failed"); + } -void DtlsTransport::ReadCertificateAndPrivateKeyFromFiles() { + // Convert to hexadecimal format in uppercase with colons. + for (unsigned int i{ 0 }; i < size; ++i) + { + std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); + } + hexFingerprint[(size * 3) - 1] = '\0'; + + MS_DEBUG_TAG(dtls, "%-7s fingerprint: %s", algorithmString.c_str(), hexFingerprint); + + // Store it in the vector. + DtlsTransport::Fingerprint fingerprint; + + fingerprint.algorithm = DtlsTransport::GetFingerprintAlgorithm(algorithmString); + fingerprint.value = hexFingerprint; + + DtlsTransport::localFingerprints.push_back(fingerprint); + } + } + + /* Instance methods. */ + + DtlsTransport::DtlsTransport(EventPoller::Ptr poller,Listener* listener) : poller(std::move(poller)), listener(listener) + { + MS_TRACE(); + + /* Set SSL. */ + + this->ssl = SSL_new(DtlsTransport::sslCtx); + + if (!this->ssl) + { + LOG_OPENSSL_ERROR("SSL_new() failed"); + + goto error; + } + + // Set this as custom data. + SSL_set_ex_data(this->ssl, 0, static_cast(this)); + + this->sslBioFromNetwork = BIO_new(BIO_s_mem()); + + if (!this->sslBioFromNetwork) + { + LOG_OPENSSL_ERROR("BIO_new() failed"); + + SSL_free(this->ssl); + + goto error; + } + + this->sslBioToNetwork = BIO_new(BIO_s_mem()); + + if (!this->sslBioToNetwork) + { + LOG_OPENSSL_ERROR("BIO_new() failed"); + + BIO_free(this->sslBioFromNetwork); + SSL_free(this->ssl); + + goto error; + } + + SSL_set_bio(this->ssl, this->sslBioFromNetwork, this->sslBioToNetwork); + + // Set the MTU so that we don't send packets that are too large with no fragmentation. + SSL_set_mtu(this->ssl, DtlsMtu); + DTLS_set_link_mtu(this->ssl, DtlsMtu); + + // Set callback handler for setting DTLS timer interval. + DTLS_set_timer_cb(this->ssl, onSslDtlsTimer); + + return; + + error: + + // NOTE: At this point SSL_set_bio() was not called so we must free BIOs as + // well. + if (this->sslBioFromNetwork) + BIO_free(this->sslBioFromNetwork); + + if (this->sslBioToNetwork) + BIO_free(this->sslBioToNetwork); + + if (this->ssl) + SSL_free(this->ssl); + + // NOTE: If this is not catched by the caller the program will abort, but + // this should never happen. + MS_THROW_ERROR("DtlsTransport instance creation failed"); + } + + DtlsTransport::~DtlsTransport() + { + MS_TRACE(); + + if (IsRunning()) + { + // Send close alert to the peer. + SSL_shutdown(this->ssl); + SendPendingOutgoingDtlsData(); + } + + if (this->ssl) + { + SSL_free(this->ssl); + + this->ssl = nullptr; + this->sslBioFromNetwork = nullptr; + this->sslBioToNetwork = nullptr; + } + + // Close the DTLS timer. + this->timer = nullptr; + } + + void DtlsTransport::Dump() const + { + MS_TRACE(); + + std::string state{ "new" }; + std::string role{ "none " }; + + switch (this->state) + { + case DtlsState::CONNECTING: + state = "connecting"; + break; + case DtlsState::CONNECTED: + state = "connected"; + break; + case DtlsState::FAILED: + state = "failed"; + break; + case DtlsState::CLOSED: + state = "closed"; + break; + default:; + } + + switch (this->localRole) + { + case Role::AUTO: + role = "auto"; + break; + case Role::SERVER: + role = "server"; + break; + case Role::CLIENT: + role = "client"; + break; + default:; + } + + MS_DUMP(""); + MS_DUMP(" state : %s", state.c_str()); + MS_DUMP(" role : %s", role.c_str()); + MS_DUMP(" handshake done: : %s", this->handshakeDone ? "yes" : "no"); + MS_DUMP(""); + } + + void DtlsTransport::Run(Role localRole) + { + MS_TRACE(); + + MS_ASSERT( + localRole == Role::CLIENT || localRole == Role::SERVER, + "local DTLS role must be 'client' or 'server'"); + + Role previousLocalRole = this->localRole; + + if (localRole == previousLocalRole) + { + MS_ERROR("same local DTLS role provided, doing nothing"); + + return; + } + + // If the previous local DTLS role was 'client' or 'server' do reset. + if (previousLocalRole == Role::CLIENT || previousLocalRole == Role::SERVER) + { + MS_DEBUG_TAG(dtls, "resetting DTLS due to local role change"); + + Reset(); + } + + // Update local role. + this->localRole = localRole; + + // Set state and notify the listener. + this->state = DtlsState::CONNECTING; + this->listener->OnDtlsTransportConnecting(this); + + switch (this->localRole) + { + case Role::CLIENT: + { + MS_DEBUG_TAG(dtls, "running [role:client]"); + + SSL_set_connect_state(this->ssl); + SSL_do_handshake(this->ssl); + SendPendingOutgoingDtlsData(); + SetTimeout(); + + break; + } + + case Role::SERVER: + { + MS_DEBUG_TAG(dtls, "running [role:server]"); + + SSL_set_accept_state(this->ssl); + SSL_do_handshake(this->ssl); + + break; + } + + default: + { + MS_ABORT("invalid local DTLS role"); + } + } + } + + bool DtlsTransport::SetRemoteFingerprint(Fingerprint fingerprint) + { + MS_TRACE(); + + MS_ASSERT( + fingerprint.algorithm != FingerprintAlgorithm::NONE, "no fingerprint algorithm provided"); + + this->remoteFingerprint = fingerprint; + + // The remote fingerpring may have been set after DTLS handshake was done, + // so we may need to process it now. + if (this->handshakeDone && this->state != DtlsState::CONNECTED) + { + MS_DEBUG_TAG(dtls, "handshake already done, processing it right now"); + + return ProcessHandshake(); + } + + return true; + } + + void DtlsTransport::ProcessDtlsData(const uint8_t* data, size_t len) + { + MS_TRACE(); + + int written; + int read; + + if (!IsRunning()) + { + MS_ERROR("cannot process data while not running"); + + return; + } + + // Write the received DTLS data into the sslBioFromNetwork. + written = + BIO_write(this->sslBioFromNetwork, static_cast(data), static_cast(len)); + + if (written != static_cast(len)) + { + MS_WARN_TAG( + dtls, + "OpenSSL BIO_write() wrote less (%zu bytes) than given data (%zu bytes)", + static_cast(written), + len); + } + + // Must call SSL_read() to process received DTLS data. + read = SSL_read(this->ssl, static_cast(DtlsTransport::sslReadBuffer), SslReadBufferSize); + + // Send data if it's ready. + SendPendingOutgoingDtlsData(); + + // Check SSL status and return if it is bad/closed. + if (!CheckStatus(read)) + return; + + // Set/update the DTLS timeout. + if (!SetTimeout()) + return; + + // Application data received. Notify to the listener. + if (read > 0) + { + // It is allowed to receive DTLS data even before validating remote fingerprint. + if (!this->handshakeDone) + { + MS_WARN_TAG(dtls, "ignoring application data received while DTLS handshake not done"); + + return; + } + + // Notify the listener. + this->listener->OnDtlsTransportApplicationDataReceived( + this, (uint8_t*)DtlsTransport::sslReadBuffer, static_cast(read)); + } + } + + void DtlsTransport::SendApplicationData(const uint8_t* data, size_t len) + { + MS_TRACE(); + + // We cannot send data to the peer if its remote fingerprint is not validated. + if (this->state != DtlsState::CONNECTED) + { + MS_WARN_TAG(dtls, "cannot send application data while DTLS is not fully connected"); + + return; + } + + if (len == 0) + { + MS_WARN_TAG(dtls, "ignoring 0 length data"); + + return; + } + + int written; + + written = SSL_write(this->ssl, static_cast(data), static_cast(len)); + + if (written < 0) + { + LOG_OPENSSL_ERROR("SSL_write() failed"); + + if (!CheckStatus(written)) + return; + } + else if (written != static_cast(len)) + { + MS_WARN_TAG( + dtls, "OpenSSL SSL_write() wrote less (%d bytes) than given data (%zu bytes)", written, len); + } + + // Send data. + SendPendingOutgoingDtlsData(); + } + + void DtlsTransport::Reset() + { + MS_TRACE(); + + int ret; + + if (!IsRunning()) + return; + + MS_WARN_TAG(dtls, "resetting DTLS transport"); + + // Stop the DTLS timer. + this->timer = nullptr; + + // We need to reset the SSL instance so we need to "shutdown" it, but we + // don't want to send a Close Alert to the peer, so just don't call + // SendPendingOutgoingDTLSData(). + SSL_shutdown(this->ssl); + + this->localRole = Role::NONE; + this->state = DtlsState::NEW; + this->handshakeDone = false; + this->handshakeDoneNow = false; + + // Reset SSL status. + // NOTE: For this to properly work, SSL_shutdown() must be called before. + // NOTE: This may fail if not enough DTLS handshake data has been received, + // but we don't care so just clear the error queue. + ret = SSL_clear(this->ssl); + + if (ret == 0) + ERR_clear_error(); + } + + inline bool DtlsTransport::CheckStatus(int returnCode) + { + MS_TRACE(); + + int err; + bool wasHandshakeDone = this->handshakeDone; + + err = SSL_get_error(this->ssl, returnCode); + + switch (err) + { + case SSL_ERROR_NONE: + break; + + case SSL_ERROR_SSL: + LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SSL"); + break; + + case SSL_ERROR_WANT_READ: + break; + + case SSL_ERROR_WANT_WRITE: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_WRITE"); + break; + + case SSL_ERROR_WANT_X509_LOOKUP: + MS_DEBUG_TAG(dtls, "SSL status: SSL_ERROR_WANT_X509_LOOKUP"); + break; + + case SSL_ERROR_SYSCALL: + LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SYSCALL"); + break; + + case SSL_ERROR_ZERO_RETURN: + break; + + case SSL_ERROR_WANT_CONNECT: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_CONNECT"); + break; + + case SSL_ERROR_WANT_ACCEPT: + MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_ACCEPT"); + break; + + default: + MS_WARN_TAG(dtls, "SSL status: unknown error"); + } + + // Check if the handshake (or re-handshake) has been done right now. + if (this->handshakeDoneNow) + { + this->handshakeDoneNow = false; + this->handshakeDone = true; + + // Stop the timer. + this->timer = nullptr; + + // Process the handshake just once (ignore if DTLS renegotiation). + if (!wasHandshakeDone && this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE) + return ProcessHandshake(); + + return true; + } + // Check if the peer sent close alert or a fatal error happened. + else if (((SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN) != 0) || err == SSL_ERROR_SSL || err == SSL_ERROR_SYSCALL) + { + if (this->state == DtlsState::CONNECTED) + { + MS_DEBUG_TAG(dtls, "disconnected"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::CLOSED; + this->listener->OnDtlsTransportClosed(this); + } + else + { + MS_WARN_TAG(dtls, "connection failed"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + } + + return false; + } + else + { + return true; + } + } + + inline void DtlsTransport::SendPendingOutgoingDtlsData() + { + MS_TRACE(); + + if (BIO_eof(this->sslBioToNetwork)) + return; + + int64_t read; + char* data{ nullptr }; + + read = BIO_get_mem_data(this->sslBioToNetwork, &data); // NOLINT + + if (read <= 0) + return; + + MS_DEBUG_DEV("%" PRIu64 " bytes of DTLS data ready to sent to the peer", read); + + // Notify the listener. + this->listener->OnDtlsTransportSendData( + this, reinterpret_cast(data), static_cast(read)); + + // Clear the BIO buffer. + // NOTE: the (void) avoids the -Wunused-value warning. + (void)BIO_reset(this->sslBioToNetwork); + } + + inline bool DtlsTransport::SetTimeout() + { + MS_TRACE(); + + MS_ASSERT( + this->state == DtlsState::CONNECTING || this->state == DtlsState::CONNECTED, + "invalid DTLS state"); + + int64_t ret; + struct timeval dtlsTimeout{ 0, 0 }; + uint64_t timeoutMs; + + // NOTE: If ret == 0 then ignore the value in dtlsTimeout. + // NOTE: No DTLSv_1_2_get_timeout() or DTLS_get_timeout() in OpenSSL 1.1.0-dev. + ret = DTLSv1_get_timeout(this->ssl, static_cast(&dtlsTimeout)); // NOLINT + + if (ret == 0) + return true; + + timeoutMs = (dtlsTimeout.tv_sec * static_cast(1000)) + (dtlsTimeout.tv_usec / 1000); + + if (timeoutMs == 0) + { + return true; + } + else if (timeoutMs < 30000) + { + MS_DEBUG_DEV("DTLS timer set in %" PRIu64 "ms", timeoutMs); + + weak_ptr weak_self = shared_from_this(); + this->timer = std::make_shared(timeoutMs / 1000.0f, [weak_self](){ + auto strong_self = weak_self.lock(); + if(strong_self){ + strong_self->OnTimer(); + } + return true; + }, this->poller); + + return true; + } + // NOTE: Don't start the timer again if the timeout is greater than 30 seconds. + else + { + MS_WARN_TAG(dtls, "DTLS timeout too high (%" PRIu64 "ms), resetting DLTS", timeoutMs); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + + return false; + } + } + + inline bool DtlsTransport::ProcessHandshake() + { + MS_TRACE(); + + MS_ASSERT(this->handshakeDone, "handshake not done yet"); + MS_ASSERT( + this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, "remote fingerprint not set"); + + // Validate the remote fingerprint. + if (!CheckRemoteFingerprint()) + { + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + + return false; + } + + // Get the negotiated SRTP crypto suite. + RTC::SrtpSession::CryptoSuite srtpCryptoSuite = GetNegotiatedSrtpCryptoSuite(); + + if (srtpCryptoSuite != RTC::SrtpSession::CryptoSuite::NONE) + { + // Extract the SRTP keys (will notify the listener with them). + ExtractSrtpKeys(srtpCryptoSuite); + + return true; + } + + // NOTE: We assume that "use_srtp" DTLS extension is required even if + // there is no audio/video. + MS_WARN_2TAGS(dtls, srtp, "SRTP crypto suite not negotiated"); + + Reset(); + + // Set state and notify the listener. + this->state = DtlsState::FAILED; + this->listener->OnDtlsTransportFailed(this); + + return false; + } + + inline bool DtlsTransport::CheckRemoteFingerprint() + { + MS_TRACE(); + + MS_ASSERT( + this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, "remote fingerprint not set"); + + X509* certificate; + uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; + unsigned int size{ 0 }; + char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; + const EVP_MD* hashFunction; + int ret; + + certificate = SSL_get_peer_certificate(this->ssl); + + if (!certificate) + { + MS_WARN_TAG(dtls, "no certificate was provided by the peer"); + + return false; + } + + switch (this->remoteFingerprint.algorithm) + { + case FingerprintAlgorithm::SHA1: + hashFunction = EVP_sha1(); + break; + + case FingerprintAlgorithm::SHA224: + hashFunction = EVP_sha224(); + break; + + case FingerprintAlgorithm::SHA256: + hashFunction = EVP_sha256(); + break; + + case FingerprintAlgorithm::SHA384: + hashFunction = EVP_sha384(); + break; + + case FingerprintAlgorithm::SHA512: + hashFunction = EVP_sha512(); + break; + + default: + MS_ABORT("unknown algorithm"); + } + + // Compare the remote fingerprint with the value given via signaling. + ret = X509_digest(certificate, hashFunction, binaryFingerprint, &size); + + if (ret == 0) + { + MS_ERROR("X509_digest() failed"); + + X509_free(certificate); + + return false; + } + + // Convert to hexadecimal format in uppercase with colons. + for (unsigned int i{ 0 }; i < size; ++i) + { + std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); + } + hexFingerprint[(size * 3) - 1] = '\0'; + + if (this->remoteFingerprint.value != hexFingerprint) + { + MS_WARN_TAG( + dtls, + "fingerprint in the remote certificate (%s) does not match the announced one (%s)", + hexFingerprint, + this->remoteFingerprint.value.c_str()); + + //todo 先屏蔽检查客户端签名 #if 0 - MS_TRACE(); + X509_free(certificate); - FILE* file{nullptr}; - - file = fopen(Settings::configuration.dtlsCertificateFile.c_str(), "r"); - - if (!file) { - MS_ERROR("error reading DTLS certificate file: %s", std::strerror(errno)); - - goto error; - } - - DtlsTransport::certificate = PEM_read_X509(file, nullptr, nullptr, nullptr); - - if (!DtlsTransport::certificate) { - LOG_OPENSSL_ERROR("PEM_read_X509() failed"); - - goto error; - } - - fclose(file); - - file = fopen(Settings::configuration.dtlsPrivateKeyFile.c_str(), "r"); - - if (!file) { - MS_ERROR("error reading DTLS private key file: %s", std::strerror(errno)); - - goto error; - } - - DtlsTransport::privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr); - - if (!DtlsTransport::privateKey) { - LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed"); - - goto error; - } - - fclose(file); - - return; - - error: - - MS_THROW_ERROR("error reading DTLS certificate and private key PEM files"); + return false; #endif -} + } -void DtlsTransport::CreateSslCtx() { - MS_TRACE(); + MS_DEBUG_TAG(dtls, "valid remote fingerprint"); - std::string dtlsSrtpCryptoSuites; - int ret; + // Get the remote certificate in PEM format. - /* Set the global DTLS context. */ + BIO* bio = BIO_new(BIO_s_mem()); - // Both DTLS 1.0 and 1.2 (requires OpenSSL >= 1.1.0). - DtlsTransport::sslCtx = SSL_CTX_new(DTLS_method()); + // Ensure the underlying BUF_MEM structure is also freed. + // NOTE: Avoid stupid "warning: value computed is not used [-Wunused-value]" since + // BIO_set_close() always returns 1. + (void)BIO_set_close(bio, BIO_CLOSE); - if (!DtlsTransport::sslCtx) { - LOG_OPENSSL_ERROR("SSL_CTX_new() failed"); + ret = PEM_write_bio_X509(bio, certificate); - goto error; - } + if (ret != 1) + { + LOG_OPENSSL_ERROR("PEM_write_bio_X509() failed"); - ret = SSL_CTX_use_certificate(DtlsTransport::sslCtx, DtlsTransport::certificate); + X509_free(certificate); + BIO_free(bio); - if (ret == 0) { - LOG_OPENSSL_ERROR("SSL_CTX_use_certificate() failed"); + return false; + } - goto error; - } + BUF_MEM* mem; - ret = SSL_CTX_use_PrivateKey(DtlsTransport::sslCtx, DtlsTransport::privateKey); + BIO_get_mem_ptr(bio, &mem); // NOLINT[cppcoreguidelines-pro-type-cstyle-cast] - if (ret == 0) { - LOG_OPENSSL_ERROR("SSL_CTX_use_PrivateKey() failed"); + if (!mem || !mem->data || mem->length == 0u) + { + LOG_OPENSSL_ERROR("BIO_get_mem_ptr() failed"); - goto error; - } + X509_free(certificate); + BIO_free(bio); - ret = SSL_CTX_check_private_key(DtlsTransport::sslCtx); + return false; + } - if (ret == 0) { - LOG_OPENSSL_ERROR("SSL_CTX_check_private_key() failed"); + this->remoteCert = std::string(mem->data, mem->length); - goto error; - } - - // Set options. - SSL_CTX_set_options(DtlsTransport::sslCtx, SSL_OP_CIPHER_SERVER_PREFERENCE | SSL_OP_NO_TICKET | - SSL_OP_SINGLE_ECDH_USE | SSL_OP_NO_QUERY_MTU); - - // Don't use sessions cache. - SSL_CTX_set_session_cache_mode(DtlsTransport::sslCtx, SSL_SESS_CACHE_OFF); - - // Read always as much into the buffer as possible. - // NOTE: This is the default for DTLS, but a bug in non latest OpenSSL - // versions makes this call required. - SSL_CTX_set_read_ahead(DtlsTransport::sslCtx, 1); - - SSL_CTX_set_verify_depth(DtlsTransport::sslCtx, 4); - - // Require certificate from peer. - SSL_CTX_set_verify(DtlsTransport::sslCtx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, - onSslCertificateVerify); - - // Set SSL info callback. - SSL_CTX_set_info_callback(DtlsTransport::sslCtx, onSslInfo); - - // Set ciphers. - ret = SSL_CTX_set_cipher_list(DtlsTransport::sslCtx, - "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); - - if (ret == 0) { - LOG_OPENSSL_ERROR("SSL_CTX_set_cipher_list() failed"); - - goto error; - } - - // Enable ECDH ciphers. - // DOC: http://en.wikibooks.org/wiki/OpenSSL/Diffie-Hellman_parameters - // NOTE: https://code.google.com/p/chromium/issues/detail?id=406458 - // NOTE: https://bugs.ruby-lang.org/issues/12324 - - // For OpenSSL >= 1.0.2. - SSL_CTX_set_ecdh_auto(DtlsTransport::sslCtx, 1); - - // Set the "use_srtp" DTLS extension. - for (auto it = DtlsTransport::srtpCryptoSuites.begin(); - it != DtlsTransport::srtpCryptoSuites.end(); ++it) { - if (it != DtlsTransport::srtpCryptoSuites.begin()) dtlsSrtpCryptoSuites += ":"; - - SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(*it); - dtlsSrtpCryptoSuites += cryptoSuiteEntry->name; - } - - MS_DEBUG_2TAGS(dtls, srtp, "setting SRTP cryptoSuites for DTLS: %s", - dtlsSrtpCryptoSuites.c_str()); - - // NOTE: This function returns 0 on success. - ret = SSL_CTX_set_tlsext_use_srtp(DtlsTransport::sslCtx, dtlsSrtpCryptoSuites.c_str()); - - if (ret != 0) { - MS_ERROR("SSL_CTX_set_tlsext_use_srtp() failed when entering '%s'", - dtlsSrtpCryptoSuites.c_str()); - LOG_OPENSSL_ERROR("SSL_CTX_set_tlsext_use_srtp() failed"); - - goto error; - } - - return; - -error: - - if (DtlsTransport::sslCtx) { - SSL_CTX_free(DtlsTransport::sslCtx); - DtlsTransport::sslCtx = nullptr; - } - - MS_THROW_ERROR("SSL context creation failed"); -} - -void DtlsTransport::GenerateFingerprints() { - MS_TRACE(); - - for (auto& kv : DtlsTransport::string2FingerprintAlgorithm) { - const std::string& algorithmString = kv.first; - FingerprintAlgorithm algorithm = kv.second; - uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; - unsigned int size{0}; - char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; - const EVP_MD* hashFunction; - int ret; - - switch (algorithm) { - case FingerprintAlgorithm::SHA1: - hashFunction = EVP_sha1(); - break; - - case FingerprintAlgorithm::SHA224: - hashFunction = EVP_sha224(); - break; - - case FingerprintAlgorithm::SHA256: - hashFunction = EVP_sha256(); - break; - - case FingerprintAlgorithm::SHA384: - hashFunction = EVP_sha384(); - break; - - case FingerprintAlgorithm::SHA512: - hashFunction = EVP_sha512(); - break; - - default: - MS_THROW_ERROR("unknown algorithm"); - } - - ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size); - - if (ret == 0) { - MS_ERROR("X509_digest() failed"); - MS_THROW_ERROR("Fingerprints generation failed"); - } - - // Convert to hexadecimal format in uppercase with colons. - for (unsigned int i{0}; i < size; ++i) { - std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); - } - hexFingerprint[(size * 3) - 1] = '\0'; - - MS_DEBUG_TAG(dtls, "%-7s fingerprint: %s", algorithmString.c_str(), hexFingerprint); - // Store it in the vector. - DtlsTransport::Fingerprint fingerprint; - - fingerprint.algorithm = DtlsTransport::GetFingerprintAlgorithm(algorithmString); - fingerprint.value = hexFingerprint; - - DtlsTransport::localFingerprints.push_back(fingerprint); - } -} - -/* Instance methods. */ - -DtlsTransport::DtlsTransport(Listener* listener) : listener(listener) { - MS_TRACE(); - - /* Set SSL. */ - - this->ssl = SSL_new(DtlsTransport::sslCtx); - - if (!this->ssl) { - LOG_OPENSSL_ERROR("SSL_new() failed"); - - goto error; - } - - // Set this as custom data. - SSL_set_ex_data(this->ssl, 0, static_cast(this)); - - this->sslBioFromNetwork = BIO_new(BIO_s_mem()); - - if (!this->sslBioFromNetwork) { - LOG_OPENSSL_ERROR("BIO_new() failed"); - - SSL_free(this->ssl); - - goto error; - } - - this->sslBioToNetwork = BIO_new(BIO_s_mem()); - - if (!this->sslBioToNetwork) { - LOG_OPENSSL_ERROR("BIO_new() failed"); - - BIO_free(this->sslBioFromNetwork); - SSL_free(this->ssl); - - goto error; - } - - SSL_set_bio(this->ssl, this->sslBioFromNetwork, this->sslBioToNetwork); - - // Set the MTU so that we don't send packets that are too large with no fragmentation. - SSL_set_mtu(this->ssl, DtlsMtu); - DTLS_set_link_mtu(this->ssl, DtlsMtu); - - // Set callback handler for setting DTLS timer interval. - DTLS_set_timer_cb(this->ssl, onSslDtlsTimer); - - // Set the DTLS timer. - // this->timer = new Timer(this); - - return; - -error: - - // NOTE: At this point SSL_set_bio() was not called so we must free BIOs as - // well. - if (this->sslBioFromNetwork) BIO_free(this->sslBioFromNetwork); - - if (this->sslBioToNetwork) BIO_free(this->sslBioToNetwork); - - if (this->ssl) SSL_free(this->ssl); - - // NOTE: If this is not catched by the caller the program will abort, but - // this should never happen. - MS_THROW_ERROR("DtlsTransport instance creation failed"); -} - -DtlsTransport::~DtlsTransport() { - MS_TRACE(); - - if (IsRunning()) { - // Send close alert to the peer. - SSL_shutdown(this->ssl); - SendPendingOutgoingDtlsData(); - } - - if (this->ssl) { - SSL_free(this->ssl); - - this->ssl = nullptr; - this->sslBioFromNetwork = nullptr; - this->sslBioToNetwork = nullptr; - } - - // Close the DTLS timer. - // delete this->timer; -} - -void DtlsTransport::Dump() const { - MS_TRACE(); - - std::string state{"new"}; - std::string role{"none "}; - - switch (this->state) { - case DtlsState::CONNECTING: - state = "connecting"; - break; - case DtlsState::CONNECTED: - state = "connected"; - break; - case DtlsState::FAILED: - state = "failed"; - break; - case DtlsState::CLOSED: - state = "closed"; - break; - default:; - } - - switch (this->localRole) { - case Role::AUTO: - role = "auto"; - break; - case Role::SERVER: - role = "server"; - break; - case Role::CLIENT: - role = "client"; - break; - default:; - } - - MS_DUMP(""); - MS_DUMP(" state : %s", state.c_str()); - MS_DUMP(" role : %s", role.c_str()); - MS_DUMP(" handshake done: : %s", this->handshakeDone ? "yes" : "no"); - MS_DUMP(""); -} - -void DtlsTransport::Run(Role localRole) { - MS_TRACE(); - - MS_ASSERT(localRole == Role::CLIENT || localRole == Role::SERVER, - "local DTLS role must be 'client' or 'server'"); - - Role previousLocalRole = this->localRole; - - if (localRole == previousLocalRole) { - MS_ERROR("same local DTLS role provided, doing nothing"); - - return; - } - - // If the previous local DTLS role was 'client' or 'server' do reset. - if (previousLocalRole == Role::CLIENT || previousLocalRole == Role::SERVER) { - MS_DEBUG_TAG(dtls, "resetting DTLS due to local role change"); - - Reset(); - } - - // Update local role. - this->localRole = localRole; - - // Set state and notify the listener. - this->state = DtlsState::CONNECTING; - this->listener->OnDtlsTransportConnecting(this); - - switch (this->localRole) { - case Role::CLIENT: { - MS_DEBUG_TAG(dtls, "running [role:client]"); - - SSL_set_connect_state(this->ssl); - SSL_do_handshake(this->ssl); - SendPendingOutgoingDtlsData(); - SetTimeout(); - - break; - } - - case Role::SERVER: { - MS_DEBUG_TAG(dtls, "running [role:server]"); - - SSL_set_accept_state(this->ssl); - SSL_do_handshake(this->ssl); - - break; - } - - default: { - MS_ABORT("invalid local DTLS role"); - } - } -} - -bool DtlsTransport::SetRemoteFingerprint(Fingerprint fingerprint) { - MS_TRACE(); - - MS_ASSERT(fingerprint.algorithm != FingerprintAlgorithm::NONE, - "no fingerprint algorithm provided"); - - this->remoteFingerprint = fingerprint; - - // The remote fingerpring may have been set after DTLS handshake was done, - // so we may need to process it now. - if (this->handshakeDone && this->state != DtlsState::CONNECTED) { - MS_DEBUG_TAG(dtls, "handshake already done, processing it right now"); - - return ProcessHandshake(); - } - - return true; -} - -void DtlsTransport::ProcessDtlsData(const uint8_t* data, size_t len) { - MS_TRACE(); - - int written; - int read; - - if (!IsRunning()) { - MS_ERROR("cannot process data while not running"); - - return; - } - - // Write the received DTLS data into the sslBioFromNetwork. - written = - BIO_write(this->sslBioFromNetwork, static_cast(data), static_cast(len)); - - if (written != static_cast(len)) { - MS_WARN_TAG(dtls, "OpenSSL BIO_write() wrote less (%zu bytes) than given data (%zu bytes)", - static_cast(written), len); - } - - // Must call SSL_read() to process received DTLS data. - read = SSL_read(this->ssl, static_cast(DtlsTransport::sslReadBuffer), SslReadBufferSize); - - // Send data if it's ready. - SendPendingOutgoingDtlsData(); - - // Check SSL status and return if it is bad/closed. - if (!CheckStatus(read)) return; - - // Set/update the DTLS timeout. - if (!SetTimeout()) return; - - // Application data received. Notify to the listener. - if (read > 0) { - // It is allowed to receive DTLS data even before validating remote fingerprint. - if (!this->handshakeDone) { - MS_WARN_TAG(dtls, "ignoring application data received while DTLS handshake not done"); - - return; - } - - // Notify the listener. - this->listener->OnDtlsTransportApplicationDataReceived( - this, (uint8_t*)DtlsTransport::sslReadBuffer, static_cast(read)); - } -} - -void DtlsTransport::SendApplicationData(const uint8_t* data, size_t len) { - MS_TRACE(); - - // We cannot send data to the peer if its remote fingerprint is not validated. - if (this->state != DtlsState::CONNECTED) { - MS_WARN_TAG(dtls, "cannot send application data while DTLS is not fully connected"); - - return; - } - - if (len == 0) { - MS_WARN_TAG(dtls, "ignoring 0 length data"); - - return; - } - - int written; - - written = SSL_write(this->ssl, static_cast(data), static_cast(len)); - - if (written < 0) { - LOG_OPENSSL_ERROR("SSL_write() failed"); - - if (!CheckStatus(written)) return; - } else if (written != static_cast(len)) { - MS_WARN_TAG(dtls, "OpenSSL SSL_write() wrote less (%d bytes) than given data (%zu bytes)", - written, len); - } - - // Send data. - SendPendingOutgoingDtlsData(); -} - -void DtlsTransport::Reset() { - MS_TRACE(); - - int ret; - - if (!IsRunning()) return; - - MS_WARN_TAG(dtls, "resetting DTLS transport"); - - // Stop the DTLS timer. - // this->timer->Stop(); - - // We need to reset the SSL instance so we need to "shutdown" it, but we - // don't want to send a Close Alert to the peer, so just don't call - // SendPendingOutgoingDTLSData(). - SSL_shutdown(this->ssl); - - this->localRole = Role::NONE; - this->state = DtlsState::NEW; - this->handshakeDone = false; - this->handshakeDoneNow = false; - - // Reset SSL status. - // NOTE: For this to properly work, SSL_shutdown() must be called before. - // NOTE: This may fail if not enough DTLS handshake data has been received, - // but we don't care so just clear the error queue. - ret = SSL_clear(this->ssl); - - if (ret == 0) ERR_clear_error(); -} - -inline bool DtlsTransport::CheckStatus(int returnCode) { - MS_TRACE(); - - int err; - bool wasHandshakeDone = this->handshakeDone; - - err = SSL_get_error(this->ssl, returnCode); - - switch (err) { - case SSL_ERROR_NONE: - break; - - case SSL_ERROR_SSL: - LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SSL"); - break; - - case SSL_ERROR_WANT_READ: - break; - - case SSL_ERROR_WANT_WRITE: - MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_WRITE"); - break; - - case SSL_ERROR_WANT_X509_LOOKUP: - MS_DEBUG_TAG(dtls, "SSL status: SSL_ERROR_WANT_X509_LOOKUP"); - break; - - case SSL_ERROR_SYSCALL: - LOG_OPENSSL_ERROR("SSL status: SSL_ERROR_SYSCALL"); - break; - - case SSL_ERROR_ZERO_RETURN: - break; - - case SSL_ERROR_WANT_CONNECT: - MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_CONNECT"); - break; - - case SSL_ERROR_WANT_ACCEPT: - MS_WARN_TAG(dtls, "SSL status: SSL_ERROR_WANT_ACCEPT"); - break; - - default: - MS_WARN_TAG(dtls, "SSL status: unknown error"); - } - - // Check if the handshake (or re-handshake) has been done right now. - if (this->handshakeDoneNow) { - this->handshakeDoneNow = false; - this->handshakeDone = true; - - // Stop the timer. - // this->timer->Stop(); - - // Process the handshake just once (ignore if DTLS renegotiation). - // if (!wasHandshakeDone && this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE) - // return ProcessHandshake(); - if (!wasHandshakeDone) { - return ProcessHandshake(); - } - - return true; - } - // Check if the peer sent close alert or a fatal error happened. - else if (((SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN) != 0) || err == SSL_ERROR_SSL || - err == SSL_ERROR_SYSCALL) { - if (this->state == DtlsState::CONNECTED) { - MS_DEBUG_TAG(dtls, "disconnected"); - - Reset(); - - // Set state and notify the listener. - this->state = DtlsState::CLOSED; - this->listener->OnDtlsTransportClosed(this); - } else { - MS_WARN_TAG(dtls, "connection failed"); - - Reset(); - - // Set state and notify the listener. - this->state = DtlsState::FAILED; - this->listener->OnDtlsTransportFailed(this); - } - - return false; - } else { - return true; - } -} - -inline void DtlsTransport::SendPendingOutgoingDtlsData() { - MS_TRACE(); - - if (BIO_eof(this->sslBioToNetwork)) return; - - int64_t read; - char* data{nullptr}; - - read = BIO_get_mem_data(this->sslBioToNetwork, &data); // NOLINT - - if (read <= 0) return; - - MS_DEBUG_DEV("%ld bytes of DTLS data ready to sent to the peer", read); - - // Notify the listener. - this->listener->OnDtlsTransportSendData(this, reinterpret_cast(data), - static_cast(read)); - - // Clear the BIO buffer. - // NOTE: the (void) avoids the -Wunused-value warning. - (void)BIO_reset(this->sslBioToNetwork); -} - -inline bool DtlsTransport::SetTimeout() { - MS_TRACE(); - - MS_ASSERT(this->state == DtlsState::CONNECTING || this->state == DtlsState::CONNECTED, - "invalid DTLS state"); - - int64_t ret; - uv_timeval_t dtlsTimeout{0, 0}; - uint64_t timeoutMs; - - // NOTE: If ret == 0 then ignore the value in dtlsTimeout. - // NOTE: No DTLSv_1_2_get_timeout() or DTLS_get_timeout() in OpenSSL 1.1.0-dev. - ret = DTLSv1_get_timeout(this->ssl, static_cast(&dtlsTimeout)); // NOLINT - - if (ret == 0) return true; - - timeoutMs = (dtlsTimeout.tv_sec * static_cast(1000)) + (dtlsTimeout.tv_usec / 1000); - - if (timeoutMs == 0) { - return true; - } else if (timeoutMs < 30000) { - MS_DEBUG_DEV("DTLS timer set in %lu ms", timeoutMs); - - // this->timer->Start(timeoutMs); - - return true; - } - // NOTE: Don't start the timer again if the timeout is greater than 30 seconds. - else { - MS_WARN_TAG(dtls, "DTLS timeout too high (%lu ms), resetting DLTS", timeoutMs); - - Reset(); - - // Set state and notify the listener. - this->state = DtlsState::FAILED; - this->listener->OnDtlsTransportFailed(this); - - return false; - } -} - -inline bool DtlsTransport::ProcessHandshake() { - MS_TRACE(); - - MS_ASSERT(this->handshakeDone, "handshake not done yet"); -// MS_ASSERT(this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, -// "remote fingerprint not set"); - - // Validate the remote fingerprint. - // if (!CheckRemoteFingerprint()) { - // Reset(); - - // // Set state and notify the listener. - // this->state = DtlsState::FAILED; - // this->listener->OnDtlsTransportFailed(this); - - // return false; - // } - - // Get the negotiated SRTP crypto suite. - RTC::CryptoSuite srtpCryptoSuite = GetNegotiatedSrtpCryptoSuite(); - - if (srtpCryptoSuite != RTC::CryptoSuite::NONE) { - // Extract the SRTP keys (will notify the listener with them). - ExtractSrtpKeys(srtpCryptoSuite); - - return true; - } - - // NOTE: We assume that "use_srtp" DTLS extension is required even if - // there is no audio/video. - MS_WARN_2TAGS(dtls, srtp, "SRTP crypto suite not negotiated"); - - Reset(); - - // Set state and notify the listener. - this->state = DtlsState::FAILED; - this->listener->OnDtlsTransportFailed(this); - - return false; -} - -inline bool DtlsTransport::CheckRemoteFingerprint() { - MS_TRACE(); - - MS_ASSERT(this->remoteFingerprint.algorithm != FingerprintAlgorithm::NONE, - "remote fingerprint not set"); - - X509* certificate; - uint8_t binaryFingerprint[EVP_MAX_MD_SIZE]; - unsigned int size{0}; - char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1]; - const EVP_MD* hashFunction; - int ret; - - certificate = SSL_get_peer_certificate(this->ssl); - - if (!certificate) { - MS_WARN_TAG(dtls, "no certificate was provided by the peer"); - - return false; - } - - switch (this->remoteFingerprint.algorithm) { - case FingerprintAlgorithm::SHA1: - hashFunction = EVP_sha1(); - break; - - case FingerprintAlgorithm::SHA224: - hashFunction = EVP_sha224(); - break; - - case FingerprintAlgorithm::SHA256: - hashFunction = EVP_sha256(); - break; - - case FingerprintAlgorithm::SHA384: - hashFunction = EVP_sha384(); - break; - - case FingerprintAlgorithm::SHA512: - hashFunction = EVP_sha512(); - break; - - default: - MS_ABORT("unknown algorithm"); - } - - // Compare the remote fingerprint with the value given via signaling. - ret = X509_digest(certificate, hashFunction, binaryFingerprint, &size); - - if (ret == 0) { - MS_ERROR("X509_digest() failed"); - - X509_free(certificate); - - return false; - } - - // Convert to hexadecimal format in uppercase with colons. - for (unsigned int i{0}; i < size; ++i) { - std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]); - } - hexFingerprint[(size * 3) - 1] = '\0'; - - if (this->remoteFingerprint.value != hexFingerprint) { - MS_WARN_TAG(dtls, - "fingerprint in the remote certificate (%s) does not match the announced one (%s)", - hexFingerprint, this->remoteFingerprint.value.c_str()); - - X509_free(certificate); - - return false; - } - - MS_DEBUG_TAG(dtls, "valid remote fingerprint"); - - // Get the remote certificate in PEM format. - - BIO* bio = BIO_new(BIO_s_mem()); - - // Ensure the underlying BUF_MEM structure is also freed. - // NOTE: Avoid stupid "warning: value computed is not used [-Wunused-value]" since - // BIO_set_close() always returns 1. - (void)BIO_set_close(bio, BIO_CLOSE); - - ret = PEM_write_bio_X509(bio, certificate); - - if (ret != 1) { - LOG_OPENSSL_ERROR("PEM_write_bio_X509() failed"); - - X509_free(certificate); - BIO_free(bio); - - return false; - } - - BUF_MEM* mem; - - BIO_get_mem_ptr(bio, &mem); // NOLINT[cppcoreguidelines-pro-type-cstyle-cast] - - if (!mem || !mem->data || mem->length == 0u) { - LOG_OPENSSL_ERROR("BIO_get_mem_ptr() failed"); - - X509_free(certificate); - BIO_free(bio); - - return false; - } - - this->remoteCert = std::string(mem->data, mem->length); - - X509_free(certificate); - BIO_free(bio); - - return true; -} - -inline void DtlsTransport::ExtractSrtpKeys(RTC::CryptoSuite srtpCryptoSuite) { - MS_TRACE(); - - size_t srtpKeyLength{0}; - size_t srtpSaltLength{0}; - size_t srtpMasterLength{0}; - - switch (srtpCryptoSuite) { - case RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_80: - case RTC::CryptoSuite::AES_CM_128_HMAC_SHA1_32: { - srtpKeyLength = SrtpMasterKeyLength; - srtpSaltLength = SrtpMasterSaltLength; - srtpMasterLength = SrtpMasterLength; - - break; - } - - case RTC::CryptoSuite::AEAD_AES_256_GCM: { - srtpKeyLength = SrtpAesGcm256MasterKeyLength; - srtpSaltLength = SrtpAesGcm256MasterSaltLength; - srtpMasterLength = SrtpAesGcm256MasterLength; - - break; - } - - case RTC::CryptoSuite::AEAD_AES_128_GCM: { - srtpKeyLength = SrtpAesGcm128MasterKeyLength; - srtpSaltLength = SrtpAesGcm128MasterSaltLength; - srtpMasterLength = SrtpAesGcm128MasterLength; - - break; - } - - default: { - MS_ABORT("unknown SRTP crypto suite"); - } - } - - auto* srtpMaterial = new uint8_t[srtpMasterLength * 2]; - uint8_t* srtpLocalKey{nullptr}; - uint8_t* srtpLocalSalt{nullptr}; - uint8_t* srtpRemoteKey{nullptr}; - uint8_t* srtpRemoteSalt{nullptr}; - auto* srtpLocalMasterKey = new uint8_t[srtpMasterLength]; - auto* srtpRemoteMasterKey = new uint8_t[srtpMasterLength]; - int ret; - - ret = SSL_export_keying_material(this->ssl, srtpMaterial, srtpMasterLength * 2, - "EXTRACTOR-dtls_srtp", 19, nullptr, 0, 0); - - MS_ASSERT(ret != 0, "SSL_export_keying_material() failed"); - - switch (this->localRole) { - case Role::SERVER: { - srtpRemoteKey = srtpMaterial; - srtpLocalKey = srtpRemoteKey + srtpKeyLength; - srtpRemoteSalt = srtpLocalKey + srtpKeyLength; - srtpLocalSalt = srtpRemoteSalt + srtpSaltLength; - - break; - } - - case Role::CLIENT: { - srtpLocalKey = srtpMaterial; - srtpRemoteKey = srtpLocalKey + srtpKeyLength; - srtpLocalSalt = srtpRemoteKey + srtpKeyLength; - srtpRemoteSalt = srtpLocalSalt + srtpSaltLength; - - break; - } - - default: { - MS_ABORT("no DTLS role set"); - } - } - - // Create the SRTP local master key. - std::memcpy(srtpLocalMasterKey, srtpLocalKey, srtpKeyLength); - std::memcpy(srtpLocalMasterKey + srtpKeyLength, srtpLocalSalt, srtpSaltLength); - // Create the SRTP remote master key. - std::memcpy(srtpRemoteMasterKey, srtpRemoteKey, srtpKeyLength); - std::memcpy(srtpRemoteMasterKey + srtpKeyLength, srtpRemoteSalt, srtpSaltLength); - - // Set state and notify the listener. - this->state = DtlsState::CONNECTED; - this->listener->OnDtlsTransportConnected(this, srtpCryptoSuite, srtpLocalMasterKey, - srtpMasterLength, srtpRemoteMasterKey, srtpMasterLength, - this->remoteCert); - - delete[] srtpMaterial; - delete[] srtpLocalMasterKey; - delete[] srtpRemoteMasterKey; -} - -inline RTC::CryptoSuite DtlsTransport::GetNegotiatedSrtpCryptoSuite() { - MS_TRACE(); - - RTC::CryptoSuite negotiatedSrtpCryptoSuite = RTC::CryptoSuite::NONE; - - // Ensure that the SRTP crypto suite has been negotiated. - // NOTE: This is a OpenSSL type. - SRTP_PROTECTION_PROFILE* sslSrtpCryptoSuite = SSL_get_selected_srtp_profile(this->ssl); - - if (!sslSrtpCryptoSuite) return negotiatedSrtpCryptoSuite; - - // Get the negotiated SRTP crypto suite. - for (auto& srtpCryptoSuite : DtlsTransport::srtpCryptoSuites) { - SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(srtpCryptoSuite); - - if (std::strcmp(sslSrtpCryptoSuite->name, cryptoSuiteEntry->name) == 0) { - MS_DEBUG_2TAGS(dtls, srtp, "chosen SRTP crypto suite: %s", cryptoSuiteEntry->name); - - negotiatedSrtpCryptoSuite = cryptoSuiteEntry->cryptoSuite; - } - } - - MS_ASSERT(negotiatedSrtpCryptoSuite != RTC::CryptoSuite::NONE, - "chosen SRTP crypto suite is not an available one"); - - return negotiatedSrtpCryptoSuite; -} - -inline void DtlsTransport::OnSslInfo(int where, int ret) { - MS_TRACE(); - - int w = where & -SSL_ST_MASK; - const char* role; - - if ((w & SSL_ST_CONNECT) != 0) - role = "client"; - else if ((w & SSL_ST_ACCEPT) != 0) - role = "server"; - else - role = "undefined"; - - if ((where & SSL_CB_LOOP) != 0) { - MS_DEBUG_TAG(dtls, "[role:%s, action:'%s']", role, SSL_state_string_long(this->ssl)); - } else if ((where & SSL_CB_ALERT) != 0) { - const char* alertType; - - switch (*SSL_alert_type_string(ret)) { - case 'W': - alertType = "warning"; - break; - - case 'F': - alertType = "fatal"; - break; - - default: - alertType = "undefined"; - } - - if ((where & SSL_CB_READ) != 0) { - MS_WARN_TAG(dtls, "received DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); - } else if ((where & SSL_CB_WRITE) != 0) { - MS_DEBUG_TAG(dtls, "sending DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); - } else { - MS_DEBUG_TAG(dtls, "DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); - } - } else if ((where & SSL_CB_EXIT) != 0) { - if (ret == 0) - MS_DEBUG_TAG(dtls, "[role:%s, failed:'%s']", role, SSL_state_string_long(this->ssl)); - else if (ret < 0) - MS_DEBUG_TAG(dtls, "role: %s, waiting:'%s']", role, SSL_state_string_long(this->ssl)); - } else if ((where & SSL_CB_HANDSHAKE_START) != 0) { - MS_DEBUG_TAG(dtls, "DTLS handshake start"); - } else if ((where & SSL_CB_HANDSHAKE_DONE) != 0) { - MS_DEBUG_TAG(dtls, "DTLS handshake done"); - - this->handshakeDoneNow = true; - } - - // NOTE: checking SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN here upon - // receipt of a close alert does not work (the flag is set after this callback). -} - -inline void DtlsTransport::OnTimer() { - MS_TRACE(); - - // Workaround for https://github.com/openssl/openssl/issues/7998. - if (this->handshakeDone) { - MS_DEBUG_DEV("handshake is done so return"); - - return; - } - - DTLSv1_handle_timeout(this->ssl); - - // If required, send DTLS data. - SendPendingOutgoingDtlsData(); - - // Set the DTLS timer again. - SetTimeout(); -} -} // namespace RTC + X509_free(certificate); + BIO_free(bio); + + return true; + } + + inline void DtlsTransport::ExtractSrtpKeys(RTC::SrtpSession::CryptoSuite srtpCryptoSuite) + { + MS_TRACE(); + + size_t srtpKeyLength{ 0 }; + size_t srtpSaltLength{ 0 }; + size_t srtpMasterLength{ 0 }; + + switch (srtpCryptoSuite) + { + case RTC::SrtpSession::CryptoSuite::AES_CM_128_HMAC_SHA1_80: + case RTC::SrtpSession::CryptoSuite::AES_CM_128_HMAC_SHA1_32: + { + srtpKeyLength = SrtpMasterKeyLength; + srtpSaltLength = SrtpMasterSaltLength; + srtpMasterLength = SrtpMasterLength; + + break; + } + + case RTC::SrtpSession::CryptoSuite::AEAD_AES_256_GCM: + { + srtpKeyLength = SrtpAesGcm256MasterKeyLength; + srtpSaltLength = SrtpAesGcm256MasterSaltLength; + srtpMasterLength = SrtpAesGcm256MasterLength; + + break; + } + + case RTC::SrtpSession::CryptoSuite::AEAD_AES_128_GCM: + { + srtpKeyLength = SrtpAesGcm128MasterKeyLength; + srtpSaltLength = SrtpAesGcm128MasterSaltLength; + srtpMasterLength = SrtpAesGcm128MasterLength; + + break; + } + + default: + { + MS_ABORT("unknown SRTP crypto suite"); + } + } + + auto* srtpMaterial = new uint8_t[srtpMasterLength * 2]; + uint8_t* srtpLocalKey{ nullptr }; + uint8_t* srtpLocalSalt{ nullptr }; + uint8_t* srtpRemoteKey{ nullptr }; + uint8_t* srtpRemoteSalt{ nullptr }; + auto* srtpLocalMasterKey = new uint8_t[srtpMasterLength]; + auto* srtpRemoteMasterKey = new uint8_t[srtpMasterLength]; + int ret; + + ret = SSL_export_keying_material( + this->ssl, srtpMaterial, srtpMasterLength * 2, "EXTRACTOR-dtls_srtp", 19, nullptr, 0, 0); + + MS_ASSERT(ret != 0, "SSL_export_keying_material() failed"); + + switch (this->localRole) + { + case Role::SERVER: + { + srtpRemoteKey = srtpMaterial; + srtpLocalKey = srtpRemoteKey + srtpKeyLength; + srtpRemoteSalt = srtpLocalKey + srtpKeyLength; + srtpLocalSalt = srtpRemoteSalt + srtpSaltLength; + + break; + } + + case Role::CLIENT: + { + srtpLocalKey = srtpMaterial; + srtpRemoteKey = srtpLocalKey + srtpKeyLength; + srtpLocalSalt = srtpRemoteKey + srtpKeyLength; + srtpRemoteSalt = srtpLocalSalt + srtpSaltLength; + + break; + } + + default: + { + MS_ABORT("no DTLS role set"); + } + } + + // Create the SRTP local master key. + std::memcpy(srtpLocalMasterKey, srtpLocalKey, srtpKeyLength); + std::memcpy(srtpLocalMasterKey + srtpKeyLength, srtpLocalSalt, srtpSaltLength); + // Create the SRTP remote master key. + std::memcpy(srtpRemoteMasterKey, srtpRemoteKey, srtpKeyLength); + std::memcpy(srtpRemoteMasterKey + srtpKeyLength, srtpRemoteSalt, srtpSaltLength); + + // Set state and notify the listener. + this->state = DtlsState::CONNECTED; + this->listener->OnDtlsTransportConnected( + this, + srtpCryptoSuite, + srtpLocalMasterKey, + srtpMasterLength, + srtpRemoteMasterKey, + srtpMasterLength, + this->remoteCert); + + delete[] srtpMaterial; + delete[] srtpLocalMasterKey; + delete[] srtpRemoteMasterKey; + } + + inline RTC::SrtpSession::CryptoSuite DtlsTransport::GetNegotiatedSrtpCryptoSuite() + { + MS_TRACE(); + + RTC::SrtpSession::CryptoSuite negotiatedSrtpCryptoSuite = RTC::SrtpSession::CryptoSuite::NONE; + + // Ensure that the SRTP crypto suite has been negotiated. + // NOTE: This is a OpenSSL type. + SRTP_PROTECTION_PROFILE* sslSrtpCryptoSuite = SSL_get_selected_srtp_profile(this->ssl); + + if (!sslSrtpCryptoSuite) + return negotiatedSrtpCryptoSuite; + + // Get the negotiated SRTP crypto suite. + for (auto& srtpCryptoSuite : DtlsTransport::srtpCryptoSuites) + { + SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(srtpCryptoSuite); + + if (std::strcmp(sslSrtpCryptoSuite->name, cryptoSuiteEntry->name) == 0) + { + MS_DEBUG_2TAGS(dtls, srtp, "chosen SRTP crypto suite: %s", cryptoSuiteEntry->name); + + negotiatedSrtpCryptoSuite = cryptoSuiteEntry->cryptoSuite; + } + } + + MS_ASSERT( + negotiatedSrtpCryptoSuite != RTC::SrtpSession::CryptoSuite::NONE, + "chosen SRTP crypto suite is not an available one"); + + return negotiatedSrtpCryptoSuite; + } + + inline void DtlsTransport::OnSslInfo(int where, int ret) + { + MS_TRACE(); + + int w = where & -SSL_ST_MASK; + const char* role; + + if ((w & SSL_ST_CONNECT) != 0) + role = "client"; + else if ((w & SSL_ST_ACCEPT) != 0) + role = "server"; + else + role = "undefined"; + + if ((where & SSL_CB_LOOP) != 0) + { + MS_DEBUG_TAG(dtls, "[role:%s, action:'%s']", role, SSL_state_string_long(this->ssl)); + } + else if ((where & SSL_CB_ALERT) != 0) + { + const char* alertType; + + switch (*SSL_alert_type_string(ret)) + { + case 'W': + alertType = "warning"; + break; + + case 'F': + alertType = "fatal"; + break; + + default: + alertType = "undefined"; + } + + if ((where & SSL_CB_READ) != 0) + { + MS_WARN_TAG(dtls, "received DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } + else if ((where & SSL_CB_WRITE) != 0) + { + MS_DEBUG_TAG(dtls, "sending DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } + else + { + MS_DEBUG_TAG(dtls, "DTLS %s alert: %s", alertType, SSL_alert_desc_string_long(ret)); + } + } + else if ((where & SSL_CB_EXIT) != 0) + { + if (ret == 0) + MS_DEBUG_TAG(dtls, "[role:%s, failed:'%s']", role, SSL_state_string_long(this->ssl)); + else if (ret < 0) + MS_DEBUG_TAG(dtls, "role: %s, waiting:'%s']", role, SSL_state_string_long(this->ssl)); + } + else if ((where & SSL_CB_HANDSHAKE_START) != 0) + { + MS_DEBUG_TAG(dtls, "DTLS handshake start"); + } + else if ((where & SSL_CB_HANDSHAKE_DONE) != 0) + { + MS_DEBUG_TAG(dtls, "DTLS handshake done"); + + this->handshakeDoneNow = true; + } + + // NOTE: checking SSL_get_shutdown(this->ssl) & SSL_RECEIVED_SHUTDOWN here upon + // receipt of a close alert does not work (the flag is set after this callback). + } + + inline void DtlsTransport::OnTimer() + { + MS_TRACE(); + + // Workaround for https://github.com/openssl/openssl/issues/7998. + if (this->handshakeDone) + { + MS_DEBUG_DEV("handshake is done so return"); + + return; + } + + DTLSv1_handle_timeout(this->ssl); + + // If required, send DTLS data. + SendPendingOutgoingDtlsData(); + + // Set the DTLS timer again. + SetTimeout(); + } +} // namespace RTC diff --git a/webrtc/rtc_dtls_transport.h b/webrtc/rtc_dtls_transport.h index 628f966b..b3e58b1d 100644 --- a/webrtc/rtc_dtls_transport.h +++ b/webrtc/rtc_dtls_transport.h @@ -1,187 +1,224 @@ #ifndef MS_RTC_DTLS_TRANSPORT_HPP #define MS_RTC_DTLS_TRANSPORT_HPP +#include "srtp_session.h" #include #include #include - #include #include #include +#include "Poller/Timer.h" +#include "Poller/EventPoller.h" +using namespace toolkit; -namespace RTC { -enum class CryptoSuite { - NONE = 0, - AES_CM_128_HMAC_SHA1_80 = 1, - AES_CM_128_HMAC_SHA1_32, - AEAD_AES_256_GCM, - AEAD_AES_128_GCM -}; -class DtlsTransport { - public: - enum class DtlsState { NEW = 1, CONNECTING, CONNECTED, FAILED, CLOSED }; +namespace RTC +{ +class DtlsTransport : public std::enable_shared_from_this + { + public: + enum class DtlsState + { + NEW = 1, + CONNECTING, + CONNECTED, + FAILED, + CLOSED + }; - public: - enum class Role { NONE = 0, AUTO = 1, CLIENT, SERVER }; + public: + enum class Role + { + NONE = 0, + AUTO = 1, + CLIENT, + SERVER + }; - public: - enum class FingerprintAlgorithm { NONE = 0, SHA1 = 1, SHA224, SHA256, SHA384, SHA512 }; + public: + enum class FingerprintAlgorithm + { + NONE = 0, + SHA1 = 1, + SHA224, + SHA256, + SHA384, + SHA512 + }; - public: - struct Fingerprint { - FingerprintAlgorithm algorithm{FingerprintAlgorithm::NONE}; - std::string value; - }; + public: + struct Fingerprint + { + FingerprintAlgorithm algorithm{ FingerprintAlgorithm::NONE }; + std::string value; + }; - private: - struct SrtpCryptoSuiteMapEntry { - RTC::CryptoSuite cryptoSuite; - const char* name; - }; + private: + struct SrtpCryptoSuiteMapEntry + { + RTC::SrtpSession::CryptoSuite cryptoSuite; + const char* name; + }; - public: - class Listener { - public: - // DTLS is in the process of negotiating a secure connection. Incoming - // media can flow through. - // NOTE: The caller MUST NOT call any method during this callback. - virtual void OnDtlsTransportConnecting(const RTC::DtlsTransport* dtlsTransport) = 0; - // DTLS has completed negotiation of a secure connection (including DTLS-SRTP - // and remote fingerprint verification). Outgoing media can now flow through. - // NOTE: The caller MUST NOT call any method during this callback. - virtual void OnDtlsTransportConnected(const RTC::DtlsTransport* dtlsTransport, - RTC::CryptoSuite srtpCryptoSuite, uint8_t* srtpLocalKey, - size_t srtpLocalKeyLen, uint8_t* srtpRemoteKey, - size_t srtpRemoteKeyLen, std::string& remoteCert) = 0; - // The DTLS connection has been closed as the result of an error (such as a - // DTLS alert or a failure to validate the remote fingerprint). - virtual void OnDtlsTransportFailed(const RTC::DtlsTransport* dtlsTransport) = 0; - // The DTLS connection has been closed due to receipt of a close_notify alert. - virtual void OnDtlsTransportClosed(const RTC::DtlsTransport* dtlsTransport) = 0; - // Need to send DTLS data to the peer. - virtual void OnDtlsTransportSendData(const RTC::DtlsTransport* dtlsTransport, - const uint8_t* data, size_t len) = 0; - // DTLS application data received. - virtual void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport* dtlsTransport, - const uint8_t* data, size_t len) = 0; - }; + public: + class Listener + { + public: + // DTLS is in the process of negotiating a secure connection. Incoming + // media can flow through. + // NOTE: The caller MUST NOT call any method during this callback. + virtual void OnDtlsTransportConnecting(const RTC::DtlsTransport* dtlsTransport) = 0; + // DTLS has completed negotiation of a secure connection (including DTLS-SRTP + // and remote fingerprint verification). Outgoing media can now flow through. + // NOTE: The caller MUST NOT call any method during this callback. + virtual void OnDtlsTransportConnected( + const RTC::DtlsTransport* dtlsTransport, + RTC::SrtpSession::CryptoSuite srtpCryptoSuite, + uint8_t* srtpLocalKey, + size_t srtpLocalKeyLen, + uint8_t* srtpRemoteKey, + size_t srtpRemoteKeyLen, + std::string& remoteCert) = 0; + // The DTLS connection has been closed as the result of an error (such as a + // DTLS alert or a failure to validate the remote fingerprint). + virtual void OnDtlsTransportFailed(const RTC::DtlsTransport* dtlsTransport) = 0; + // The DTLS connection has been closed due to receipt of a close_notify alert. + virtual void OnDtlsTransportClosed(const RTC::DtlsTransport* dtlsTransport) = 0; + // Need to send DTLS data to the peer. + virtual void OnDtlsTransportSendData( + const RTC::DtlsTransport* dtlsTransport, const uint8_t* data, size_t len) = 0; + // DTLS application data received. + virtual void OnDtlsTransportApplicationDataReceived( + const RTC::DtlsTransport* dtlsTransport, const uint8_t* data, size_t len) = 0; + }; - public: - static void ClassInit(); - static void ClassDestroy(); - static Role StringToRole(const std::string& role) { - auto it = DtlsTransport::string2Role.find(role); + public: + static void ClassInit(); + static void ClassDestroy(); + static Role StringToRole(const std::string& role) + { + auto it = DtlsTransport::string2Role.find(role); - if (it != DtlsTransport::string2Role.end()) - return it->second; - else - return DtlsTransport::Role::NONE; - } - static FingerprintAlgorithm GetFingerprintAlgorithm(const std::string& fingerprint) { - auto it = DtlsTransport::string2FingerprintAlgorithm.find(fingerprint); + if (it != DtlsTransport::string2Role.end()) + return it->second; + else + return DtlsTransport::Role::NONE; + } + static FingerprintAlgorithm GetFingerprintAlgorithm(const std::string& fingerprint) + { + auto it = DtlsTransport::string2FingerprintAlgorithm.find(fingerprint); - if (it != DtlsTransport::string2FingerprintAlgorithm.end()) - return it->second; - else - return DtlsTransport::FingerprintAlgorithm::NONE; - } - static std::string& GetFingerprintAlgorithmString(FingerprintAlgorithm fingerprint) { - auto it = DtlsTransport::fingerprintAlgorithm2String.find(fingerprint); + if (it != DtlsTransport::string2FingerprintAlgorithm.end()) + return it->second; + else + return DtlsTransport::FingerprintAlgorithm::NONE; + } + static std::string& GetFingerprintAlgorithmString(FingerprintAlgorithm fingerprint) + { + auto it = DtlsTransport::fingerprintAlgorithm2String.find(fingerprint); - return it->second; - } - static bool IsDtls(const uint8_t* data, size_t len) { - // clang-format off + return it->second; + } + static bool IsDtls(const uint8_t* data, size_t len) + { + // clang-format off return ( // Minimum DTLS record length is 13 bytes. (len >= 13) && // DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes (data[0] > 19 && data[0] < 64) ); - // clang-format on - } + // clang-format on + } - private: - static void GenerateCertificateAndPrivateKey(); - static void ReadCertificateAndPrivateKeyFromFiles(); - static void CreateSslCtx(); - static void GenerateFingerprints(); + private: + static void GenerateCertificateAndPrivateKey(); + static void ReadCertificateAndPrivateKeyFromFiles(); + static void CreateSslCtx(); + static void GenerateFingerprints(); - private: - static X509* certificate; - static EVP_PKEY* privateKey; - static SSL_CTX* sslCtx; - static uint8_t sslReadBuffer[]; - static std::map string2Role; - static std::map string2FingerprintAlgorithm; - static std::map fingerprintAlgorithm2String; - static std::vector localFingerprints; - static std::vector srtpCryptoSuites; + private: + static X509* certificate; + static EVP_PKEY* privateKey; + static SSL_CTX* sslCtx; + static uint8_t sslReadBuffer[]; + static std::map string2Role; + static std::map string2FingerprintAlgorithm; + static std::map fingerprintAlgorithm2String; + static std::vector localFingerprints; + static std::vector srtpCryptoSuites; - public: - explicit DtlsTransport(Listener* listener); - ~DtlsTransport(); + public: + DtlsTransport(EventPoller::Ptr poller, Listener* listener); + ~DtlsTransport(); - public: - void Dump() const; - void Run(Role localRole); - std::vector& GetLocalFingerprints() const { - return DtlsTransport::localFingerprints; - } - bool SetRemoteFingerprint(Fingerprint fingerprint); - void ProcessDtlsData(const uint8_t* data, size_t len); - DtlsState GetState() const { return this->state; } - Role GetLocalRole() const { return this->localRole; } - void SendApplicationData(const uint8_t* data, size_t len); + public: + void Dump() const; + void Run(Role localRole); + std::vector& GetLocalFingerprints() const + { + return DtlsTransport::localFingerprints; + } + bool SetRemoteFingerprint(Fingerprint fingerprint); + void ProcessDtlsData(const uint8_t* data, size_t len); + DtlsState GetState() const + { + return this->state; + } + Role GetLocalRole() const + { + return this->localRole; + } + void SendApplicationData(const uint8_t* data, size_t len); - private: - bool IsRunning() const { - switch (this->state) { - case DtlsState::NEW: - return false; - case DtlsState::CONNECTING: - case DtlsState::CONNECTED: - return true; - case DtlsState::FAILED: - case DtlsState::CLOSED: - return false; - } + private: + bool IsRunning() const + { + switch (this->state) + { + case DtlsState::NEW: + return false; + case DtlsState::CONNECTING: + case DtlsState::CONNECTED: + return true; + case DtlsState::FAILED: + case DtlsState::CLOSED: + return false; + } - // Make GCC 4.9 happy. - return false; - } - void Reset(); - bool CheckStatus(int returnCode); - void SendPendingOutgoingDtlsData(); - bool SetTimeout(); - bool ProcessHandshake(); - bool CheckRemoteFingerprint(); - void ExtractSrtpKeys(RTC::CryptoSuite srtpCryptoSuite); - RTC::CryptoSuite GetNegotiatedSrtpCryptoSuite(); + // Make GCC 4.9 happy. + return false; + } + void Reset(); + bool CheckStatus(int returnCode); + void SendPendingOutgoingDtlsData(); + bool SetTimeout(); + bool ProcessHandshake(); + bool CheckRemoteFingerprint(); + void ExtractSrtpKeys(RTC::SrtpSession::CryptoSuite srtpCryptoSuite); + RTC::SrtpSession::CryptoSuite GetNegotiatedSrtpCryptoSuite(); - /* Callbacks fired by OpenSSL events. */ - public: - void OnSslInfo(int where, int ret); + private: + void OnSslInfo(int where, int ret); + void OnTimer(); - /* Pure virtual methods inherited from Timer::Listener. */ - public: - void OnTimer(); - - private: - // Passed by argument. - Listener* listener{nullptr}; - // Allocated by this. - SSL* ssl{nullptr}; - BIO* sslBioFromNetwork{nullptr}; // The BIO from which ssl reads. - BIO* sslBioToNetwork{nullptr}; // The BIO in which ssl writes. - // Others. - DtlsState state{DtlsState::NEW}; - Role localRole{Role::NONE}; - Fingerprint remoteFingerprint; - bool handshakeDone{false}; - bool handshakeDoneNow{false}; - std::string remoteCert; -}; -} // namespace RTC + private: + EventPoller::Ptr poller; + // Passed by argument. + Listener* listener{ nullptr }; + // Allocated by this. + SSL* ssl{ nullptr }; + BIO* sslBioFromNetwork{ nullptr }; // The BIO from which ssl reads. + BIO* sslBioToNetwork{ nullptr }; // The BIO in which ssl writes. + Timer::Ptr timer; + // Others. + DtlsState state{ DtlsState::NEW }; + Role localRole{ Role::NONE }; + Fingerprint remoteFingerprint; + bool handshakeDone{ false }; + bool handshakeDoneNow{ false }; + std::string remoteCert; + }; +} // namespace RTC #endif diff --git a/webrtc/srtp_session.cc b/webrtc/srtp_session.cc index b21a83cd..fd8de70d 100644 --- a/webrtc/srtp_session.cc +++ b/webrtc/srtp_session.cc @@ -2,268 +2,287 @@ // #define MS_LOG_DEV_LEVEL 3 #include "srtp_session.h" - -#include // std::memset(), std::memcpy() -#include - +#include // std::memset(), std::memcpy() #include "logger.h" -namespace RTC { -/* Static. */ +namespace RTC +{ + /* Static. */ -static constexpr size_t EncryptBufferSize{65536}; -static uint8_t EncryptBuffer[EncryptBufferSize]; + static constexpr size_t EncryptBufferSize{ 65536 }; + static uint8_t EncryptBuffer[EncryptBufferSize]; -/* Class methods. */ - -std::vector DepLibSRTP::errors = { - // From 0 (srtp_err_status_ok) to 24 (srtp_err_status_pfkey_err). - "success (srtp_err_status_ok)", - "unspecified failure (srtp_err_status_fail)", - "unsupported parameter (srtp_err_status_bad_param)", - "couldn't allocate memory (srtp_err_status_alloc_fail)", - "couldn't deallocate memory (srtp_err_status_dealloc_fail)", - "couldn't initialize (srtp_err_status_init_fail)", - "can’t process as much data as requested (srtp_err_status_terminus)", - "authentication failure (srtp_err_status_auth_fail)", - "cipher failure (srtp_err_status_cipher_fail)", - "replay check failed (bad index) (srtp_err_status_replay_fail)", - "replay check failed (index too old) (srtp_err_status_replay_old)", - "algorithm failed test routine (srtp_err_status_algo_fail)", - "unsupported operation (srtp_err_status_no_such_op)", - "no appropriate context found (srtp_err_status_no_ctx)", - "unable to perform desired validation (srtp_err_status_cant_check)", - "can’t use key any more (srtp_err_status_key_expired)", - "error in use of socket (srtp_err_status_socket_err)", - "error in use POSIX signals (srtp_err_status_signal_err)", - "nonce check failed (srtp_err_status_nonce_bad)", - "couldn’t read data (srtp_err_status_read_fail)", - "couldn’t write data (srtp_err_status_write_fail)", - "error parsing data (srtp_err_status_parse_err)", - "error encoding data (srtp_err_status_encode_err)", - "error while using semaphores (srtp_err_status_semaphore_err)", - "error while using pfkey (srtp_err_status_pfkey_err)"}; + std::vector DepLibSRTP::errors = { + // From 0 (srtp_err_status_ok) to 24 (srtp_err_status_pfkey_err). + "success (srtp_err_status_ok)", + "unspecified failure (srtp_err_status_fail)", + "unsupported parameter (srtp_err_status_bad_param)", + "couldn't allocate memory (srtp_err_status_alloc_fail)", + "couldn't deallocate memory (srtp_err_status_dealloc_fail)", + "couldn't initialize (srtp_err_status_init_fail)", + "can’t process as much data as requested (srtp_err_status_terminus)", + "authentication failure (srtp_err_status_auth_fail)", + "cipher failure (srtp_err_status_cipher_fail)", + "replay check failed (bad index) (srtp_err_status_replay_fail)", + "replay check failed (index too old) (srtp_err_status_replay_old)", + "algorithm failed test routine (srtp_err_status_algo_fail)", + "unsupported operation (srtp_err_status_no_such_op)", + "no appropriate context found (srtp_err_status_no_ctx)", + "unable to perform desired validation (srtp_err_status_cant_check)", + "can’t use key any more (srtp_err_status_key_expired)", + "error in use of socket (srtp_err_status_socket_err)", + "error in use POSIX signals (srtp_err_status_signal_err)", + "nonce check failed (srtp_err_status_nonce_bad)", + "couldn’t read data (srtp_err_status_read_fail)", + "couldn’t write data (srtp_err_status_write_fail)", + "error parsing data (srtp_err_status_parse_err)", + "error encoding data (srtp_err_status_encode_err)", + "error while using semaphores (srtp_err_status_semaphore_err)", + "error while using pfkey (srtp_err_status_pfkey_err)"}; // clang-format on /* Static methods. */ -void DepLibSRTP::ClassInit() { - MS_TRACE(); + void DepLibSRTP::ClassInit() { + MS_TRACE(); - MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string()); + MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string()); - srtp_err_status_t err = srtp_init(); - - if (DepLibSRTP::IsError(err)) - MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err)); -} - -void DepLibSRTP::ClassDestroy() { - MS_TRACE(); - - srtp_shutdown(); -} - -void SrtpSession::ClassInit() { - // Set libsrtp event handler. - srtp_err_status_t err = - srtp_install_event_handler(static_cast(OnSrtpEvent)); - if (DepLibSRTP::IsError(err)) { - MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err)); - std::cout << "srtp_install_event_handler() failed :" << DepLibSRTP::GetErrorString(err); - } -} - -void SrtpSession::OnSrtpEvent(srtp_event_data_t *data) { - MS_TRACE(); - - switch (data->event) { - case event_ssrc_collision: - MS_WARN_TAG(srtp, "SSRC collision occurred"); - break; - - case event_key_soft_limit: - MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon"); - break; - - case event_key_hard_limit: - MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired"); - break; - - case event_packet_index_limit: - MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)"); - break; - } -} - -/* Instance methods. */ - -SrtpSession::SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t *key, size_t keyLen) { - MS_TRACE(); - - srtp_policy_t policy;// NOLINT(cppcoreguidelines-pro-type-member-init) - - // Set all policy fields to 0. - std::memset(&policy, 0, sizeof(srtp_policy_t)); - - switch (cryptoSuite) { - case CryptoSuite::AES_CM_128_HMAC_SHA1_80: { - srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp); - srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); - - break; - } - - case CryptoSuite::AES_CM_128_HMAC_SHA1_32: { - srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); - // NOTE: Must be 80 for RTCP. - srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); - - break; - } - - case CryptoSuite::AEAD_AES_256_GCM: { - srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp); - srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp); - - break; - } - - case CryptoSuite::AEAD_AES_128_GCM: { - srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp); - srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp); - - break; - } - - default: { - MS_ABORT("unknown SRTP crypto suite"); - } - } - - MS_ASSERT((int) keyLen == policy.rtp.cipher_key_len, - "given keyLen does not match policy.rtp.cipher_keyLen"); - - switch (type) { - case Type::INBOUND: - policy.ssrc.type = ssrc_any_inbound; - break; - - case Type::OUTBOUND: - policy.ssrc.type = ssrc_any_outbound; - break; - } - - policy.ssrc.value = 0; - policy.key = key; - // Required for sending RTP retransmission without RTX. - policy.allow_repeat_tx = 1; - policy.window_size = 1024; - policy.next = nullptr; - - // Set the SRTP session. - srtp_err_status_t err = srtp_create(&this->session, &policy); - if (DepLibSRTP::IsError(err)) { - is_init = false; - MS_THROW_ERROR("srtp_create() failed: %s", DepLibSRTP::GetErrorString(err)); - } else { - is_init = true; - } -} - -SrtpSession::~SrtpSession() { - MS_TRACE(); - - if (this->session != nullptr) { - srtp_err_status_t err = srtp_dealloc(this->session); + srtp_err_status_t err = srtp_init(); if (DepLibSRTP::IsError(err)) - MS_ABORT("srtp_dealloc() failed: %s", DepLibSRTP::GetErrorString(err)); - } -} - -bool SrtpSession::EncryptRtp(const uint8_t **data, size_t *len) { - MS_TRACE(); - if (!is_init) { - return false; - } - // Ensure that the resulting SRTP packet fits into the encrypt buffer. - if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) { - MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len); - - return false; - } - std::memcpy(EncryptBuffer, *data, *len); - - srtp_err_status_t err = - srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); - - if (DepLibSRTP::IsError(err)) { - MS_WARN_TAG(srtp, "srtp_protect() failed: %s", DepLibSRTP::GetErrorString(err)); - - return false; + MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err)); } - // Update the given data pointer. - *data = (const uint8_t *) EncryptBuffer; + void DepLibSRTP::ClassDestroy() { + MS_TRACE(); - return true; -} - -bool SrtpSession::DecryptSrtp(uint8_t *data, size_t *len) { - MS_TRACE(); - - srtp_err_status_t err = - srtp_unprotect(this->session, static_cast(data), reinterpret_cast(len)); - - if (DepLibSRTP::IsError(err)) { - MS_DEBUG_TAG(srtp, "srtp_unprotect() failed: %s", DepLibSRTP::GetErrorString(err)); - - return false; + srtp_shutdown(); } - return true; -} + /* Class methods. */ -bool SrtpSession::EncryptRtcp(const uint8_t **data, size_t *len) { - MS_TRACE(); + void SrtpSession::ClassInit() + { + // Set libsrtp event handler. + srtp_err_status_t err = + srtp_install_event_handler(static_cast(OnSrtpEvent)); - // Ensure that the resulting SRTCP packet fits into the encrypt buffer. - if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) { - MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len); + if (DepLibSRTP::IsError(err)) + { + MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err)); + } + } - return false; - } + void SrtpSession::OnSrtpEvent(srtp_event_data_t* data) + { + MS_TRACE(); - std::memcpy(EncryptBuffer, *data, *len); + switch (data->event) + { + case event_ssrc_collision: + MS_WARN_TAG(srtp, "SSRC collision occurred"); + break; - srtp_err_status_t err = srtp_protect_rtcp(this->session, static_cast(EncryptBuffer), - reinterpret_cast(len)); + case event_key_soft_limit: + MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon"); + break; - if (DepLibSRTP::IsError(err)) { - MS_WARN_TAG(srtp, "srtp_protect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + case event_key_hard_limit: + MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired"); + break; - return false; - } + case event_packet_index_limit: + MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)"); + break; + } + } - // Update the given data pointer. - *data = (const uint8_t *) EncryptBuffer; + /* Instance methods. */ - return true; -} + SrtpSession::SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen) + { + MS_TRACE(); -bool SrtpSession::DecryptSrtcp(uint8_t *data, size_t *len) { - MS_TRACE(); + srtp_policy_t policy; // NOLINT(cppcoreguidelines-pro-type-member-init) - srtp_err_status_t err = - srtp_unprotect_rtcp(this->session, static_cast(data), reinterpret_cast(len)); + // Set all policy fields to 0. + std::memset(&policy, 0, sizeof(srtp_policy_t)); - if (DepLibSRTP::IsError(err)) { - MS_DEBUG_TAG(srtp, "srtp_unprotect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + switch (cryptoSuite) + { + case CryptoSuite::AES_CM_128_HMAC_SHA1_80: + { + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp); + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); - return false; - } + break; + } - return true; -} -}// namespace RTC + case CryptoSuite::AES_CM_128_HMAC_SHA1_32: + { + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); + // NOTE: Must be 80 for RTCP. + srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); + + break; + } + + case CryptoSuite::AEAD_AES_256_GCM: + { + srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp); + srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp); + + break; + } + + case CryptoSuite::AEAD_AES_128_GCM: + { + srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp); + srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp); + + break; + } + + default: + { + MS_ABORT("unknown SRTP crypto suite"); + } + } + + MS_ASSERT( + (int)keyLen == policy.rtp.cipher_key_len, + "given keyLen does not match policy.rtp.cipher_keyLen"); + + switch (type) + { + case Type::INBOUND: + policy.ssrc.type = ssrc_any_inbound; + break; + + case Type::OUTBOUND: + policy.ssrc.type = ssrc_any_outbound; + break; + } + + policy.ssrc.value = 0; + policy.key = key; + // Required for sending RTP retransmission without RTX. + policy.allow_repeat_tx = 1; + policy.window_size = 1024; + policy.next = nullptr; + + // Set the SRTP session. + srtp_err_status_t err = srtp_create(&this->session, &policy); + + if (DepLibSRTP::IsError(err)) + MS_THROW_ERROR("srtp_create() failed: %s", DepLibSRTP::GetErrorString(err)); + } + + SrtpSession::~SrtpSession() + { + MS_TRACE(); + + if (this->session != nullptr) + { + srtp_err_status_t err = srtp_dealloc(this->session); + + if (DepLibSRTP::IsError(err)) + MS_ABORT("srtp_dealloc() failed: %s", DepLibSRTP::GetErrorString(err)); + } + } + + bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len) + { + MS_TRACE(); + + // Ensure that the resulting SRTP packet fits into the encrypt buffer. + if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) + { + MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len); + + return false; + } + + std::memcpy(EncryptBuffer, *data, *len); + + srtp_err_status_t err = + srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) + { + MS_WARN_TAG(srtp, "srtp_protect() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + // Update the given data pointer. + *data = (const uint8_t*)EncryptBuffer; + + return true; + } + + bool SrtpSession::DecryptSrtp(uint8_t* data, size_t* len) + { + MS_TRACE(); + + srtp_err_status_t err = + srtp_unprotect(this->session, static_cast(data), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) + { + MS_DEBUG_TAG(srtp, "srtp_unprotect() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + return true; + } + + bool SrtpSession::EncryptRtcp(const uint8_t** data, size_t* len) + { + MS_TRACE(); + + // Ensure that the resulting SRTCP packet fits into the encrypt buffer. + if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) + { + MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len); + + return false; + } + + std::memcpy(EncryptBuffer, *data, *len); + + srtp_err_status_t err = srtp_protect_rtcp( + this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) + { + MS_WARN_TAG(srtp, "srtp_protect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + // Update the given data pointer. + *data = (const uint8_t*)EncryptBuffer; + + return true; + } + + bool SrtpSession::DecryptSrtcp(uint8_t* data, size_t* len) + { + MS_TRACE(); + + srtp_err_status_t err = + srtp_unprotect_rtcp(this->session, static_cast(data), reinterpret_cast(len)); + + if (DepLibSRTP::IsError(err)) + { + MS_DEBUG_TAG(srtp, "srtp_unprotect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err)); + + return false; + } + + return true; + } +} // namespace RTC diff --git a/webrtc/srtp_session.h b/webrtc/srtp_session.h index 95da5fbf..a2cc30de 100644 --- a/webrtc/srtp_session.h +++ b/webrtc/srtp_session.h @@ -1,54 +1,69 @@ #ifndef MS_RTC_SRTP_SESSION_HPP #define MS_RTC_SRTP_SESSION_HPP -#include "rtc_dtls_transport.h" #include "utils.h" #include #include -namespace RTC { +namespace RTC +{ + class DepLibSRTP { + public: + static void ClassInit(); + static void ClassDestroy(); + static bool IsError(srtp_err_status_t code) { return (code != srtp_err_status_ok); } + static const char *GetErrorString(srtp_err_status_t code) { + // This throws out_of_range if the given index is not in the vector. + return DepLibSRTP::errors.at(code); + } -class DepLibSRTP { -public: - static void ClassInit(); - static void ClassDestroy(); - static bool IsError(srtp_err_status_t code) { return (code != srtp_err_status_ok); } - static const char *GetErrorString(srtp_err_status_t code) { - // This throws out_of_range if the given index is not in the vector. - return DepLibSRTP::errors.at(code); - } + private: + static std::vector errors; + }; -private: - static std::vector errors; -}; + class SrtpSession + { + public: + enum class CryptoSuite + { + NONE = 0, + AES_CM_128_HMAC_SHA1_80 = 1, + AES_CM_128_HMAC_SHA1_32, + AEAD_AES_256_GCM, + AEAD_AES_128_GCM + }; -class SrtpSession { -public: -public: - enum class Type { INBOUND = 1, OUTBOUND }; + public: + enum class Type + { + INBOUND = 1, + OUTBOUND + }; -public: - static void ClassInit(); + public: + static void ClassInit(); -private: - static void OnSrtpEvent(srtp_event_data_t *data); + private: + static void OnSrtpEvent(srtp_event_data_t* data); -public: - SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t *key, size_t keyLen); - ~SrtpSession(); + public: + SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen); + ~SrtpSession(); -public: - bool EncryptRtp(const uint8_t **data, size_t *len); - bool DecryptSrtp(uint8_t *data, size_t *len); - bool EncryptRtcp(const uint8_t **data, size_t *len); - bool DecryptSrtcp(uint8_t *data, size_t *len); - void RemoveStream(uint32_t ssrc) { srtp_remove_stream(this->session, uint32_t{htonl(ssrc)}); } + public: + bool EncryptRtp(const uint8_t** data, size_t* len); + bool DecryptSrtp(uint8_t* data, size_t* len); + bool EncryptRtcp(const uint8_t** data, size_t* len); + bool DecryptSrtcp(uint8_t* data, size_t* len); + void RemoveStream(uint32_t ssrc) + { + srtp_remove_stream(this->session, uint32_t{ htonl(ssrc) }); + } -private: - bool is_init = false; - // Allocated by this. - srtp_t session{nullptr}; -}; -}// namespace RTC + private: + // Allocated by this. + srtp_t session{ nullptr }; + }; +} // namespace RTC #endif diff --git a/webrtc/stun_packet.cc b/webrtc/stun_packet.cc index 926f863b..1977f7f1 100644 --- a/webrtc/stun_packet.cc +++ b/webrtc/stun_packet.cc @@ -6,10 +6,79 @@ #include // std::snprintf() #include // std::memcmp(), std::memcpy() -#include "utils.h" - namespace RTC { +static const uint32_t crc32Table[] = +{ + 0x00000000, 0x77073096, 0xee0e612c, 0x990951ba, 0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3, + 0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988, 0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91, + 0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de, 0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7, + 0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec, 0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5, + 0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172, 0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b, + 0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940, 0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59, + 0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116, 0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f, + 0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924, 0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d, + 0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, 0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433, + 0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818, 0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01, + 0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e, 0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457, + 0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c, 0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65, + 0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2, 0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb, + 0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0, 0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9, + 0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086, 0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f, + 0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4, 0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad, + 0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a, 0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683, + 0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8, 0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1, + 0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe, 0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7, + 0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc, 0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5, + 0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252, 0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b, + 0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60, 0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79, + 0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236, 0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f, + 0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04, 0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d, + 0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a, 0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713, + 0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38, 0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21, + 0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e, 0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777, + 0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c, 0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45, + 0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2, 0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db, + 0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0, 0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9, + 0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6, 0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf, + 0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94, 0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d +}; + +inline uint32_t GetCRC32(const uint8_t *data, size_t size) { + uint32_t crc{0xFFFFFFFF}; + const uint8_t *p = data; + + while (size--) { + crc = crc32Table[(crc ^ *p++) & 0xFF] ^ (crc >> 8); + } + + return crc ^ ~0U; +} + +static std::string openssl_HMACsha1(const void *key, size_t key_len, const void *data, size_t data_len){ + std::string str; + str.resize(20); + unsigned int out_len; +#if defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) + //openssl 1.1.0新增api，老版本api作废 + HMAC_CTX *ctx = HMAC_CTX_new(); + HMAC_CTX_reset(ctx); + HMAC_Init_ex(ctx, key, (int)key_len, EVP_sha1(), NULL); + HMAC_Update(ctx, (unsigned char*)data, data_len); + HMAC_Final(ctx, (unsigned char *)str.data(), &out_len); + HMAC_CTX_reset(ctx); + HMAC_CTX_free(ctx); +#else + HMAC_CTX ctx; + HMAC_CTX_init(&ctx); + HMAC_Init_ex(&ctx, key, key_len, EVP_sha1(), NULL); + HMAC_Update(&ctx, (unsigned char*)data, data_len); + HMAC_Final(&ctx, (unsigned char *)str.data(), &out_len); + HMAC_CTX_cleanup(&ctx); +#endif //defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) + return str; +} + /* Class variables. */ const uint8_t StunPacket::kMagicCookie[] = {0x21, 0x12, 0xA4, 0x42}; @@ -258,7 +327,7 @@ StunPacket* StunPacket::Parse(const uint8_t* data, size_t len) { if (hasFingerprint) { // Compute the CRC32 of the received packet up to (but excluding) the // FINGERPRINT attribute and XOR it with 0x5354554e. - uint32_t computedFingerprint = Utils::Crypto::GetCRC32(data, fingerprintAttrPos) ^ 0x5354554e; + uint32_t computedFingerprint = GetCRC32(data, fingerprintAttrPos) ^ 0x5354554e; // Compare with the FINGERPRINT value in the packet. if (fingerprint != computedFingerprint) { @@ -290,79 +359,6 @@ StunPacket::~StunPacket() { // MS_TRACE(); } -void StunPacket::Dump() const { - // MS_TRACE(); - - // MS_DUMP(""); - - std::string klass; - switch (this->klass) { - case Class::REQUEST: - klass = "Request"; - break; - case Class::INDICATION: - klass = "Indication"; - break; - case Class::SUCCESS_RESPONSE: - klass = "SuccessResponse"; - break; - case Class::ERROR_RESPONSE: - klass = "ErrorResponse"; - break; - } - if (this->method == Method::BINDING) { - // MS_DUMP(" Binding %s", klass.c_str()); - } else { - // This prints the unknown method number. Example: TURN Allocate => 0x003. - // MS_DUMP(" %s with unknown method %#.3x", klass.c_str(), - // static_cast(this->method)); - } - // MS_DUMP(" size: %zu bytes", this->size); - - static char transactionId[25]; - - for (int i{0}; i < 12; ++i) { - // NOTE: n must be 3 because snprintf adds a \0 after printed chars. - std::snprintf(transactionId + (i * 2), 3, "%.2x", this->transactionId[i]); - } - // MS_DUMP(" transactionId: %s", transactionId); - if (this->errorCode != 0u) - // MS_DUMP(" errorCode: %" PRIu16, this->errorCode); - if (!this->username.empty()) - // MS_DUMP(" username: %s", this->username.c_str()); - if (this->priority != 0u) - // MS_DUMP(" priority: %" PRIu32, this->priority); - if (this->iceControlling != 0u) - // MS_DUMP(" iceControlling: %" PRIu64, this->iceControlling); - if (this->iceControlled != 0u) - // MS_DUMP(" iceControlled: %" PRIu64, this->iceControlled); - if (this->hasUseCandidate) - // MS_DUMP(" useCandidate"); - if (this->xorMappedAddress != nullptr) { - int family; - uint16_t port; - std::string ip; - - Utils::IP::GetAddressInfo(this->xorMappedAddress, family, ip, port); - - // MS_DUMP(" xorMappedAddress: %s : %" PRIu16, ip.c_str(), port); - } - if (this->messageIntegrity != nullptr) { - static char messageIntegrity[41]; - - for (int i{0}; i < 20; ++i) { - std::snprintf(messageIntegrity + (i * 2), 3, "%.2x", this->messageIntegrity[i]); - } - - // MS_DUMP(" messageIntegrity: %s", messageIntegrity); - } - if (this->hasFingerprint) { - } - // MS_DUMP(" has fingerprint"); - - // MS_DUMP(""); -} - StunPacket::Authentication StunPacket::CheckAuthentication(const std::string& localUsername, const std::string& localPassword) { // MS_TRACE(); @@ -402,13 +398,13 @@ StunPacket::Authentication StunPacket::CheckAuthentication(const std::string& lo Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20 - 8)); // Calculate the HMAC-SHA1 of the message according to MESSAGE-INTEGRITY rules. - const uint8_t* computedMessageIntegrity = Utils::Crypto::GetHmacShA1( - localPassword, this->data, (this->messageIntegrity - 4) - this->data); + auto computedMessageIntegrity = openssl_HMACsha1( + localPassword.data(),localPassword.size(), this->data, (this->messageIntegrity - 4) - this->data); Authentication result; // Compare the computed HMAC-SHA1 with the MESSAGE-INTEGRITY in the packet. - if (std::memcmp(this->messageIntegrity, computedMessageIntegrity, 20) == 0) + if (std::memcmp(this->messageIntegrity, computedMessageIntegrity.data(), computedMessageIntegrity.size()) == 0) result = Authentication::OK; else result = Authentication::UNAUTHORIZED; @@ -670,12 +666,11 @@ void StunPacket::Serialize(uint8_t* buffer) { Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20 - 8)); // Calculate the HMAC-SHA1 of the packet according to MESSAGE-INTEGRITY rules. - const uint8_t* computedMessageIntegrity = - Utils::Crypto::GetHmacShA1(this->password, buffer, pos); + auto computedMessageIntegrity = openssl_HMACsha1(this->password.data(), this->password.size(), buffer, pos); Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::MESSAGE_INTEGRITY)); Utils::Byte::Set2Bytes(buffer, pos + 2, 20); - std::memcpy(buffer + pos + 4, computedMessageIntegrity, 20); + std::memcpy(buffer + pos + 4, computedMessageIntegrity.data(), computedMessageIntegrity.size()); // Update the pointer. this->messageIntegrity = buffer + pos + 4; @@ -692,7 +687,7 @@ void StunPacket::Serialize(uint8_t* buffer) { if (addFingerprint) { // Compute the CRC32 of the packet up to (but excluding) the FINGERPRINT // attribute and XOR it with 0x5354554e. - uint32_t computedFingerprint = Utils::Crypto::GetCRC32(buffer, pos) ^ 0x5354554e; + uint32_t computedFingerprint = GetCRC32(buffer, pos) ^ 0x5354554e; Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::FINGERPRINT)); Utils::Byte::Set2Bytes(buffer, pos + 2, 4); diff --git a/webrtc/utils.cc b/webrtc/utils.cc deleted file mode 100644 index ab9f1fda..00000000 --- a/webrtc/utils.cc +++ /dev/null @@ -1,139 +0,0 @@ -#define MS_CLASS "Utils::Crypto" -// #define MS_LOG_DEV - -#include "utils.h" - -#include "openssl/sha.h" - -namespace Utils { -/* Static variables. */ - -uint32_t Crypto::seed; -HMAC_CTX *Crypto::hmacSha1Ctx{nullptr}; -uint8_t Crypto::hmacSha1Buffer[20];// SHA-1 result is 20 bytes long. -// clang-format off -const uint32_t Crypto::crc32Table[] = -{ - 0x00000000, 0x77073096, 0xee0e612c, 0x990951ba, 0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3, - 0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988, 0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91, - 0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de, 0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7, - 0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec, 0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5, - 0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172, 0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b, - 0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940, 0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59, - 0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116, 0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f, - 0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924, 0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d, - 0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, 0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433, - 0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818, 0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01, - 0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e, 0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457, - 0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c, 0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65, - 0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2, 0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb, - 0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0, 0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9, - 0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086, 0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f, - 0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4, 0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad, - 0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a, 0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683, - 0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8, 0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1, - 0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe, 0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7, - 0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc, 0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5, - 0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252, 0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b, - 0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60, 0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79, - 0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236, 0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f, - 0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04, 0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d, - 0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a, 0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713, - 0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38, 0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21, - 0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e, 0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777, - 0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c, 0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45, - 0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2, 0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db, - 0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0, 0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9, - 0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6, 0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf, - 0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94, 0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d -}; -// clang-format on - -/* Static methods. */ - -void Crypto::ClassInit() { - // MS_TRACE(); - - // Init the vrypto seed with a random number taken from the address - // of the seed variable itself (which is random). - Crypto::seed = static_cast(reinterpret_cast(std::addressof(Crypto::seed))); - - // Create an OpenSSL HMAC_CTX context for HMAC SHA1 calculation. - // Crypto::hmacSha1Ctx = HMAC_CTX_new(); - if (Crypto::hmacSha1Ctx == nullptr) { - Crypto::hmacSha1Ctx = HMAC_CTX_new(); - } -} - -void Crypto::ClassDestroy() { - // MS_TRACE(); - - if (Crypto::hmacSha1Ctx != nullptr) { - HMAC_CTX_free(Crypto::hmacSha1Ctx); - } -} - -const uint8_t *Crypto::GetHmacShA1(const std::string &key, const uint8_t *data, size_t len) { - // MS_TRACE(); - - size_t ret; - - ret = HMAC_Init_ex(Crypto::hmacSha1Ctx, key.c_str(), key.length(), EVP_sha1(), nullptr); - - // MS_ASSERT(ret == 1, "OpenSSL HMAC_Init_ex() failed with key '%s'", key.c_str()); - - ret = HMAC_Update(Crypto::hmacSha1Ctx, data, static_cast(len)); - /* - MS_ASSERT( - ret == 1, - "OpenSSL HMAC_Update() failed with key '%s' and data length %zu bytes", - key.c_str(), - len); - */ - uint32_t resultLen; - - ret = HMAC_Final(Crypto::hmacSha1Ctx, (uint8_t *) Crypto::hmacSha1Buffer, &resultLen); - - /* - MS_ASSERT( - ret == 1, "OpenSSL HMAC_Final() failed with key '%s' and data length %zu bytes", key.c_str(), - len); MS_ASSERT(resultLen == 20, "OpenSSL HMAC_Final() resultLen is %u instead of 20", resultLen); - */ - return Crypto::hmacSha1Buffer; -} -}// namespace Utils - -namespace Utils { - -static std::string inet_ntoa(struct in_addr in) { - char buf[20]; - unsigned char *p = (unsigned char *) &(in); - snprintf(buf, sizeof(buf), "%u.%u.%u.%u", p[0], p[1], p[2], p[3]); - return buf; -} - -void IP::GetAddressInfo(const struct sockaddr *addr, int &family, std::string &ip, uint16_t &port) { - char ipBuffer[INET6_ADDRSTRLEN + 1]; - - switch (addr->sa_family) { - case AF_INET: { - ip = Utils::inet_ntoa(reinterpret_cast(addr)->sin_addr); - port = static_cast(ntohs(reinterpret_cast(addr)->sin_port)); - break; - } - - case AF_INET6: { - port = static_cast(ntohs(reinterpret_cast(addr)->sin6_port)); - break; - } - - default: { - // MS_ABORT("unknown network family: %d", static_cast(addr->sa_family)); - } - } - - family = addr->sa_family; - ip.assign(ipBuffer); -} - -}// namespace Utils \ No newline at end of file diff --git a/webrtc/utils.h b/webrtc/utils.h index 1cd6ba9d..6b33b9f5 100644 --- a/webrtc/utils.h +++ b/webrtc/utils.h @@ -30,76 +30,6 @@ #include namespace Utils { -class IP { -public: - static int GetFamily(const char *ip, size_t ipLen); - static int GetFamily(const std::string &ip); - static void GetAddressInfo(const struct sockaddr *addr, int &family, std::string &ip, - uint16_t &port); - static bool CompareAddresses(const struct sockaddr *addr1, const struct sockaddr *addr2); - static struct sockaddr_storage CopyAddress(const struct sockaddr *addr); - static void NormalizeIp(std::string &ip); -}; - -/* Inline static methods. */ - -inline int IP::GetFamily(const std::string &ip) { return GetFamily(ip.c_str(), ip.size()); } - -inline bool IP::CompareAddresses(const struct sockaddr *addr1, const struct sockaddr *addr2) { - // Compare family. - if (addr1->sa_family != addr2->sa_family || - (addr1->sa_family != AF_INET && addr1->sa_family != AF_INET6)) { - return false; - } - - // Compare port. - if (reinterpret_cast(addr1)->sin_port != - reinterpret_cast(addr2)->sin_port) { - return false; - } - - // Compare IP. - switch (addr1->sa_family) { - case AF_INET: { - return (reinterpret_cast(addr1)->sin_addr.s_addr == - reinterpret_cast(addr2)->sin_addr.s_addr); - } - - case AF_INET6: { - return (std::memcmp( - std::addressof(reinterpret_cast(addr1)->sin6_addr), - std::addressof(reinterpret_cast(addr2)->sin6_addr), - 16) == 0 - ? true - : false); - } - - default: { - return false; - } - } -} - -inline struct sockaddr_storage IP::CopyAddress(const struct sockaddr *addr) { - struct sockaddr_storage copiedAddr; - - switch (addr->sa_family) { - case AF_INET: - std::memcpy(std::addressof(copiedAddr), addr, sizeof(struct sockaddr_in)); - break; - - case AF_INET6: - std::memcpy(std::addressof(copiedAddr), addr, sizeof(struct sockaddr_in6)); - break; - } - - return copiedAddr; -} - -class File { -public: - static void CheckFile(const char *file); -}; class Byte { public: @@ -181,138 +111,6 @@ inline uint16_t Byte::PadTo4Bytes(uint16_t size) { return size; } -inline uint32_t Byte::PadTo4Bytes(uint32_t size) { - // If size is not multiple of 32 bits then pad it. - if (size & 0x03) - return (size & 0xFFFFFFFC) + 4; - else - return size; -} - -class Bits { -public: - static size_t CountSetBits(const uint16_t mask); -}; - -/* Inline static methods. */ - -class Crypto { -public: - static void ClassInit(); - static void ClassDestroy(); - static uint32_t GetRandomUInt(uint32_t min, uint32_t max); - static const std::string GetRandomString(size_t len); - static uint32_t GetCRC32(const uint8_t *data, size_t size); - static const uint8_t *GetHmacShA1(const std::string &key, const uint8_t *data, size_t len); - -private: - static uint32_t seed; - static HMAC_CTX *hmacSha1Ctx; - static uint8_t hmacSha1Buffer[]; - static const uint32_t crc32Table[256]; -}; - -/* Inline static methods. */ - -inline uint32_t Crypto::GetRandomUInt(uint32_t min, uint32_t max) { - // NOTE: This is the original, but produces very small values. - // Crypto::seed = (214013 * Crypto::seed) + 2531011; - // return (((Crypto::seed>>16)&0x7FFF) % (max - min + 1)) + min; - - // This seems to produce better results. - Crypto::seed = uint32_t{((214013 * Crypto::seed) + 2531011)}; - - return (((Crypto::seed >> 4) & 0x7FFF7FFF) % (max - min + 1)) + min; -} - -inline const std::string Crypto::GetRandomString(size_t len) { - static char buffer[64]; - static const char chars[] = {'0', '1', '2', '3', '4', '5', '6', '7', '8', '9', 'a', 'b', - 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', - 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z'}; - - if (len > 64) len = 64; - - for (size_t i{0}; i < len; ++i) { - buffer[i] = chars[GetRandomUInt(0, sizeof(chars) - 1)]; - } - - return std::string(buffer, len); -} - -inline uint32_t Crypto::GetCRC32(const uint8_t *data, size_t size) { - uint32_t crc{0xFFFFFFFF}; - const uint8_t *p = data; - - while (size--) { - crc = Crypto::crc32Table[(crc ^ *p++) & 0xFF] ^ (crc >> 8); - } - - return crc ^ ~0U; -} - -class String { -public: - static void ToLowerCase(std::string &str); -}; - -inline void String::ToLowerCase(std::string &str) { - std::transform(str.begin(), str.end(), str.begin(), ::tolower); -} - -class Time { - // Seconds from Jan 1, 1900 to Jan 1, 1970. - static constexpr uint32_t UnixNtpOffset{0x83AA7E80}; - // NTP fractional unit. - static constexpr uint64_t NtpFractionalUnit{1LL << 32}; - -public: - struct Ntp { - uint32_t seconds; - uint32_t fractions; - }; - - static Time::Ntp TimeMs2Ntp(uint64_t ms); - static uint64_t Ntp2TimeMs(Time::Ntp ntp); - static bool IsNewerTimestamp(uint32_t timestamp, uint32_t prevTimestamp); - static uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2); -}; - -inline Time::Ntp Time::TimeMs2Ntp(uint64_t ms) { - Time::Ntp ntp;// NOLINT(cppcoreguidelines-pro-type-member-init) - - ntp.seconds = uint32_t(ms / 1000); - ntp.fractions = - static_cast((static_cast(ms % 1000) / 1000) * NtpFractionalUnit); - - return ntp; -} - -inline uint64_t Time::Ntp2TimeMs(Time::Ntp ntp) { - // clang-format off - return ( - static_cast(ntp.seconds) * 1000 + - static_cast(std::round((static_cast(ntp.fractions) * 1000) / NtpFractionalUnit)) - ); - // clang-format on -} - -inline bool Time::IsNewerTimestamp(uint32_t timestamp, uint32_t prevTimestamp) { - // Distinguish between elements that are exactly 0x80000000 apart. - // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true, - // IsNewer(t2,t1)=false - // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false. - if (static_cast(timestamp - prevTimestamp) == 0x80000000) - return timestamp > prevTimestamp; - - return timestamp != prevTimestamp && - static_cast(timestamp - prevTimestamp) < 0x80000000; -} - -inline uint32_t Time::LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) { - return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2; -} - }// namespace Utils #endif diff --git a/webrtc/webrtc_transport.cc b/webrtc/webrtc_transport.cc index a475f2ac..f064b6db 100644 --- a/webrtc/webrtc_transport.cc +++ b/webrtc/webrtc_transport.cc @@ -4,31 +4,81 @@ WebRtcTransport::WebRtcTransport() { static onceToken token([](){ - Utils::Crypto::ClassInit(); RTC::DtlsTransport::ClassInit(); RTC::DepLibSRTP::ClassInit(); RTC::SrtpSession::ClassInit(); }); - ice_server_ = std::make_shared(Utils::Crypto::GetRandomString(4), Utils::Crypto::GetRandomString(24)); - ice_server_->SetIceServerCompletedCB([this]() { - this->OnIceServerCompleted(); - }); - ice_server_->SetSendCB([this](char *buf, size_t len, struct sockaddr_in *remote_address) { - this->WritePacket(buf, len, remote_address); - }); - - // todo dtls服务器或客户端模式 - dtls_transport_ = std::make_shared(true); - dtls_transport_->SetHandshakeCompletedCB([this](std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite) { - this->OnDtlsCompleted(client_key, server_key, srtp_crypto_suite); - }); - dtls_transport_->SetOutPutCB([this](char *buf, size_t len) { this->WritePacket(buf, len); }); + dtls_transport_ = std::make_shared(EventPollerPool::Instance().getFirstPoller(), this); + ice_server_ = std::make_shared(this, makeRandStr(4), makeRandStr(24)); } WebRtcTransport::~WebRtcTransport() {} +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) { + onWrite((char *)packet->GetData(), packet->GetSize(), (struct sockaddr_in *)tuple); +} + +void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) { + InfoL; +} + +void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) { + InfoL; + dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); +} + +void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) { + InfoL; +} + +void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) { + InfoL; +} + +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void WebRtcTransport::OnDtlsTransportConnected( + const RTC::DtlsTransport *dtlsTransport, + RTC::SrtpSession::CryptoSuite srtpCryptoSuite, + uint8_t *srtpLocalKey, + size_t srtpLocalKeyLen, + uint8_t *srtpRemoteKey, + size_t srtpRemoteKeyLen, + std::string &remoteCert) { + InfoL; + srtp_session_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); + onDtlsConnected(); +} + +void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) { + onWrite((char *)data, len); +} + +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void WebRtcTransport::onWrite(const char *buf, size_t len){ + auto tuple = ice_server_->GetSelectedTuple(); + assert(tuple); + onWrite(buf, len, (struct sockaddr_in *)tuple); +} + std::string WebRtcTransport::GetLocalSdp() { + RTC::DtlsTransport::Fingerprint remote_fingerprint; + remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm("sha-256"); + remote_fingerprint.value = ""; + dtls_transport_->SetRemoteFingerprint(remote_fingerprint); + + string finger_print_sha256; + auto finger_prints = dtls_transport_->GetLocalFingerprints(); + for (size_t i = 0; i < finger_prints.size(); i++) { + if (finger_prints[i].algorithm == RTC::DtlsTransport::FingerprintAlgorithm::SHA256) { + finger_print_sha256 = finger_prints[i].value; + } + } + char sdp[1024 * 10] = {0}; auto ssrc = getSSRC(); auto ip = getIP(); @@ -60,22 +110,10 @@ std::string WebRtcTransport::GetLocalSdp() { "a=candidate:%s 1 udp %u %s %u typ %s\r\n", ip.c_str(), port, pt, ip.c_str(), ice_server_->GetUsernameFragment().c_str(),ice_server_->GetPassword().c_str(), - dtls_transport_->GetMyFingerprint().c_str(), pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); + finger_print_sha256.c_str(), pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); return sdp; } -void WebRtcTransport::OnIceServerCompleted() { - InfoL; - dtls_transport_->Start(); - onIceConnected(); -} - -void WebRtcTransport::OnDtlsCompleted(std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite) { - InfoL << client_key << " " << server_key << " " << (int)srtp_crypto_suite; - srtp_session_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtp_crypto_suite, (uint8_t *) client_key.c_str(), client_key.size()); - onDtlsCompleted(); -} - bool is_dtls(char *buf) { return ((*buf > 19) && (*buf < 64)); } @@ -90,25 +128,23 @@ bool is_rtcp(char *buf) { return ((header->pt >= 64) && (header->pt < 96)); } -void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, struct sockaddr_in *remote_address) { +void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple) { if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { - InfoL << "stun:" << hexdump(buf, len); RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len); if (packet == nullptr) { WarnL << "parse stun error" << std::endl; return; } - ice_server_->ProcessStunPacket(packet, remote_address); + ice_server_->ProcessStunPacket(packet, tuple); return; } - if (DtlsTransport::IsDtlsPacket(buf, len)) { - InfoL << "dtls:" << hexdump(buf, len); - dtls_transport_->InputData(buf, len); + if (is_dtls(buf)) { + dtls_transport_->ProcessDtlsData((uint8_t *)buf, len); return; } if (is_rtp(buf)) { RtpHeader *header = (RtpHeader *) buf; - InfoL << "rtp:" << header->dumpString(len); +// InfoL << "rtp:" << header->dumpString(len); return; } if (is_rtcp(buf)) { @@ -118,10 +154,6 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, struct sockaddr_i } } -void WebRtcTransport::WritePacket(char *buf, size_t len, struct sockaddr_in *remote_address) { - onWrite(buf, len, remote_address ? remote_address : (ice_server_ ? ice_server_->GetSelectAddr() : nullptr)); -} - void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { const uint8_t *p = (uint8_t *) buf; bool ret = false; @@ -129,7 +161,7 @@ void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { ret = srtp_session_->EncryptRtp(&p, &len); } if (ret) { - onWrite((char *) p, len, ice_server_->GetSelectAddr()); + onWrite((char *) p, len); } } @@ -139,8 +171,8 @@ WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) { _socket = Socket::createSocket(poller, false); //随机端口，绑定全部网卡 _socket->bindUdpSock(0); - _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len){ - OnInputDataPacket(buf->data(), buf->size(), (struct sockaddr_in*)addr); + _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable { + OnInputDataPacket(buf->data(), buf->size(), addr); }); } @@ -149,7 +181,7 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { _src = src; } -void WebRtcTransportImp::onDtlsCompleted() { +void WebRtcTransportImp::onDtlsConnected() { _reader = _src->getRing()->attach(_socket->getPoller(), true); weak_ptr weak_self = shared_from_this(); _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt){ @@ -167,14 +199,9 @@ void WebRtcTransportImp::onDtlsCompleted() { }); } -void WebRtcTransportImp::onIceConnected(){ - -} - void WebRtcTransportImp::onWrite(const char *buf, size_t len, struct sockaddr_in *dst) { auto ptr = BufferRaw::create(); ptr->assign(buf, len); -// InfoL << len << " " << SockUtil::inet_ntoa(dst->sin_addr) << " " << ntohs(dst->sin_port); _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr)); } @@ -201,15 +228,5 @@ std::string WebRtcTransportImp::getIP() const { /////////////////////////////////////////////////////////////////// -INSTANCE_IMP(WebRtcManager) - -WebRtcManager::WebRtcManager() { - -} - -WebRtcManager::~WebRtcManager() { - -} - diff --git a/webrtc/webrtc_transport.h b/webrtc/webrtc_transport.h index 2289e5f1..b9fcf347 100644 --- a/webrtc/webrtc_transport.h +++ b/webrtc/webrtc_transport.h @@ -3,12 +3,12 @@ #include #include -#include "dtls_transport.h" +#include "rtc_dtls_transport.h" #include "ice_server.h" #include "srtp_session.h" #include "stun_packet.h" -class WebRtcTransport { +class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: using Ptr = std::shared_ptr; WebRtcTransport(); @@ -22,13 +22,38 @@ public: /// \param buf /// \param len /// \param remote_address - void OnInputDataPacket(char *buf, size_t len, struct sockaddr_in *remote_address); + void OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple); /// 发送rtp /// \param buf /// \param len void WritRtpPacket(char *buf, size_t len); +protected: + // dtls相关的回调 + void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override {}; + void OnDtlsTransportConnected( + const RTC::DtlsTransport *dtlsTransport, + RTC::SrtpSession::CryptoSuite srtpCryptoSuite, + uint8_t *srtpLocalKey, + size_t srtpLocalKeyLen, + uint8_t *srtpRemoteKey, + size_t srtpRemoteKeyLen, + std::string &remoteCert) override; + + void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override {}; + void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override {}; + void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; + void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override {}; + +protected: + //ice相关的回调 + void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override; + void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override; + void OnIceServerConnected(const RTC::IceServer *iceServer) override; + void OnIceServerCompleted(const RTC::IceServer *iceServer) override; + void OnIceServerDisconnected(const RTC::IceServer *iceServer) override; + protected: /// 输出udp数据 /// \param buf @@ -39,17 +64,14 @@ protected: virtual uint16_t getPort() const = 0; virtual std::string getIP() const = 0; virtual int getPayloadType() const = 0; - virtual void onIceConnected() = 0; - virtual void onDtlsCompleted() = 0; + virtual void onDtlsConnected() = 0; private: - void OnIceServerCompleted(); - void OnDtlsCompleted(std::string client_key, std::string server_key, RTC::CryptoSuite srtp_crypto_suite); - void WritePacket(char *buf, size_t len, struct sockaddr_in *remote_address = nullptr); + void onWrite(const char *buf, size_t len); private: - IceServer::Ptr ice_server_; - DtlsTransport::Ptr dtls_transport_; + std::shared_ptr ice_server_; + std::shared_ptr dtls_transport_; std::shared_ptr srtp_session_; }; @@ -74,8 +96,7 @@ protected: uint32_t getSSRC() const override; uint16_t getPort() const override; std::string getIP() const override; - void onIceConnected() override; - void onDtlsCompleted() override; + void onDtlsConnected() override; private: Socket::Ptr _socket; @@ -83,16 +104,6 @@ private: RtspMediaSource::RingType::RingReader::Ptr _reader; }; -class WebRtcManager : public std::enable_shared_from_this { -public: - ~WebRtcManager(); - static WebRtcManager& Instance(); - -private: - WebRtcManager(); - -}; - diff --git a/www/webrtc/index.html b/www/webrtc/index.html index a9603957..fbc4c2d2 100644 --- a/www/webrtc/index.html +++ b/www/webrtc/index.html @@ -22,7 +22,7 @@

ip_address

- +
From bdf2783a6b559dd3119c58667730932e120762d2 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 09:13:10 +0800 Subject: [PATCH 004/218] =?UTF-8?q?=E8=BF=98=E5=8E=9F=E5=8E=9F=E5=A7=8B?= =?UTF-8?q?=E6=96=87=E4=BB=B6=E5=90=8D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 2 +- ...tc_dtls_transport.cc => DtlsTransport.cpp} | 2 +- ...rtc_dtls_transport.h => DtlsTransport.hpp} | 2 +- webrtc/{ice_server.cc => IceServer.cpp} | 2 +- webrtc/{ice_server.h => IceServer.hpp} | 2 +- webrtc/{srtp_session.cc => SrtpSession.cpp} | 2 +- webrtc/{srtp_session.h => SrtpSession.hpp} | 2 +- webrtc/StunPacket.cpp | 862 ++++++++++++++++++ webrtc/StunPacket.hpp | 195 ++++ webrtc/{utils.h => Utils.hpp} | 0 ...ebrtc_transport.cc => WebRtcTransport.cpp} | 2 +- .../{webrtc_transport.h => WebRtcTransport.h} | 8 +- webrtc/logger.h | 3 - webrtc/stun_packet.cc | 705 -------------- webrtc/stun_packet.h | 179 ---- 15 files changed, 1069 insertions(+), 899 deletions(-) rename webrtc/{rtc_dtls_transport.cc => DtlsTransport.cpp} (99%) rename webrtc/{rtc_dtls_transport.h => DtlsTransport.hpp} (99%) rename webrtc/{ice_server.cc => IceServer.cpp} (99%) rename webrtc/{ice_server.h => IceServer.hpp} (99%) rename webrtc/{srtp_session.cc => SrtpSession.cpp} (99%) rename webrtc/{srtp_session.h => SrtpSession.hpp} (98%) create mode 100644 webrtc/StunPacket.cpp create mode 100644 webrtc/StunPacket.hpp rename webrtc/{utils.h => Utils.hpp} (100%) rename webrtc/{webrtc_transport.cc => WebRtcTransport.cpp} (99%) rename webrtc/{webrtc_transport.h => WebRtcTransport.h} (97%) delete mode 100644 webrtc/stun_packet.cc delete mode 100644 webrtc/stun_packet.h diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 4d0dba28..3eacb1eb 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -36,7 +36,7 @@ #include "Rtp/RtpServer.h" #endif #ifdef ENABLE_WEBRTC -#include "../webrtc/webrtc_transport.h" +#include "../webrtc/WebRtcTransport.h" #endif using namespace toolkit; diff --git a/webrtc/rtc_dtls_transport.cc b/webrtc/DtlsTransport.cpp similarity index 99% rename from webrtc/rtc_dtls_transport.cc rename to webrtc/DtlsTransport.cpp index e66f6469..1c001a05 100644 --- a/webrtc/rtc_dtls_transport.cc +++ b/webrtc/DtlsTransport.cpp @@ -1,7 +1,7 @@ #define MS_CLASS "RTC::DtlsTransport" // #define MS_LOG_DEV_LEVEL 3 -#include "rtc_dtls_transport.h" +#include "DtlsTransport.hpp" #include "logger.h" #include #include diff --git a/webrtc/rtc_dtls_transport.h b/webrtc/DtlsTransport.hpp similarity index 99% rename from webrtc/rtc_dtls_transport.h rename to webrtc/DtlsTransport.hpp index b3e58b1d..07359583 100644 --- a/webrtc/rtc_dtls_transport.h +++ b/webrtc/DtlsTransport.hpp @@ -1,7 +1,7 @@ #ifndef MS_RTC_DTLS_TRANSPORT_HPP #define MS_RTC_DTLS_TRANSPORT_HPP -#include "srtp_session.h" +#include "SrtpSession.hpp" #include #include #include diff --git a/webrtc/ice_server.cc b/webrtc/IceServer.cpp similarity index 99% rename from webrtc/ice_server.cc rename to webrtc/IceServer.cpp index 7dc64ec5..38963c58 100644 --- a/webrtc/ice_server.cc +++ b/webrtc/IceServer.cpp @@ -2,7 +2,7 @@ // #define MS_LOG_DEV_LEVEL 3 #include -#include "ice_server.h" +#include "IceServer.hpp" namespace RTC { diff --git a/webrtc/ice_server.h b/webrtc/IceServer.hpp similarity index 99% rename from webrtc/ice_server.h rename to webrtc/IceServer.hpp index 437d9d9f..ad5d93eb 100644 --- a/webrtc/ice_server.h +++ b/webrtc/IceServer.hpp @@ -1,7 +1,7 @@ #ifndef MS_RTC_ICE_SERVER_HPP #define MS_RTC_ICE_SERVER_HPP -#include "stun_packet.h" +#include "StunPacket.hpp" #include "logger.h" #include #include diff --git a/webrtc/srtp_session.cc b/webrtc/SrtpSession.cpp similarity index 99% rename from webrtc/srtp_session.cc rename to webrtc/SrtpSession.cpp index fd8de70d..13eeebd4 100644 --- a/webrtc/srtp_session.cc +++ b/webrtc/SrtpSession.cpp @@ -1,7 +1,7 @@ #define MS_CLASS "RTC::SrtpSession" // #define MS_LOG_DEV_LEVEL 3 -#include "srtp_session.h" +#include "SrtpSession.hpp" #include // std::memset(), std::memcpy() #include "logger.h" diff --git a/webrtc/srtp_session.h b/webrtc/SrtpSession.hpp similarity index 98% rename from webrtc/srtp_session.h rename to webrtc/SrtpSession.hpp index a2cc30de..3a4c32f4 100644 --- a/webrtc/srtp_session.h +++ b/webrtc/SrtpSession.hpp @@ -1,7 +1,7 @@ #ifndef MS_RTC_SRTP_SESSION_HPP #define MS_RTC_SRTP_SESSION_HPP -#include "utils.h" +#include "Utils.hpp" #include #include diff --git a/webrtc/StunPacket.cpp b/webrtc/StunPacket.cpp new file mode 100644 index 00000000..019fadd9 --- /dev/null +++ b/webrtc/StunPacket.cpp @@ -0,0 +1,862 @@ +#define MS_CLASS "RTC::StunPacket" +// #define MS_LOG_DEV_LEVEL 3 + +#include "StunPacket.hpp" +#include // std::snprintf() +#include // std::memcmp(), std::memcpy() + +namespace RTC +{ + static const uint32_t crc32Table[] = + { + 0x00000000, 0x77073096, 0xee0e612c, 0x990951ba, 0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3, + 0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988, 0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91, + 0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de, 0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7, + 0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec, 0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5, + 0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172, 0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b, + 0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940, 0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59, + 0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116, 0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f, + 0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924, 0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d, + 0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, 0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433, + 0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818, 0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01, + 0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e, 0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457, + 0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c, 0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65, + 0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2, 0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb, + 0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0, 0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9, + 0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086, 0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f, + 0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4, 0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad, + 0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a, 0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683, + 0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8, 0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1, + 0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe, 0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7, + 0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc, 0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5, + 0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252, 0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b, + 0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60, 0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79, + 0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236, 0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f, + 0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04, 0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d, + 0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a, 0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713, + 0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38, 0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21, + 0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e, 0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777, + 0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c, 0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45, + 0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2, 0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db, + 0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0, 0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9, + 0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6, 0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf, + 0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94, 0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d + }; + + inline uint32_t GetCRC32(const uint8_t *data, size_t size) { + uint32_t crc{0xFFFFFFFF}; + const uint8_t *p = data; + + while (size--) { + crc = crc32Table[(crc ^ *p++) & 0xFF] ^ (crc >> 8); + } + + return crc ^ ~0U; + } + + static std::string openssl_HMACsha1(const void *key, size_t key_len, const void *data, size_t data_len){ + std::string str; + str.resize(20); + unsigned int out_len; +#if defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) + //openssl 1.1.0新增api，老版本api作废 + HMAC_CTX *ctx = HMAC_CTX_new(); + HMAC_CTX_reset(ctx); + HMAC_Init_ex(ctx, key, (int)key_len, EVP_sha1(), NULL); + HMAC_Update(ctx, (unsigned char*)data, data_len); + HMAC_Final(ctx, (unsigned char *)str.data(), &out_len); + HMAC_CTX_reset(ctx); + HMAC_CTX_free(ctx); +#else + HMAC_CTX ctx; + HMAC_CTX_init(&ctx); + HMAC_Init_ex(&ctx, key, key_len, EVP_sha1(), NULL); + HMAC_Update(&ctx, (unsigned char*)data, data_len); + HMAC_Final(&ctx, (unsigned char *)str.data(), &out_len); + HMAC_CTX_cleanup(&ctx); +#endif //defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) + return str; + } + + /* Class variables. */ + + const uint8_t StunPacket::magicCookie[] = { 0x21, 0x12, 0xA4, 0x42 }; + + /* Class methods. */ + + StunPacket* StunPacket::Parse(const uint8_t* data, size_t len) + { + MS_TRACE(); + + if (!StunPacket::IsStun(data, len)) + return nullptr; + + /* + The message type field is decomposed further into the following + structure: + + 0 1 + 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ + |M |M |M|M|M|C|M|M|M|C|M|M|M|M| + |11|10|9|8|7|1|6|5|4|0|3|2|1|0| + +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ + + Figure 3: Format of STUN Message Type Field + + Here the bits in the message type field are shown as most significant + (M11) through least significant (M0). M11 through M0 represent a 12- + bit encoding of the method. C1 and C0 represent a 2-bit encoding of + the class. + */ + + // Get type field. + uint16_t msgType = Utils::Byte::Get2Bytes(data, 0); + + // Get length field. + uint16_t msgLength = Utils::Byte::Get2Bytes(data, 2); + + // length field must be total size minus header's 20 bytes, and must be multiple of 4 Bytes. + if ((static_cast(msgLength) != len - 20) || ((msgLength & 0x03) != 0)) + { + MS_WARN_TAG( + ice, + "length field + 20 does not match total size (or it is not multiple of 4 bytes), " + "packet discarded"); + + return nullptr; + } + + // Get STUN method. + uint16_t msgMethod = (msgType & 0x000f) | ((msgType & 0x00e0) >> 1) | ((msgType & 0x3E00) >> 2); + + // Get STUN class. + uint16_t msgClass = ((data[0] & 0x01) << 1) | ((data[1] & 0x10) >> 4); + + // Create a new StunPacket (data + 8 points to the received TransactionID field). + auto* packet = new StunPacket( + static_cast(msgClass), static_cast(msgMethod), data + 8, data, len); + + /* + STUN Attributes + + After the STUN header are zero or more attributes. Each attribute + MUST be TLV encoded, with a 16-bit type, 16-bit length, and value. + Each STUN attribute MUST end on a 32-bit boundary. As mentioned + above, all fields in an attribute are transmitted most significant + bit first. + + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | Type | Length | + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | Value (variable) .... + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + + // Start looking for attributes after STUN header (Byte #20). + size_t pos{ 20 }; + // Flags (positions) for special MESSAGE-INTEGRITY and FINGERPRINT attributes. + bool hasMessageIntegrity{ false }; + bool hasFingerprint{ false }; + size_t fingerprintAttrPos; // Will point to the beginning of the attribute. + uint32_t fingerprint; // Holds the value of the FINGERPRINT attribute. + + // Ensure there are at least 4 remaining bytes (attribute with 0 length). + while (pos + 4 <= len) + { + // Get the attribute type. + auto attrType = static_cast(Utils::Byte::Get2Bytes(data, pos)); + + // Get the attribute length. + uint16_t attrLength = Utils::Byte::Get2Bytes(data, pos + 2); + + // Ensure the attribute length is not greater than the remaining size. + if ((pos + 4 + attrLength) > len) + { + MS_WARN_TAG(ice, "the attribute length exceeds the remaining size, packet discarded"); + + delete packet; + return nullptr; + } + + // FINGERPRINT must be the last attribute. + if (hasFingerprint) + { + MS_WARN_TAG(ice, "attribute after FINGERPRINT is not allowed, packet discarded"); + + delete packet; + return nullptr; + } + + // After a MESSAGE-INTEGRITY attribute just FINGERPRINT is allowed. + if (hasMessageIntegrity && attrType != Attribute::FINGERPRINT) + { + MS_WARN_TAG( + ice, + "attribute after MESSAGE-INTEGRITY other than FINGERPRINT is not allowed, " + "packet discarded"); + + delete packet; + return nullptr; + } + + const uint8_t* attrValuePos = data + pos + 4; + + switch (attrType) + { + case Attribute::USERNAME: + { + packet->SetUsername( + reinterpret_cast(attrValuePos), static_cast(attrLength)); + + break; + } + + case Attribute::PRIORITY: + { + // Ensure attribute length is 4 bytes. + if (attrLength != 4) + { + MS_WARN_TAG(ice, "attribute PRIORITY must be 4 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetPriority(Utils::Byte::Get4Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::ICE_CONTROLLING: + { + // Ensure attribute length is 8 bytes. + if (attrLength != 8) + { + MS_WARN_TAG(ice, "attribute ICE-CONTROLLING must be 8 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetIceControlling(Utils::Byte::Get8Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::ICE_CONTROLLED: + { + // Ensure attribute length is 8 bytes. + if (attrLength != 8) + { + MS_WARN_TAG(ice, "attribute ICE-CONTROLLED must be 8 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetIceControlled(Utils::Byte::Get8Bytes(attrValuePos, 0)); + + break; + } + + case Attribute::USE_CANDIDATE: + { + // Ensure attribute length is 0 bytes. + if (attrLength != 0) + { + MS_WARN_TAG(ice, "attribute USE-CANDIDATE must be 0 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + packet->SetUseCandidate(); + + break; + } + + case Attribute::MESSAGE_INTEGRITY: + { + // Ensure attribute length is 20 bytes. + if (attrLength != 20) + { + MS_WARN_TAG(ice, "attribute MESSAGE-INTEGRITY must be 20 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + hasMessageIntegrity = true; + packet->SetMessageIntegrity(attrValuePos); + + break; + } + + case Attribute::FINGERPRINT: + { + // Ensure attribute length is 4 bytes. + if (attrLength != 4) + { + MS_WARN_TAG(ice, "attribute FINGERPRINT must be 4 bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + hasFingerprint = true; + fingerprintAttrPos = pos; + fingerprint = Utils::Byte::Get4Bytes(attrValuePos, 0); + packet->SetFingerprint(); + + break; + } + + case Attribute::ERROR_CODE: + { + // Ensure attribute length >= 4bytes. + if (attrLength < 4) + { + MS_WARN_TAG(ice, "attribute ERROR-CODE must be >= 4bytes length, packet discarded"); + + delete packet; + return nullptr; + } + + uint8_t errorClass = Utils::Byte::Get1Byte(attrValuePos, 2); + uint8_t errorNumber = Utils::Byte::Get1Byte(attrValuePos, 3); + auto errorCode = static_cast(errorClass * 100 + errorNumber); + + packet->SetErrorCode(errorCode); + + break; + } + + default:; + } + + // Set next attribute position. + pos = + static_cast(Utils::Byte::PadTo4Bytes(static_cast(pos + 4 + attrLength))); + } + + // Ensure current position matches the total length. + if (pos != len) + { + MS_WARN_TAG(ice, "computed packet size does not match total size, packet discarded"); + + delete packet; + return nullptr; + } + + // If it has FINGERPRINT attribute then verify it. + if (hasFingerprint) + { + // Compute the CRC32 of the received packet up to (but excluding) the + // FINGERPRINT attribute and XOR it with 0x5354554e. + uint32_t computedFingerprint = GetCRC32(data, fingerprintAttrPos) ^ 0x5354554e; + + // Compare with the FINGERPRINT value in the packet. + if (fingerprint != computedFingerprint) + { + MS_WARN_TAG( + ice, + "computed FINGERPRINT value does not match the value in the packet, " + "packet discarded"); + + delete packet; + return nullptr; + } + } + + return packet; + } + + /* Instance methods. */ + + StunPacket::StunPacket( + Class klass, Method method, const uint8_t* transactionId, const uint8_t* data, size_t size) + : klass(klass), method(method), transactionId(transactionId), data(const_cast(data)), + size(size) + { + MS_TRACE(); + } + + StunPacket::~StunPacket() + { + MS_TRACE(); + } + +#if 0 + void StunPacket::Dump() const + { + MS_TRACE(); + + MS_DUMP(""); + + std::string klass; + switch (this->klass) + { + case Class::REQUEST: + klass = "Request"; + break; + case Class::INDICATION: + klass = "Indication"; + break; + case Class::SUCCESS_RESPONSE: + klass = "SuccessResponse"; + break; + case Class::ERROR_RESPONSE: + klass = "ErrorResponse"; + break; + } + if (this->method == Method::BINDING) + { + MS_DUMP(" Binding %s", klass.c_str()); + } + else + { + // This prints the unknown method number. Example: TURN Allocate => 0x003. + MS_DUMP(" %s with unknown method %#.3x", klass.c_str(), static_cast(this->method)); + } + MS_DUMP(" size: %zu bytes", this->size); + + static char transactionId[25]; + + for (int i{ 0 }; i < 12; ++i) + { + // NOTE: n must be 3 because snprintf adds a \0 after printed chars. + std::snprintf(transactionId + (i * 2), 3, "%.2x", this->transactionId[i]); + } + MS_DUMP(" transactionId: %s", transactionId); + if (this->errorCode != 0u) + MS_DUMP(" errorCode: %" PRIu16, this->errorCode); + if (!this->username.empty()) + MS_DUMP(" username: %s", this->username.c_str()); + if (this->priority != 0u) + MS_DUMP(" priority: %" PRIu32, this->priority); + if (this->iceControlling != 0u) + MS_DUMP(" iceControlling: %" PRIu64, this->iceControlling); + if (this->iceControlled != 0u) + MS_DUMP(" iceControlled: %" PRIu64, this->iceControlled); + if (this->hasUseCandidate) + MS_DUMP(" useCandidate"); + if (this->xorMappedAddress != nullptr) + { + int family; + uint16_t port; + std::string ip; + + Utils::IP::GetAddressInfo(this->xorMappedAddress, family, ip, port); + + MS_DUMP(" xorMappedAddress: %s : %" PRIu16, ip.c_str(), port); + } + if (this->messageIntegrity != nullptr) + { + static char messageIntegrity[41]; + + for (int i{ 0 }; i < 20; ++i) + { + std::snprintf(messageIntegrity + (i * 2), 3, "%.2x", this->messageIntegrity[i]); + } + + MS_DUMP(" messageIntegrity: %s", messageIntegrity); + } + if (this->hasFingerprint) + MS_DUMP(" has fingerprint"); + + MS_DUMP(""); + } +#endif + + StunPacket::Authentication StunPacket::CheckAuthentication( + const std::string& localUsername, const std::string& localPassword) + { + MS_TRACE(); + + switch (this->klass) + { + case Class::REQUEST: + case Class::INDICATION: + { + // Both USERNAME and MESSAGE-INTEGRITY must be present. + if (!this->messageIntegrity || this->username.empty()) + return Authentication::BAD_REQUEST; + + // Check that USERNAME attribute begins with our local username plus ":". + size_t localUsernameLen = localUsername.length(); + + if ( + this->username.length() <= localUsernameLen || this->username.at(localUsernameLen) != ':' || + (this->username.compare(0, localUsernameLen, localUsername) != 0)) + { + return Authentication::UNAUTHORIZED; + } + + break; + } + // This method cannot check authentication in received responses (as we + // are ICE-Lite and don't generate requests). + case Class::SUCCESS_RESPONSE: + case Class::ERROR_RESPONSE: + { + MS_ERROR("cannot check authentication for a STUN response"); + + return Authentication::BAD_REQUEST; + } + } + + // If there is FINGERPRINT it must be discarded for MESSAGE-INTEGRITY calculation, + // so the header length field must be modified (and later restored). + if (this->hasFingerprint) + // Set the header length field: full size - header length (20) - FINGERPRINT length (8). + Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20 - 8)); + + // Calculate the HMAC-SHA1 of the message according to MESSAGE-INTEGRITY rules. + auto computedMessageIntegrity = openssl_HMACsha1( + localPassword.data(),localPassword.size(), this->data, (this->messageIntegrity - 4) - this->data); + + Authentication result; + + // Compare the computed HMAC-SHA1 with the MESSAGE-INTEGRITY in the packet. + if (std::memcmp(this->messageIntegrity, computedMessageIntegrity.data(), computedMessageIntegrity.size()) == 0) + result = Authentication::OK; + else + result = Authentication::UNAUTHORIZED; + + // Restore the header length field. + if (this->hasFingerprint) + Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20)); + + return result; + } + + StunPacket* StunPacket::CreateSuccessResponse() + { + MS_TRACE(); + + MS_ASSERT( + this->klass == Class::REQUEST, + "attempt to create a success response for a non Request STUN packet"); + + return new StunPacket(Class::SUCCESS_RESPONSE, this->method, this->transactionId, nullptr, 0); + } + + StunPacket* StunPacket::CreateErrorResponse(uint16_t errorCode) + { + MS_TRACE(); + + MS_ASSERT( + this->klass == Class::REQUEST, + "attempt to create an error response for a non Request STUN packet"); + + auto* response = + new StunPacket(Class::ERROR_RESPONSE, this->method, this->transactionId, nullptr, 0); + + response->SetErrorCode(errorCode); + + return response; + } + + void StunPacket::Authenticate(const std::string& password) + { + // Just for Request, Indication and SuccessResponse messages. + if (this->klass == Class::ERROR_RESPONSE) + { + MS_ERROR("cannot set password for ErrorResponse messages"); + + return; + } + + this->password = password; + } + + void StunPacket::Serialize(uint8_t* buffer) + { + MS_TRACE(); + + // Some useful variables. + uint16_t usernamePaddedLen{ 0 }; + uint16_t xorMappedAddressPaddedLen{ 0 }; + bool addXorMappedAddress = + ((this->xorMappedAddress != nullptr) && this->method == StunPacket::Method::BINDING && + this->klass == Class::SUCCESS_RESPONSE); + bool addErrorCode = ((this->errorCode != 0u) && this->klass == Class::ERROR_RESPONSE); + bool addMessageIntegrity = (this->klass != Class::ERROR_RESPONSE && !this->password.empty()); + bool addFingerprint{ true }; // Do always. + + // Update data pointer. + this->data = buffer; + + // First calculate the total required size for the entire packet. + this->size = 20; // Header. + + if (!this->username.empty()) + { + usernamePaddedLen = Utils::Byte::PadTo4Bytes(static_cast(this->username.length())); + this->size += 4 + usernamePaddedLen; + } + + if (this->priority != 0u) + this->size += 4 + 4; + + if (this->iceControlling != 0u) + this->size += 4 + 8; + + if (this->iceControlled != 0u) + this->size += 4 + 8; + + if (this->hasUseCandidate) + this->size += 4; + + if (addXorMappedAddress) + { + switch (this->xorMappedAddress->sa_family) + { + case AF_INET: + { + xorMappedAddressPaddedLen = 8; + this->size += 4 + 8; + + break; + } + + case AF_INET6: + { + xorMappedAddressPaddedLen = 20; + this->size += 4 + 20; + + break; + } + + default: + { + MS_ERROR("invalid inet family in XOR-MAPPED-ADDRESS attribute"); + + addXorMappedAddress = false; + } + } + } + + if (addErrorCode) + this->size += 4 + 4; + + if (addMessageIntegrity) + this->size += 4 + 20; + + if (addFingerprint) + this->size += 4 + 4; + + // Merge class and method fields into type. + uint16_t typeField = (static_cast(this->method) & 0x0f80) << 2; + + typeField |= (static_cast(this->method) & 0x0070) << 1; + typeField |= (static_cast(this->method) & 0x000f); + typeField |= (static_cast(this->klass) & 0x02) << 7; + typeField |= (static_cast(this->klass) & 0x01) << 4; + + // Set type field. + Utils::Byte::Set2Bytes(buffer, 0, typeField); + // Set length field. + Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size) - 20); + // Set magic cookie. + std::memcpy(buffer + 4, StunPacket::magicCookie, 4); + // Set TransactionId field. + std::memcpy(buffer + 8, this->transactionId, 12); + // Update the transaction ID pointer. + this->transactionId = buffer + 8; + // Add atributes. + size_t pos{ 20 }; + + // Add USERNAME. + if (usernamePaddedLen != 0u) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USERNAME)); + Utils::Byte::Set2Bytes(buffer, pos + 2, static_cast(this->username.length())); + std::memcpy(buffer + pos + 4, this->username.c_str(), this->username.length()); + pos += 4 + usernamePaddedLen; + } + + // Add PRIORITY. + if (this->priority != 0u) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::PRIORITY)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + Utils::Byte::Set4Bytes(buffer, pos + 4, this->priority); + pos += 4 + 4; + } + + // Add ICE-CONTROLLING. + if (this->iceControlling != 0u) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLING)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 8); + Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlling); + pos += 4 + 8; + } + + // Add ICE-CONTROLLED. + if (this->iceControlled != 0u) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLED)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 8); + Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlled); + pos += 4 + 8; + } + + // Add USE-CANDIDATE. + if (this->hasUseCandidate) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USE_CANDIDATE)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 0); + pos += 4; + } + + // Add XOR-MAPPED-ADDRESS + if (addXorMappedAddress) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::XOR_MAPPED_ADDRESS)); + Utils::Byte::Set2Bytes(buffer, pos + 2, xorMappedAddressPaddedLen); + + uint8_t* attrValue = buffer + pos + 4; + + switch (this->xorMappedAddress->sa_family) + { + case AF_INET: + { + // Set first byte to 0. + attrValue[0] = 0; + // Set inet family. + attrValue[1] = 0x01; + // Set port and XOR it. + std::memcpy( + attrValue + 2, + &(reinterpret_cast(this->xorMappedAddress))->sin_port, + 2); + attrValue[2] ^= StunPacket::magicCookie[0]; + attrValue[3] ^= StunPacket::magicCookie[1]; + // Set address and XOR it. + std::memcpy( + attrValue + 4, + &(reinterpret_cast(this->xorMappedAddress))->sin_addr.s_addr, + 4); + attrValue[4] ^= StunPacket::magicCookie[0]; + attrValue[5] ^= StunPacket::magicCookie[1]; + attrValue[6] ^= StunPacket::magicCookie[2]; + attrValue[7] ^= StunPacket::magicCookie[3]; + + pos += 4 + 8; + + break; + } + + case AF_INET6: + { + // Set first byte to 0. + attrValue[0] = 0; + // Set inet family. + attrValue[1] = 0x02; + // Set port and XOR it. + std::memcpy( + attrValue + 2, + &(reinterpret_cast(this->xorMappedAddress))->sin6_port, + 2); + attrValue[2] ^= StunPacket::magicCookie[0]; + attrValue[3] ^= StunPacket::magicCookie[1]; + // Set address and XOR it. + std::memcpy( + attrValue + 4, + &(reinterpret_cast(this->xorMappedAddress))->sin6_addr.s6_addr, + 16); + attrValue[4] ^= StunPacket::magicCookie[0]; + attrValue[5] ^= StunPacket::magicCookie[1]; + attrValue[6] ^= StunPacket::magicCookie[2]; + attrValue[7] ^= StunPacket::magicCookie[3]; + attrValue[8] ^= this->transactionId[0]; + attrValue[9] ^= this->transactionId[1]; + attrValue[10] ^= this->transactionId[2]; + attrValue[11] ^= this->transactionId[3]; + attrValue[12] ^= this->transactionId[4]; + attrValue[13] ^= this->transactionId[5]; + attrValue[14] ^= this->transactionId[6]; + attrValue[15] ^= this->transactionId[7]; + attrValue[16] ^= this->transactionId[8]; + attrValue[17] ^= this->transactionId[9]; + attrValue[18] ^= this->transactionId[10]; + attrValue[19] ^= this->transactionId[11]; + + pos += 4 + 20; + + break; + } + } + } + + // Add ERROR-CODE. + if (addErrorCode) + { + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ERROR_CODE)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + + auto codeClass = static_cast(this->errorCode / 100); + uint8_t codeNumber = static_cast(this->errorCode) - (codeClass * 100); + + Utils::Byte::Set2Bytes(buffer, pos + 4, 0); + Utils::Byte::Set1Byte(buffer, pos + 6, codeClass); + Utils::Byte::Set1Byte(buffer, pos + 7, codeNumber); + pos += 4 + 4; + } + + // Add MESSAGE-INTEGRITY. + if (addMessageIntegrity) + { + // Ignore FINGERPRINT. + if (addFingerprint) + Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20 - 8)); + + // Calculate the HMAC-SHA1 of the packet according to MESSAGE-INTEGRITY rules. + auto computedMessageIntegrity = openssl_HMACsha1(this->password.data(), this->password.size(), buffer, pos); + + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::MESSAGE_INTEGRITY)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 20); + std::memcpy(buffer + pos + 4, computedMessageIntegrity.data(), computedMessageIntegrity.size()); + + // Update the pointer. + this->messageIntegrity = buffer + pos + 4; + pos += 4 + 20; + + // Restore length field. + if (addFingerprint) + Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20)); + } + else + { + // Unset the pointer (if it was set). + this->messageIntegrity = nullptr; + } + + // Add FINGERPRINT. + if (addFingerprint) + { + // Compute the CRC32 of the packet up to (but excluding) the FINGERPRINT + // attribute and XOR it with 0x5354554e. + uint32_t computedFingerprint = GetCRC32(buffer, pos) ^ 0x5354554e; + + Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::FINGERPRINT)); + Utils::Byte::Set2Bytes(buffer, pos + 2, 4); + Utils::Byte::Set4Bytes(buffer, pos + 4, computedFingerprint); + pos += 4 + 4; + + // Set flag. + this->hasFingerprint = true; + } + else + { + this->hasFingerprint = false; + } + + MS_ASSERT(pos == this->size, "pos != this->size"); + } +} // namespace RTC diff --git a/webrtc/StunPacket.hpp b/webrtc/StunPacket.hpp new file mode 100644 index 00000000..8f65aab6 --- /dev/null +++ b/webrtc/StunPacket.hpp @@ -0,0 +1,195 @@ +#ifndef MS_RTC_STUN_PACKET_HPP +#define MS_RTC_STUN_PACKET_HPP + + +#include "logger.h" +#include "Utils.hpp" +#include + +namespace RTC +{ + class StunPacket + { + public: + // STUN message class. + enum class Class : uint16_t + { + REQUEST = 0, + INDICATION = 1, + SUCCESS_RESPONSE = 2, + ERROR_RESPONSE = 3 + }; + + // STUN message method. + enum class Method : uint16_t + { + BINDING = 1 + }; + + // Attribute type. + enum class Attribute : uint16_t + { + MAPPED_ADDRESS = 0x0001, + USERNAME = 0x0006, + MESSAGE_INTEGRITY = 0x0008, + ERROR_CODE = 0x0009, + UNKNOWN_ATTRIBUTES = 0x000A, + REALM = 0x0014, + NONCE = 0x0015, + XOR_MAPPED_ADDRESS = 0x0020, + PRIORITY = 0x0024, + USE_CANDIDATE = 0x0025, + SOFTWARE = 0x8022, + ALTERNATE_SERVER = 0x8023, + FINGERPRINT = 0x8028, + ICE_CONTROLLED = 0x8029, + ICE_CONTROLLING = 0x802A + }; + + // Authentication result. + enum class Authentication + { + OK = 0, + UNAUTHORIZED = 1, + BAD_REQUEST = 2 + }; + + public: + static bool IsStun(const uint8_t* data, size_t len) + { + // clang-format off + return ( + // STUN headers are 20 bytes. + (len >= 20) && + // DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes + (data[0] < 3) && + // Magic cookie must match. + (data[4] == StunPacket::magicCookie[0]) && (data[5] == StunPacket::magicCookie[1]) && + (data[6] == StunPacket::magicCookie[2]) && (data[7] == StunPacket::magicCookie[3]) + ); + // clang-format on + } + static StunPacket* Parse(const uint8_t* data, size_t len); + + private: + static const uint8_t magicCookie[]; + + public: + StunPacket( + Class klass, Method method, const uint8_t* transactionId, const uint8_t* data, size_t size); + ~StunPacket(); + + void Dump() const; + Class GetClass() const + { + return this->klass; + } + Method GetMethod() const + { + return this->method; + } + const uint8_t* GetData() const + { + return this->data; + } + size_t GetSize() const + { + return this->size; + } + void SetUsername(const char* username, size_t len) + { + this->username.assign(username, len); + } + void SetPriority(uint32_t priority) + { + this->priority = priority; + } + void SetIceControlling(uint64_t iceControlling) + { + this->iceControlling = iceControlling; + } + void SetIceControlled(uint64_t iceControlled) + { + this->iceControlled = iceControlled; + } + void SetUseCandidate() + { + this->hasUseCandidate = true; + } + void SetXorMappedAddress(const struct sockaddr* xorMappedAddress) + { + this->xorMappedAddress = xorMappedAddress; + } + void SetErrorCode(uint16_t errorCode) + { + this->errorCode = errorCode; + } + void SetMessageIntegrity(const uint8_t* messageIntegrity) + { + this->messageIntegrity = messageIntegrity; + } + void SetFingerprint() + { + this->hasFingerprint = true; + } + const std::string& GetUsername() const + { + return this->username; + } + uint32_t GetPriority() const + { + return this->priority; + } + uint64_t GetIceControlling() const + { + return this->iceControlling; + } + uint64_t GetIceControlled() const + { + return this->iceControlled; + } + bool HasUseCandidate() const + { + return this->hasUseCandidate; + } + uint16_t GetErrorCode() const + { + return this->errorCode; + } + bool HasMessageIntegrity() const + { + return (this->messageIntegrity ? true : false); + } + bool HasFingerprint() const + { + return this->hasFingerprint; + } + Authentication CheckAuthentication( + const std::string& localUsername, const std::string& localPassword); + StunPacket* CreateSuccessResponse(); + StunPacket* CreateErrorResponse(uint16_t errorCode); + void Authenticate(const std::string& password); + void Serialize(uint8_t* buffer); + + private: + // Passed by argument. + Class klass; // 2 bytes. + Method method; // 2 bytes. + const uint8_t* transactionId{ nullptr }; // 12 bytes. + uint8_t* data{ nullptr }; // Pointer to binary data. + size_t size{ 0u }; // The full message size (including header). + // STUN attributes. + std::string username; // Less than 513 bytes. + uint32_t priority{ 0u }; // 4 bytes unsigned integer. + uint64_t iceControlling{ 0u }; // 8 bytes unsigned integer. + uint64_t iceControlled{ 0u }; // 8 bytes unsigned integer. + bool hasUseCandidate{ false }; // 0 bytes. + const uint8_t* messageIntegrity{ nullptr }; // 20 bytes. + bool hasFingerprint{ false }; // 4 bytes. + const struct sockaddr* xorMappedAddress{ nullptr }; // 8 or 20 bytes. + uint16_t errorCode{ 0u }; // 4 bytes (no reason phrase). + std::string password; + }; +} // namespace RTC + +#endif diff --git a/webrtc/utils.h b/webrtc/Utils.hpp similarity index 100% rename from webrtc/utils.h rename to webrtc/Utils.hpp diff --git a/webrtc/webrtc_transport.cc b/webrtc/WebRtcTransport.cpp similarity index 99% rename from webrtc/webrtc_transport.cc rename to webrtc/WebRtcTransport.cpp index f064b6db..94c66837 100644 --- a/webrtc/webrtc_transport.cc +++ b/webrtc/WebRtcTransport.cpp @@ -1,4 +1,4 @@ -#include "webrtc_transport.h" +#include "WebRtcTransport.h" #include #include "Rtcp/Rtcp.h" diff --git a/webrtc/webrtc_transport.h b/webrtc/WebRtcTransport.h similarity index 97% rename from webrtc/webrtc_transport.h rename to webrtc/WebRtcTransport.h index b9fcf347..80e0f639 100644 --- a/webrtc/webrtc_transport.h +++ b/webrtc/WebRtcTransport.h @@ -3,10 +3,10 @@ #include #include -#include "rtc_dtls_transport.h" -#include "ice_server.h" -#include "srtp_session.h" -#include "stun_packet.h" +#include "DtlsTransport.hpp" +#include "IceServer.hpp" +#include "SrtpSession.hpp" +#include "StunPacket.hpp" class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: diff --git a/webrtc/logger.h b/webrtc/logger.h index 3da3984f..6b345e98 100644 --- a/webrtc/logger.h +++ b/webrtc/logger.h @@ -2,9 +2,6 @@ #include #include -#define ELOG_DEBUG(fmt, ...) printf(fmt "\n", ##__VA_ARGS__) -#define ELOG_WARN(fmt, ...) printf(fmt "\n", ##__VA_ARGS__) - #define MS_TRACE() #define MS_ERROR(fmt, ...) printf("error:" fmt "\n", ##__VA_ARGS__) #define MS_THROW_ERROR(fmt, ...) do{ printf("throw:" fmt "\n", ##__VA_ARGS__); throw std::runtime_error("error"); } while(false); diff --git a/webrtc/stun_packet.cc b/webrtc/stun_packet.cc deleted file mode 100644 index 1977f7f1..00000000 --- a/webrtc/stun_packet.cc +++ /dev/null @@ -1,705 +0,0 @@ -#define MS_CLASS "RTC::StunPacket" -// #define MS_LOG_DEV - -#include "stun_packet.h" - -#include // std::snprintf() -#include // std::memcmp(), std::memcpy() - -namespace RTC { - -static const uint32_t crc32Table[] = -{ - 0x00000000, 0x77073096, 0xee0e612c, 0x990951ba, 0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3, - 0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988, 0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91, - 0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de, 0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7, - 0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec, 0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5, - 0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172, 0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b, - 0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940, 0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59, - 0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116, 0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f, - 0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924, 0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d, - 0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, 0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433, - 0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818, 0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01, - 0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e, 0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457, - 0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c, 0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65, - 0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2, 0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb, - 0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0, 0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9, - 0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086, 0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f, - 0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4, 0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad, - 0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a, 0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683, - 0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8, 0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1, - 0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe, 0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7, - 0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc, 0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5, - 0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252, 0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b, - 0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60, 0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79, - 0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236, 0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f, - 0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04, 0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d, - 0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a, 0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713, - 0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38, 0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21, - 0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e, 0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777, - 0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c, 0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45, - 0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2, 0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db, - 0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0, 0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9, - 0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6, 0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf, - 0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94, 0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d -}; - -inline uint32_t GetCRC32(const uint8_t *data, size_t size) { - uint32_t crc{0xFFFFFFFF}; - const uint8_t *p = data; - - while (size--) { - crc = crc32Table[(crc ^ *p++) & 0xFF] ^ (crc >> 8); - } - - return crc ^ ~0U; -} - -static std::string openssl_HMACsha1(const void *key, size_t key_len, const void *data, size_t data_len){ - std::string str; - str.resize(20); - unsigned int out_len; -#if defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) - //openssl 1.1.0新增api，老版本api作废 - HMAC_CTX *ctx = HMAC_CTX_new(); - HMAC_CTX_reset(ctx); - HMAC_Init_ex(ctx, key, (int)key_len, EVP_sha1(), NULL); - HMAC_Update(ctx, (unsigned char*)data, data_len); - HMAC_Final(ctx, (unsigned char *)str.data(), &out_len); - HMAC_CTX_reset(ctx); - HMAC_CTX_free(ctx); -#else - HMAC_CTX ctx; - HMAC_CTX_init(&ctx); - HMAC_Init_ex(&ctx, key, key_len, EVP_sha1(), NULL); - HMAC_Update(&ctx, (unsigned char*)data, data_len); - HMAC_Final(&ctx, (unsigned char *)str.data(), &out_len); - HMAC_CTX_cleanup(&ctx); -#endif //defined(OPENSSL_VERSION_NUMBER) && (OPENSSL_VERSION_NUMBER > 0x10100000L) - return str; -} - -/* Class variables. */ - -const uint8_t StunPacket::kMagicCookie[] = {0x21, 0x12, 0xA4, 0x42}; - -/* Class methods. */ - -StunPacket* StunPacket::Parse(const uint8_t* data, size_t len) { - if (!StunPacket::IsStun(data, len)) return nullptr; - - /* - The message type field is decomposed further into the following - structure: - - 0 1 - 2 3 4 5 6 7 8 9 0 1 2 3 4 5 - +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ - |M |M |M|M|M|C|M|M|M|C|M|M|M|M| - |11|10|9|8|7|1|6|5|4|0|3|2|1|0| - +--+--+-+-+-+-+-+-+-+-+-+-+-+-+ - - Figure 3: Format of STUN Message Type Field - - Here the bits in the message type field are shown as most significant - (M11) through least significant (M0). M11 through M0 represent a 12- - bit encoding of the method. C1 and C0 represent a 2-bit encoding of - the class. - */ - - // Get type field. - uint16_t msgType = Utils::Byte::Get2Bytes(data, 0); - - // Get length field. - uint16_t msgLength = Utils::Byte::Get2Bytes(data, 2); - - // length field must be total size minus header's 20 bytes, and must be multiple of 4 Bytes. - if ((static_cast(msgLength) != len - 20) || ((msgLength & 0x03) != 0)) { - ELOG_DEBUG( - "length field + 20 does not match total size (or it is not multiple of 4 bytes), " - "packet discarded"); - - return nullptr; - } - - // Get STUN method. - uint16_t msgMethod = (msgType & 0x000f) | ((msgType & 0x00e0) >> 1) | ((msgType & 0x3E00) >> 2); - - // Get STUN class. - uint16_t msgClass = ((data[0] & 0x01) << 1) | ((data[1] & 0x10) >> 4); - - // Create a new StunPacket (data + 8 points to the received TransactionID field). - auto packet = new StunPacket(static_cast(msgClass), static_cast(msgMethod), - data + 8, data, len); - - /* - STUN Attributes - - After the STUN header are zero or more attributes. Each attribute - MUST be TLV encoded, with a 16-bit type, 16-bit length, and value. - Each STUN attribute MUST end on a 32-bit boundary. As mentioned - above, all fields in an attribute are transmitted most significant - bit first. - - 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | Type | Length | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | Value (variable) .... - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ - - // Start looking for attributes after STUN header (Byte #20). - size_t pos{20}; - // Flags (positions) for special MESSAGE-INTEGRITY and FINGERPRINT attributes. - bool hasMessageIntegrity{false}; - bool hasFingerprint{false}; - size_t fingerprintAttrPos; // Will point to the beginning of the attribute. - uint32_t fingerprint; // Holds the value of the FINGERPRINT attribute. - - // Ensure there are at least 4 remaining bytes (attribute with 0 length). - while (pos + 4 <= len) { - // Get the attribute type. - auto attrType = static_cast(Utils::Byte::Get2Bytes(data, pos)); - - // Get the attribute length. - uint16_t attrLength = Utils::Byte::Get2Bytes(data, pos + 2); - - // Ensure the attribute length is not greater than the remaining size. - if ((pos + 4 + attrLength) > len) { - ELOG_DEBUG("the attribute length exceeds the remaining size, packet discarded"); - - delete packet; - return nullptr; - } - - // FINGERPRINT must be the last attribute. - if (hasFingerprint) { - ELOG_DEBUG("attribute after FINGERPRINT is not allowed, packet discarded"); - - delete packet; - return nullptr; - } - - // After a MESSAGE-INTEGRITY attribute just FINGERPRINT is allowed. - if (hasMessageIntegrity && attrType != Attribute::FINGERPRINT) { - ELOG_DEBUG( - "attribute after MESSAGE-INTEGRITY other than FINGERPRINT is not allowed, " - "packet discarded"); - - delete packet; - return nullptr; - } - - const uint8_t* attrValuePos = data + pos + 4; - - switch (attrType) { - case Attribute::USERNAME: { - packet->SetUsername(reinterpret_cast(attrValuePos), - static_cast(attrLength)); - - break; - } - - case Attribute::PRIORITY: { - // Ensure attribute length is 4 bytes. - if (attrLength != 4) { - ELOG_DEBUG("attribute PRIORITY must be 4 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - packet->SetPriority(Utils::Byte::Get4Bytes(attrValuePos, 0)); - - break; - } - - case Attribute::ICE_CONTROLLING: { - // Ensure attribute length is 8 bytes. - if (attrLength != 8) { - ELOG_DEBUG("attribute ICE-CONTROLLING must be 8 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - packet->SetIceControlling(Utils::Byte::Get8Bytes(attrValuePos, 0)); - - break; - } - - case Attribute::ICE_CONTROLLED: { - // Ensure attribute length is 8 bytes. - if (attrLength != 8) { - ELOG_DEBUG("attribute ICE-CONTROLLED must be 8 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - packet->SetIceControlled(Utils::Byte::Get8Bytes(attrValuePos, 0)); - - break; - } - - case Attribute::USE_CANDIDATE: { - // Ensure attribute length is 0 bytes. - if (attrLength != 0) { - ELOG_DEBUG("attribute USE-CANDIDATE must be 0 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - packet->SetUseCandidate(); - - break; - } - - case Attribute::MESSAGE_INTEGRITY: { - // Ensure attribute length is 20 bytes. - if (attrLength != 20) { - ELOG_DEBUG("attribute MESSAGE-INTEGRITY must be 20 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - hasMessageIntegrity = true; - packet->SetMessageIntegrity(attrValuePos); - - break; - } - - case Attribute::FINGERPRINT: { - // Ensure attribute length is 4 bytes. - if (attrLength != 4) { - ELOG_DEBUG("attribute FINGERPRINT must be 4 bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - hasFingerprint = true; - fingerprintAttrPos = pos; - fingerprint = Utils::Byte::Get4Bytes(attrValuePos, 0); - packet->SetFingerprint(); - - break; - } - - case Attribute::ERROR_CODE: { - // Ensure attribute length >= 4bytes. - if (attrLength < 4) { - ELOG_DEBUG("attribute ERROR-CODE must be >= 4bytes length, packet discarded"); - - delete packet; - return nullptr; - } - - uint8_t errorClass = Utils::Byte::Get1Byte(attrValuePos, 2); - uint8_t errorNumber = Utils::Byte::Get1Byte(attrValuePos, 3); - auto errorCode = static_cast(errorClass * 100 + errorNumber); - - packet->SetErrorCode(errorCode); - - break; - } - - default:; - } - - // Set next attribute position. - pos = static_cast(Utils::Byte::PadTo4Bytes(static_cast(pos + 4 + attrLength))); - } - - // Ensure current position matches the total length. - if (pos != len) { - ELOG_DEBUG("computed packet size does not match total size, packet discarded"); - - delete packet; - return nullptr; - } - - // If it has FINGERPRINT attribute then verify it. - if (hasFingerprint) { - // Compute the CRC32 of the received packet up to (but excluding) the - // FINGERPRINT attribute and XOR it with 0x5354554e. - uint32_t computedFingerprint = GetCRC32(data, fingerprintAttrPos) ^ 0x5354554e; - - // Compare with the FINGERPRINT value in the packet. - if (fingerprint != computedFingerprint) { - ELOG_DEBUG( - "computed FINGERPRINT value does not match the value in the packet, " - "packet discarded"); - - delete packet; - return nullptr; - } - } - - return packet; -} - -/* Instance methods. */ - -StunPacket::StunPacket(Class klass, Method method, const uint8_t* transactionId, - const uint8_t* data, size_t size) - : klass(klass), - method(method), - transactionId(transactionId), - data(const_cast(data)), - size(size) { - // MS_TRACE(); -} - -StunPacket::~StunPacket() { - // MS_TRACE(); -} - -StunPacket::Authentication StunPacket::CheckAuthentication(const std::string& localUsername, - const std::string& localPassword) { - // MS_TRACE(); - - switch (this->klass) { - case Class::REQUEST: - case Class::INDICATION: { - // Both USERNAME and MESSAGE-INTEGRITY must be present. - if (this->messageIntegrity == nullptr || this->username.empty()) - return Authentication::BAD_REQUEST; - - // Check that USERNAME attribute begins with our local username plus ":". - size_t localUsernameLen = localUsername.length(); - - if (this->username.length() <= localUsernameLen || - this->username.at(localUsernameLen) != ':' || - (this->username.compare(0, localUsernameLen, localUsername) != 0)) { - return Authentication::UNAUTHORIZED; - } - - break; - } - // This method cannot check authentication in received responses (as we - // are ICE-Lite and don't generate requests). - case Class::SUCCESS_RESPONSE: - case Class::ERROR_RESPONSE: { - // MS_ERROR("cannot check authentication for a STUN response"); - - return Authentication::BAD_REQUEST; - } - } - - // If there is FINGERPRINT it must be discarded for MESSAGE-INTEGRITY calculation, - // so the header length field must be modified (and later restored). - if (this->hasFingerprint) - // Set the header length field: full size - header length (20) - FINGERPRINT length (8). - Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20 - 8)); - - // Calculate the HMAC-SHA1 of the message according to MESSAGE-INTEGRITY rules. - auto computedMessageIntegrity = openssl_HMACsha1( - localPassword.data(),localPassword.size(), this->data, (this->messageIntegrity - 4) - this->data); - - Authentication result; - - // Compare the computed HMAC-SHA1 with the MESSAGE-INTEGRITY in the packet. - if (std::memcmp(this->messageIntegrity, computedMessageIntegrity.data(), computedMessageIntegrity.size()) == 0) - result = Authentication::OK; - else - result = Authentication::UNAUTHORIZED; - - // Restore the header length field. - if (this->hasFingerprint) - Utils::Byte::Set2Bytes(this->data, 2, static_cast(this->size - 20)); - - return result; -} - -StunPacket* StunPacket::CreateSuccessResponse() { - // MS_TRACE(); - - // MS_ASSERT( - // this->klass == Class::REQUEST, - // "attempt to create a success response for a non Request STUN packet"); - - return new StunPacket(Class::SUCCESS_RESPONSE, this->method, this->transactionId, nullptr, 0); -} - -StunPacket* StunPacket::CreateErrorResponse(uint16_t errorCode) { - // MS_TRACE(); - - // MS_ASSERT( - // this->klass == Class::REQUEST, - // "attempt to create an error response for a non Request STUN packet"); - - auto response = - new StunPacket(Class::ERROR_RESPONSE, this->method, this->transactionId, nullptr, 0); - - response->SetErrorCode(errorCode); - - return response; -} - -void StunPacket::Authenticate(const std::string& password) { - // Just for Request, Indication and SuccessResponse messages. - if (this->klass == Class::ERROR_RESPONSE) { - // MS_ERROR("cannot set password for ErrorResponse messages"); - - return; - } - - this->password = password; -} - -void StunPacket::Serialize(uint8_t* buffer) { - // MS_TRACE(); - - // Some useful variables. - uint16_t usernamePaddedLen{0}; - uint16_t xorMappedAddressPaddedLen{0}; - bool addXorMappedAddress = - ((this->xorMappedAddress != nullptr) && this->method == StunPacket::Method::BINDING && - this->klass == Class::SUCCESS_RESPONSE); - bool addErrorCode = ((this->errorCode != 0u) && this->klass == Class::ERROR_RESPONSE); - bool addMessageIntegrity = (this->klass != Class::ERROR_RESPONSE && !this->password.empty()); - bool addFingerprint{true}; // Do always. - - // Update data pointer. - this->data = buffer; - - // First calculate the total required size for the entire packet. - this->size = 20; // Header. - - if (!this->username.empty()) { - usernamePaddedLen = Utils::Byte::PadTo4Bytes(static_cast(this->username.length())); - this->size += 4 + usernamePaddedLen; - } - - if (this->priority != 0u) this->size += 4 + 4; - - if (this->iceControlling != 0u) this->size += 4 + 8; - - if (this->iceControlled != 0u) this->size += 4 + 8; - - if (this->hasUseCandidate) this->size += 4; - - if (addXorMappedAddress) { - switch (this->xorMappedAddress->sa_family) { - case AF_INET: { - xorMappedAddressPaddedLen = 8; - this->size += 4 + 8; - - break; - } - - case AF_INET6: { - xorMappedAddressPaddedLen = 20; - this->size += 4 + 20; - - break; - } - - default: { - // MS_ERROR("invalid inet family in XOR-MAPPED-ADDRESS attribute"); - - addXorMappedAddress = false; - } - } - } - - if (addErrorCode) this->size += 4 + 4; - - if (addMessageIntegrity) this->size += 4 + 20; - - if (addFingerprint) this->size += 4 + 4; - - // Merge class and method fields into type. - uint16_t typeField = (static_cast(this->method) & 0x0f80) << 2; - - typeField |= (static_cast(this->method) & 0x0070) << 1; - typeField |= (static_cast(this->method) & 0x000f); - typeField |= (static_cast(this->klass) & 0x02) << 7; - typeField |= (static_cast(this->klass) & 0x01) << 4; - - // Set type field. - Utils::Byte::Set2Bytes(buffer, 0, typeField); - // Set length field. - Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size) - 20); - // Set magic cookie. - std::memcpy(buffer + 4, StunPacket::kMagicCookie, 4); - // Set TransactionId field. - std::memcpy(buffer + 8, this->transactionId, 12); - // Update the transaction ID pointer. - this->transactionId = buffer + 8; - // Add atributes. - size_t pos{20}; - - // Add USERNAME. - if (usernamePaddedLen != 0u) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USERNAME)); - Utils::Byte::Set2Bytes(buffer, pos + 2, static_cast(this->username.length())); - std::memcpy(buffer + pos + 4, this->username.c_str(), this->username.length()); - pos += 4 + usernamePaddedLen; - } - - // Add PRIORITY. - if (this->priority != 0u) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::PRIORITY)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 4); - Utils::Byte::Set4Bytes(buffer, pos + 4, this->priority); - pos += 4 + 4; - } - - // Add ICE-CONTROLLING. - if (this->iceControlling != 0u) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLING)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 8); - Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlling); - pos += 4 + 8; - } - - // Add ICE-CONTROLLED. - if (this->iceControlled != 0u) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ICE_CONTROLLED)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 8); - Utils::Byte::Set8Bytes(buffer, pos + 4, this->iceControlled); - pos += 4 + 8; - } - - // Add USE-CANDIDATE. - if (this->hasUseCandidate) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::USE_CANDIDATE)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 0); - pos += 4; - } - - // Add XOR-MAPPED-ADDRESS - if (addXorMappedAddress) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::XOR_MAPPED_ADDRESS)); - Utils::Byte::Set2Bytes(buffer, pos + 2, xorMappedAddressPaddedLen); - - uint8_t* attrValue = buffer + pos + 4; - - switch (this->xorMappedAddress->sa_family) { - case AF_INET: { - // Set first byte to 0. - attrValue[0] = 0; - // Set inet family. - attrValue[1] = 0x01; - // Set port and XOR it. - std::memcpy(attrValue + 2, - &(reinterpret_cast(this->xorMappedAddress))->sin_port, 2); - attrValue[2] ^= StunPacket::kMagicCookie[0]; - attrValue[3] ^= StunPacket::kMagicCookie[1]; - // Set address and XOR it. - std::memcpy( - attrValue + 4, - &(reinterpret_cast(this->xorMappedAddress))->sin_addr.s_addr, 4); - attrValue[4] ^= StunPacket::kMagicCookie[0]; - attrValue[5] ^= StunPacket::kMagicCookie[1]; - attrValue[6] ^= StunPacket::kMagicCookie[2]; - attrValue[7] ^= StunPacket::kMagicCookie[3]; - - pos += 4 + 8; - - break; - } - - case AF_INET6: { - // Set first byte to 0. - attrValue[0] = 0; - // Set inet family. - attrValue[1] = 0x02; - // Set port and XOR it. - std::memcpy(attrValue + 2, - &(reinterpret_cast(this->xorMappedAddress))->sin6_port, 2); - attrValue[2] ^= StunPacket::kMagicCookie[0]; - attrValue[3] ^= StunPacket::kMagicCookie[1]; - // Set address and XOR it. - std::memcpy( - attrValue + 4, - &(reinterpret_cast(this->xorMappedAddress))->sin6_addr.s6_addr, - 16); - attrValue[4] ^= StunPacket::kMagicCookie[0]; - attrValue[5] ^= StunPacket::kMagicCookie[1]; - attrValue[6] ^= StunPacket::kMagicCookie[2]; - attrValue[7] ^= StunPacket::kMagicCookie[3]; - attrValue[8] ^= this->transactionId[0]; - attrValue[9] ^= this->transactionId[1]; - attrValue[10] ^= this->transactionId[2]; - attrValue[11] ^= this->transactionId[3]; - attrValue[12] ^= this->transactionId[4]; - attrValue[13] ^= this->transactionId[5]; - attrValue[14] ^= this->transactionId[6]; - attrValue[15] ^= this->transactionId[7]; - attrValue[16] ^= this->transactionId[8]; - attrValue[17] ^= this->transactionId[9]; - attrValue[18] ^= this->transactionId[10]; - attrValue[19] ^= this->transactionId[11]; - - pos += 4 + 20; - - break; - } - } - } - - // Add ERROR-CODE. - if (addErrorCode) { - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::ERROR_CODE)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 4); - - auto codeClass = static_cast(this->errorCode / 100); - uint8_t codeNumber = static_cast(this->errorCode) - (codeClass * 100); - - Utils::Byte::Set2Bytes(buffer, pos + 4, 0); - Utils::Byte::Set1Byte(buffer, pos + 6, codeClass); - Utils::Byte::Set1Byte(buffer, pos + 7, codeNumber); - pos += 4 + 4; - } - - // Add MESSAGE-INTEGRITY. - if (addMessageIntegrity) { - // Ignore FINGERPRINT. - if (addFingerprint) - Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20 - 8)); - - // Calculate the HMAC-SHA1 of the packet according to MESSAGE-INTEGRITY rules. - auto computedMessageIntegrity = openssl_HMACsha1(this->password.data(), this->password.size(), buffer, pos); - - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::MESSAGE_INTEGRITY)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 20); - std::memcpy(buffer + pos + 4, computedMessageIntegrity.data(), computedMessageIntegrity.size()); - - // Update the pointer. - this->messageIntegrity = buffer + pos + 4; - pos += 4 + 20; - - // Restore length field. - if (addFingerprint) Utils::Byte::Set2Bytes(buffer, 2, static_cast(this->size - 20)); - } else { - // Unset the pointer (if it was set). - this->messageIntegrity = nullptr; - } - - // Add FINGERPRINT. - if (addFingerprint) { - // Compute the CRC32 of the packet up to (but excluding) the FINGERPRINT - // attribute and XOR it with 0x5354554e. - uint32_t computedFingerprint = GetCRC32(buffer, pos) ^ 0x5354554e; - - Utils::Byte::Set2Bytes(buffer, pos, static_cast(Attribute::FINGERPRINT)); - Utils::Byte::Set2Bytes(buffer, pos + 2, 4); - Utils::Byte::Set4Bytes(buffer, pos + 4, computedFingerprint); - pos += 4 + 4; - - // Set flag. - this->hasFingerprint = true; - } else { - this->hasFingerprint = false; - } - - // MS_ASSERT(pos == this->size, "pos != this->size"); -} -} // namespace RTC diff --git a/webrtc/stun_packet.h b/webrtc/stun_packet.h deleted file mode 100644 index 861d11cd..00000000 --- a/webrtc/stun_packet.h +++ /dev/null @@ -1,179 +0,0 @@ -#ifndef MS_RTC_STUN_PACKET_HPP -#define MS_RTC_STUN_PACKET_HPP - - -#include "logger.h" -#include "utils.h" -#include - -namespace RTC { -class StunPacket { -public: - // STUN message class. - enum class Class : uint16_t { - REQUEST = 0, - INDICATION = 1, - SUCCESS_RESPONSE = 2, - ERROR_RESPONSE = 3 - }; - - // STUN message method. - enum class Method : uint16_t { BINDING = 1 }; - - // Attribute type. - enum class Attribute : uint16_t { - MAPPED_ADDRESS = 0x0001, - USERNAME = 0x0006, - MESSAGE_INTEGRITY = 0x0008, - ERROR_CODE = 0x0009, - UNKNOWN_ATTRIBUTES = 0x000A, - REALM = 0x0014, - NONCE = 0x0015, - XOR_MAPPED_ADDRESS = 0x0020, - PRIORITY = 0x0024, - USE_CANDIDATE = 0x0025, - SOFTWARE = 0x8022, - ALTERNATE_SERVER = 0x8023, - FINGERPRINT = 0x8028, - ICE_CONTROLLED = 0x8029, - ICE_CONTROLLING = 0x802A - }; - - // Authentication result. - enum class Authentication { OK = 0, UNAUTHORIZED = 1, BAD_REQUEST = 2 }; - -public: - static bool IsStun(const uint8_t *data, size_t len); - static StunPacket *Parse(const uint8_t *data, size_t len); - -private: - static const uint8_t kMagicCookie[]; - -public: - StunPacket(Class klass, Method method, const uint8_t *transactionId, const uint8_t *data, - size_t size); - ~StunPacket(); - - void Dump() const; - Class GetClass() const; - Method GetMethod() const; - const uint8_t *GetData() const; - size_t GetSize() const; - void SetUsername(const char *username, size_t len); - void SetPriority(uint32_t priority); - void SetIceControlling(uint64_t iceControlling); - void SetIceControlled(uint64_t iceControlled); - void SetUseCandidate(); - void SetXorMappedAddress(const struct sockaddr *xorMappedAddress); - void SetErrorCode(uint16_t errorCode); - void SetMessageIntegrity(const uint8_t *messageIntegrity); - void SetFingerprint(); - const std::string &GetUsername() const; - uint32_t GetPriority() const; - uint64_t GetIceControlling() const; - uint64_t GetIceControlled() const; - bool HasUseCandidate() const; - uint16_t GetErrorCode() const; - bool HasMessageIntegrity() const; - bool HasFingerprint() const; - Authentication CheckAuthentication(const std::string &localUsername, - const std::string &localPassword); - StunPacket *CreateSuccessResponse(); - StunPacket *CreateErrorResponse(uint16_t errorCode); - void Authenticate(const std::string &password); - void Serialize(uint8_t *buffer); - -private: - // Passed by argument. - Class klass; // 2 bytes. - Method method; // 2 bytes. - const uint8_t *transactionId{nullptr};// 12 bytes. - uint8_t *data{nullptr}; // Pointer to binary data. - size_t size{0}; // The full message size (including header). - // STUN attributes. - std::string username; // Less than 513 bytes. - uint32_t priority{0}; // 4 bytes unsigned integer. - uint64_t iceControlling{0}; // 8 bytes unsigned integer. - uint64_t iceControlled{0}; // 8 bytes unsigned integer. - bool hasUseCandidate{false}; // 0 bytes. - const uint8_t *messageIntegrity{nullptr}; // 20 bytes. - bool hasFingerprint{false}; // 4 bytes. - const struct sockaddr *xorMappedAddress{nullptr};// 8 or 20 bytes. - uint16_t errorCode{0}; // 4 bytes (no reason phrase). - std::string password; -}; - -/* Inline class methods. */ - -inline bool StunPacket::IsStun(const uint8_t *data, size_t len) { - // clang-format off - return ( - // STUN headers are 20 bytes. - (len >= 20) && - // DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes - (data[0] < 3) && - // Magic cookie must match. - (data[4] == StunPacket::kMagicCookie[0]) && (data[5] == StunPacket::kMagicCookie[1]) && - (data[6] == StunPacket::kMagicCookie[2]) && (data[7] == StunPacket::kMagicCookie[3]) - ); - // clang-format on -} - -/* Inline instance methods. */ - -inline StunPacket::Class StunPacket::GetClass() const { return this->klass; } - -inline StunPacket::Method StunPacket::GetMethod() const { return this->method; } - -inline const uint8_t *StunPacket::GetData() const { return this->data; } - -inline size_t StunPacket::GetSize() const { return this->size; } - -inline void StunPacket::SetUsername(const char *username, size_t len) { - this->username.assign(username, len); -} - -inline void StunPacket::SetPriority(const uint32_t priority) { this->priority = priority; } - -inline void StunPacket::SetIceControlling(const uint64_t iceControlling) { - this->iceControlling = iceControlling; -} - -inline void StunPacket::SetIceControlled(const uint64_t iceControlled) { - this->iceControlled = iceControlled; -} - -inline void StunPacket::SetUseCandidate() { this->hasUseCandidate = true; } - -inline void StunPacket::SetXorMappedAddress(const struct sockaddr *xorMappedAddress) { - this->xorMappedAddress = xorMappedAddress; -} - -inline void StunPacket::SetErrorCode(uint16_t errorCode) { this->errorCode = errorCode; } - -inline void StunPacket::SetMessageIntegrity(const uint8_t *messageIntegrity) { - this->messageIntegrity = messageIntegrity; -} - -inline void StunPacket::SetFingerprint() { this->hasFingerprint = true; } - -inline const std::string &StunPacket::GetUsername() const { return this->username; } - -inline uint32_t StunPacket::GetPriority() const { return this->priority; } - -inline uint64_t StunPacket::GetIceControlling() const { return this->iceControlling; } - -inline uint64_t StunPacket::GetIceControlled() const { return this->iceControlled; } - -inline bool StunPacket::HasUseCandidate() const { return this->hasUseCandidate; } - -inline uint16_t StunPacket::GetErrorCode() const { return this->errorCode; } - -inline bool StunPacket::HasMessageIntegrity() const { - return (this->messageIntegrity ? true : false); -} - -inline bool StunPacket::HasFingerprint() const { return this->hasFingerprint; } -}// namespace RTC - -#endif From 0d4cc2fc65e70df408494918a8f0d1f32a262b91 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 09:38:13 +0800 Subject: [PATCH 005/218] =?UTF-8?q?srtp=E8=87=AA=E5=8A=A8=E5=88=9D?= =?UTF-8?q?=E5=A7=8B=E5=8C=96=EF=BC=8C=E6=94=AF=E6=8C=81=E5=A4=9A=E7=BA=BF?= =?UTF-8?q?=E7=A8=8B?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/SrtpSession.cpp | 93 ++++++++++++++++++++------------------ webrtc/SrtpSession.hpp | 29 ++++++------ webrtc/WebRtcTransport.cpp | 7 +-- 3 files changed, 66 insertions(+), 63 deletions(-) diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index 13eeebd4..580c190b 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -4,15 +4,14 @@ #include "SrtpSession.hpp" #include // std::memset(), std::memcpy() #include "logger.h" +#include "Util/util.h" +using namespace toolkit; namespace RTC { /* Static. */ - static constexpr size_t EncryptBufferSize{ 65536 }; - static uint8_t EncryptBuffer[EncryptBufferSize]; - - std::vector DepLibSRTP::errors = { + static std::vector errors = { // From 0 (srtp_err_status_ok) to 24 (srtp_err_status_pfkey_err). "success (srtp_err_status_ok)", "unspecified failure (srtp_err_status_fail)", @@ -43,65 +42,69 @@ namespace RTC /* Static methods. */ - void DepLibSRTP::ClassInit() { + const char *DepLibSRTP::GetErrorString(srtp_err_status_t code) { + // This throws out_of_range if the given index is not in the vector. + return errors.at(code); + } + + bool DepLibSRTP::IsError(srtp_err_status_t code) { + return (code != srtp_err_status_ok); + } + + INSTANCE_IMP(DepLibSRTP); + + DepLibSRTP::DepLibSRTP(){ MS_TRACE(); MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string()); srtp_err_status_t err = srtp_init(); - if (DepLibSRTP::IsError(err)) + if (DepLibSRTP::IsError(err)) { MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err)); + } + + // Set libsrtp event handler. + err = srtp_install_event_handler([](srtp_event_data_t *data){ + MS_TRACE(); + switch (data->event) + { + case event_ssrc_collision: + MS_WARN_TAG(srtp, "SSRC collision occurred"); + break; + + case event_key_soft_limit: + MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon"); + break; + + case event_key_hard_limit: + MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired"); + break; + + case event_packet_index_limit: + MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)"); + break; + } + }); + + if (DepLibSRTP::IsError(err)) + { + MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err)); + } } - void DepLibSRTP::ClassDestroy() { + DepLibSRTP::~DepLibSRTP(){ MS_TRACE(); - srtp_shutdown(); } - /* Class methods. */ - - void SrtpSession::ClassInit() - { - // Set libsrtp event handler. - srtp_err_status_t err = - srtp_install_event_handler(static_cast(OnSrtpEvent)); - - if (DepLibSRTP::IsError(err)) - { - MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err)); - } - } - - void SrtpSession::OnSrtpEvent(srtp_event_data_t* data) - { - MS_TRACE(); - - switch (data->event) - { - case event_ssrc_collision: - MS_WARN_TAG(srtp, "SSRC collision occurred"); - break; - - case event_key_soft_limit: - MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon"); - break; - - case event_key_hard_limit: - MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired"); - break; - - case event_packet_index_limit: - MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)"); - break; - } - } + ///////////////////////////////////////////////////////////////////////////////////// /* Instance methods. */ SrtpSession::SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen) { + _env = DepLibSRTP::Instance().shared_from_this(); MS_TRACE(); srtp_policy_t policy; // NOLINT(cppcoreguidelines-pro-type-member-init) diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index 3a4c32f4..fb21ea9f 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -4,21 +4,22 @@ #include "Utils.hpp" #include #include +#include namespace RTC { - class DepLibSRTP { + class DepLibSRTP : public std::enable_shared_from_this + { public: - static void ClassInit(); - static void ClassDestroy(); - static bool IsError(srtp_err_status_t code) { return (code != srtp_err_status_ok); } - static const char *GetErrorString(srtp_err_status_t code) { - // This throws out_of_range if the given index is not in the vector. - return DepLibSRTP::errors.at(code); - } + using Ptr = std::shared_ptr; + ~DepLibSRTP(); + + static bool IsError(srtp_err_status_t code); + static const char *GetErrorString(srtp_err_status_t code); + static DepLibSRTP &Instance(); private: - static std::vector errors; + DepLibSRTP(); }; class SrtpSession @@ -40,12 +41,6 @@ namespace RTC OUTBOUND }; - public: - static void ClassInit(); - - private: - static void OnSrtpEvent(srtp_event_data_t* data); - public: SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen); ~SrtpSession(); @@ -63,6 +58,10 @@ namespace RTC private: // Allocated by this. srtp_t session{ nullptr }; + //rtp包最大1600 + static constexpr size_t EncryptBufferSize{ 1600 }; + uint8_t EncryptBuffer[EncryptBufferSize]; + DepLibSRTP::Ptr _env; }; } // namespace RTC diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 94c66837..f727876c 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -5,15 +5,16 @@ WebRtcTransport::WebRtcTransport() { static onceToken token([](){ RTC::DtlsTransport::ClassInit(); - RTC::DepLibSRTP::ClassInit(); - RTC::SrtpSession::ClassInit(); }); dtls_transport_ = std::make_shared(EventPollerPool::Instance().getFirstPoller(), this); ice_server_ = std::make_shared(this, makeRandStr(4), makeRandStr(24)); } -WebRtcTransport::~WebRtcTransport() {} +WebRtcTransport::~WebRtcTransport() { + dtls_transport_ = nullptr; + ice_server_ = nullptr; +} ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// From d0a1c72fda0abf7226a045a27eab2de5abcce1aa Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 10:03:19 +0800 Subject: [PATCH 006/218] =?UTF-8?q?dtls=E7=8E=AF=E5=A2=83=E8=87=AA?= =?UTF-8?q?=E5=8A=A8=E5=88=9D=E5=A7=8B=E5=8C=96=EF=BC=8C=E6=94=AF=E6=8C=81?= =?UTF-8?q?=E5=A4=9A=E7=BA=BF=E7=A8=8B?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/DtlsTransport.cpp | 118 ++++++++++++++++++------------------- webrtc/DtlsTransport.hpp | 43 +++++++++----- webrtc/WebRtcTransport.cpp | 4 -- 3 files changed, 84 insertions(+), 81 deletions(-) diff --git a/webrtc/DtlsTransport.cpp b/webrtc/DtlsTransport.cpp index 1c001a05..950f9620 100644 --- a/webrtc/DtlsTransport.cpp +++ b/webrtc/DtlsTransport.cpp @@ -10,6 +10,7 @@ #include #include // std::sprintf(), std::fopen() #include // std::memcpy(), std::strcmp() +#include "Util/util.h" #define LOG_OPENSSL_ERROR(desc) \ do \ @@ -53,7 +54,6 @@ namespace RTC // clang-format off static constexpr int DtlsMtu{ 1350 }; - static constexpr int SslReadBufferSize{ 65536 }; // AES-HMAC: http://tools.ietf.org/html/rfc3711 static constexpr size_t SrtpMasterKeyLength{ 16 }; static constexpr size_t SrtpMasterSaltLength{ 14 }; @@ -68,11 +68,6 @@ namespace RTC // clang-format on /* Class variables. */ - - X509* DtlsTransport::certificate{ nullptr }; - EVP_PKEY* DtlsTransport::privateKey{ nullptr }; - SSL_CTX* DtlsTransport::sslCtx{ nullptr }; - uint8_t DtlsTransport::sslReadBuffer[SslReadBufferSize]; // clang-format off std::map DtlsTransport::string2FingerprintAlgorithm = { @@ -96,7 +91,6 @@ namespace RTC { "client", DtlsTransport::Role::CLIENT }, { "server", DtlsTransport::Role::SERVER } }; - std::vector DtlsTransport::localFingerprints; std::vector DtlsTransport::srtpCryptoSuites = { { RTC::SrtpSession::CryptoSuite::AEAD_AES_256_GCM, "SRTP_AEAD_AES_256_GCM" }, @@ -106,13 +100,14 @@ namespace RTC }; // clang-format on + INSTANCE_IMP(DtlsTransport::DtlsEnvironment); + /* Class methods. */ - void DtlsTransport::ClassInit() + DtlsTransport::DtlsEnvironment::DtlsEnvironment() { MS_TRACE(); - // Generate a X509 certificate and private key (unless PEM files are provided). if (true /* Settings::configuration.dtlsCertificateFile.empty() || @@ -132,19 +127,19 @@ namespace RTC GenerateFingerprints(); } - void DtlsTransport::ClassDestroy() + DtlsTransport::DtlsEnvironment::~DtlsEnvironment() { MS_TRACE(); - if (DtlsTransport::privateKey) - EVP_PKEY_free(DtlsTransport::privateKey); - if (DtlsTransport::certificate) - X509_free(DtlsTransport::certificate); - if (DtlsTransport::sslCtx) - SSL_CTX_free(DtlsTransport::sslCtx); + if (privateKey) + EVP_PKEY_free(privateKey); + if (certificate) + X509_free(certificate); + if (sslCtx) + SSL_CTX_free(sslCtx); } - void DtlsTransport::GenerateCertificateAndPrivateKey() + void DtlsTransport::DtlsEnvironment::GenerateCertificateAndPrivateKey() { MS_TRACE(); @@ -177,9 +172,9 @@ namespace RTC } // Create a private key object. - DtlsTransport::privateKey = EVP_PKEY_new(); + privateKey = EVP_PKEY_new(); - if (!DtlsTransport::privateKey) + if (!privateKey) { LOG_OPENSSL_ERROR("EVP_PKEY_new() failed"); @@ -187,7 +182,7 @@ namespace RTC } // NOLINTNEXTLINE(cppcoreguidelines-pro-type-cstyle-cast) - ret = EVP_PKEY_assign_EC_KEY(DtlsTransport::privateKey, ecKey); + ret = EVP_PKEY_assign_EC_KEY(privateKey, ecKey); if (ret == 0) { @@ -200,9 +195,9 @@ namespace RTC ecKey = nullptr; // Create the X509 certificate. - DtlsTransport::certificate = X509_new(); + certificate = X509_new(); - if (!DtlsTransport::certificate) + if (!certificate) { LOG_OPENSSL_ERROR("X509_new() failed"); @@ -210,19 +205,19 @@ namespace RTC } // Set version 3 (note that 0 means version 1). - X509_set_version(DtlsTransport::certificate, 2); + X509_set_version(certificate, 2); // Set serial number (avoid default 0). ASN1_INTEGER_set( - X509_get_serialNumber(DtlsTransport::certificate), + X509_get_serialNumber(certificate), static_cast(rand() % 999999 + 100000)); // Set valid period. - X509_gmtime_adj(X509_get_notBefore(DtlsTransport::certificate), -315360000); // -10 years. - X509_gmtime_adj(X509_get_notAfter(DtlsTransport::certificate), 315360000); // 10 years. + X509_gmtime_adj(X509_get_notBefore(certificate), -315360000); // -10 years. + X509_gmtime_adj(X509_get_notAfter(certificate), 315360000); // 10 years. // Set the public key for the certificate using the key. - ret = X509_set_pubkey(DtlsTransport::certificate, DtlsTransport::privateKey); + ret = X509_set_pubkey(certificate, privateKey); if (ret == 0) { @@ -232,7 +227,7 @@ namespace RTC } // Set certificate fields. - certName = X509_get_subject_name(DtlsTransport::certificate); + certName = X509_get_subject_name(certificate); if (!certName) { @@ -247,7 +242,7 @@ namespace RTC certName, "CN", MBSTRING_ASC, reinterpret_cast(subject.c_str()), -1, -1, 0); // It is self-signed so set the issuer name to be the same as the subject. - ret = X509_set_issuer_name(DtlsTransport::certificate, certName); + ret = X509_set_issuer_name(certificate, certName); if (ret == 0) { @@ -257,7 +252,7 @@ namespace RTC } // Sign the certificate with its own private key. - ret = X509_sign(DtlsTransport::certificate, DtlsTransport::privateKey, EVP_sha1()); + ret = X509_sign(certificate, privateKey, EVP_sha1()); if (ret == 0) { @@ -273,16 +268,16 @@ namespace RTC if (ecKey) EC_KEY_free(ecKey); - if (DtlsTransport::privateKey) - EVP_PKEY_free(DtlsTransport::privateKey); // NOTE: This also frees the EC key. + if (privateKey) + EVP_PKEY_free(privateKey); // NOTE: This also frees the EC key. - if (DtlsTransport::certificate) - X509_free(DtlsTransport::certificate); + if (certificate) + X509_free(certificate); MS_THROW_ERROR("DTLS certificate and private key generation failed"); } - void DtlsTransport::ReadCertificateAndPrivateKeyFromFiles() + void DtlsTransport::DtlsEnvironment::ReadCertificateAndPrivateKeyFromFiles() { #if 0 MS_TRACE(); @@ -298,9 +293,9 @@ namespace RTC goto error; } - DtlsTransport::certificate = PEM_read_X509(file, nullptr, nullptr, nullptr); + certificate = PEM_read_X509(file, nullptr, nullptr, nullptr); - if (!DtlsTransport::certificate) + if (!certificate) { LOG_OPENSSL_ERROR("PEM_read_X509() failed"); @@ -318,9 +313,9 @@ namespace RTC goto error; } - DtlsTransport::privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr); + privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr); - if (!DtlsTransport::privateKey) + if (!privateKey) { LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed"); @@ -337,7 +332,7 @@ namespace RTC #endif } - void DtlsTransport::CreateSslCtx() + void DtlsTransport::DtlsEnvironment::CreateSslCtx() { MS_TRACE(); @@ -347,16 +342,16 @@ namespace RTC /* Set the global DTLS context. */ // Both DTLS 1.0 and 1.2 (requires OpenSSL >= 1.1.0). - DtlsTransport::sslCtx = SSL_CTX_new(DTLS_method()); + sslCtx = SSL_CTX_new(DTLS_method()); - if (!DtlsTransport::sslCtx) + if (!sslCtx) { LOG_OPENSSL_ERROR("SSL_CTX_new() failed"); goto error; } - ret = SSL_CTX_use_certificate(DtlsTransport::sslCtx, DtlsTransport::certificate); + ret = SSL_CTX_use_certificate(sslCtx, certificate); if (ret == 0) { @@ -365,7 +360,7 @@ namespace RTC goto error; } - ret = SSL_CTX_use_PrivateKey(DtlsTransport::sslCtx, DtlsTransport::privateKey); + ret = SSL_CTX_use_PrivateKey(sslCtx, privateKey); if (ret == 0) { @@ -374,7 +369,7 @@ namespace RTC goto error; } - ret = SSL_CTX_check_private_key(DtlsTransport::sslCtx); + ret = SSL_CTX_check_private_key(sslCtx); if (ret == 0) { @@ -385,31 +380,31 @@ namespace RTC // Set options. SSL_CTX_set_options( - DtlsTransport::sslCtx, + sslCtx, SSL_OP_CIPHER_SERVER_PREFERENCE | SSL_OP_NO_TICKET | SSL_OP_SINGLE_ECDH_USE | SSL_OP_NO_QUERY_MTU); // Don't use sessions cache. - SSL_CTX_set_session_cache_mode(DtlsTransport::sslCtx, SSL_SESS_CACHE_OFF); + SSL_CTX_set_session_cache_mode(sslCtx, SSL_SESS_CACHE_OFF); // Read always as much into the buffer as possible. // NOTE: This is the default for DTLS, but a bug in non latest OpenSSL // versions makes this call required. - SSL_CTX_set_read_ahead(DtlsTransport::sslCtx, 1); + SSL_CTX_set_read_ahead(sslCtx, 1); - SSL_CTX_set_verify_depth(DtlsTransport::sslCtx, 4); + SSL_CTX_set_verify_depth(sslCtx, 4); // Require certificate from peer. SSL_CTX_set_verify( - DtlsTransport::sslCtx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, onSslCertificateVerify); + sslCtx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, onSslCertificateVerify); // Set SSL info callback. - SSL_CTX_set_info_callback(DtlsTransport::sslCtx, [](const SSL* ssl, int where, int ret){ + SSL_CTX_set_info_callback(sslCtx, [](const SSL* ssl, int where, int ret){ static_cast(SSL_get_ex_data(ssl, 0))->OnSslInfo(where, ret); }); // Set ciphers. ret = SSL_CTX_set_cipher_list( - DtlsTransport::sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); + sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); if (ret == 0) { @@ -424,7 +419,7 @@ namespace RTC // NOTE: https://bugs.ruby-lang.org/issues/12324 // For OpenSSL >= 1.0.2. - SSL_CTX_set_ecdh_auto(DtlsTransport::sslCtx, 1); + SSL_CTX_set_ecdh_auto(sslCtx, 1); // Set the "use_srtp" DTLS extension. for (auto it = DtlsTransport::srtpCryptoSuites.begin(); @@ -441,7 +436,7 @@ namespace RTC MS_DEBUG_2TAGS(dtls, srtp, "setting SRTP cryptoSuites for DTLS: %s", dtlsSrtpCryptoSuites.c_str()); // NOTE: This function returns 0 on success. - ret = SSL_CTX_set_tlsext_use_srtp(DtlsTransport::sslCtx, dtlsSrtpCryptoSuites.c_str()); + ret = SSL_CTX_set_tlsext_use_srtp(sslCtx, dtlsSrtpCryptoSuites.c_str()); if (ret != 0) { @@ -456,16 +451,16 @@ namespace RTC error: - if (DtlsTransport::sslCtx) + if (sslCtx) { - SSL_CTX_free(DtlsTransport::sslCtx); - DtlsTransport::sslCtx = nullptr; + SSL_CTX_free(sslCtx); + sslCtx = nullptr; } MS_THROW_ERROR("SSL context creation failed"); } - void DtlsTransport::GenerateFingerprints() + void DtlsTransport::DtlsEnvironment::GenerateFingerprints() { MS_TRACE(); @@ -505,7 +500,7 @@ namespace RTC MS_THROW_ERROR("unknown algorithm"); } - ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size); + ret = X509_digest(certificate, hashFunction, binaryFingerprint, &size); if (ret == 0) { @@ -528,7 +523,7 @@ namespace RTC fingerprint.algorithm = DtlsTransport::GetFingerprintAlgorithm(algorithmString); fingerprint.value = hexFingerprint; - DtlsTransport::localFingerprints.push_back(fingerprint); + localFingerprints.push_back(fingerprint); } } @@ -537,10 +532,11 @@ namespace RTC DtlsTransport::DtlsTransport(EventPoller::Ptr poller,Listener* listener) : poller(std::move(poller)), listener(listener) { MS_TRACE(); + env = DtlsEnvironment::Instance().shared_from_this(); /* Set SSL. */ - this->ssl = SSL_new(DtlsTransport::sslCtx); + this->ssl = SSL_new(env->sslCtx); if (!this->ssl) { diff --git a/webrtc/DtlsTransport.hpp b/webrtc/DtlsTransport.hpp index 07359583..f1b10ca7 100644 --- a/webrtc/DtlsTransport.hpp +++ b/webrtc/DtlsTransport.hpp @@ -14,7 +14,7 @@ using namespace toolkit; namespace RTC { -class DtlsTransport : public std::enable_shared_from_this + class DtlsTransport : public std::enable_shared_from_this { public: enum class DtlsState @@ -60,6 +60,27 @@ class DtlsTransport : public std::enable_shared_from_this const char* name; }; + class DtlsEnvironment : public std::enable_shared_from_this + { + public: + using Ptr = std::shared_ptr; + ~DtlsEnvironment(); + static DtlsEnvironment& Instance(); + + private: + DtlsEnvironment(); + void GenerateCertificateAndPrivateKey(); + void ReadCertificateAndPrivateKeyFromFiles(); + void CreateSslCtx(); + void GenerateFingerprints(); + + public: + X509* certificate{ nullptr }; + EVP_PKEY* privateKey{ nullptr }; + SSL_CTX* sslCtx{ nullptr }; + std::vector localFingerprints; + }; + public: class Listener { @@ -93,8 +114,6 @@ class DtlsTransport : public std::enable_shared_from_this }; public: - static void ClassInit(); - static void ClassDestroy(); static Role StringToRole(const std::string& role) { auto it = DtlsTransport::string2Role.find(role); @@ -132,20 +151,9 @@ class DtlsTransport : public std::enable_shared_from_this } private: - static void GenerateCertificateAndPrivateKey(); - static void ReadCertificateAndPrivateKeyFromFiles(); - static void CreateSslCtx(); - static void GenerateFingerprints(); - - private: - static X509* certificate; - static EVP_PKEY* privateKey; - static SSL_CTX* sslCtx; - static uint8_t sslReadBuffer[]; static std::map string2Role; static std::map string2FingerprintAlgorithm; static std::map fingerprintAlgorithm2String; - static std::vector localFingerprints; static std::vector srtpCryptoSuites; public: @@ -157,7 +165,7 @@ class DtlsTransport : public std::enable_shared_from_this void Run(Role localRole); std::vector& GetLocalFingerprints() const { - return DtlsTransport::localFingerprints; + return env->localFingerprints; } bool SetRemoteFingerprint(Fingerprint fingerprint); void ProcessDtlsData(const uint8_t* data, size_t len); @@ -203,6 +211,7 @@ class DtlsTransport : public std::enable_shared_from_this void OnTimer(); private: + DtlsEnvironment::Ptr env; EventPoller::Ptr poller; // Passed by argument. Listener* listener{ nullptr }; @@ -218,7 +227,9 @@ class DtlsTransport : public std::enable_shared_from_this bool handshakeDone{ false }; bool handshakeDoneNow{ false }; std::string remoteCert; - }; + static constexpr int SslReadBufferSize{ 65536 }; + uint8_t sslReadBuffer[SslReadBufferSize]; +}; } // namespace RTC #endif diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index f727876c..77c0c28f 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -3,10 +3,6 @@ #include "Rtcp/Rtcp.h" WebRtcTransport::WebRtcTransport() { - static onceToken token([](){ - RTC::DtlsTransport::ClassInit(); - }); - dtls_transport_ = std::make_shared(EventPollerPool::Instance().getFirstPoller(), this); ice_server_ = std::make_shared(this, makeRandStr(4), makeRandStr(24)); } From cc960a324f6806114840259f194e18602d480102 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 10:07:06 +0800 Subject: [PATCH 007/218] =?UTF-8?q?=E6=94=AF=E6=8C=81=E5=A4=9A=E7=BA=BF?= =?UTF-8?q?=E7=A8=8B?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/DtlsTransport.hpp | 3 ++- webrtc/IceServer.cpp | 4 ---- webrtc/IceServer.hpp | 3 +++ 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/webrtc/DtlsTransport.hpp b/webrtc/DtlsTransport.hpp index f1b10ca7..b3fa244e 100644 --- a/webrtc/DtlsTransport.hpp +++ b/webrtc/DtlsTransport.hpp @@ -227,7 +227,8 @@ namespace RTC bool handshakeDone{ false }; bool handshakeDoneNow{ false }; std::string remoteCert; - static constexpr int SslReadBufferSize{ 65536 }; + //最大不超过mtu + static constexpr int SslReadBufferSize{ 1600 }; uint8_t sslReadBuffer[SslReadBufferSize]; }; } // namespace RTC diff --git a/webrtc/IceServer.cpp b/webrtc/IceServer.cpp index 38963c58..d80ef0f0 100644 --- a/webrtc/IceServer.cpp +++ b/webrtc/IceServer.cpp @@ -7,10 +7,6 @@ namespace RTC { /* Static. */ - - static constexpr size_t StunSerializeBufferSize{ 65536 }; - static uint8_t StunSerializeBuffer[StunSerializeBufferSize]; - /* Instance methods. */ IceServer::IceServer(Listener* listener, const std::string& usernameFragment, const std::string& password) diff --git a/webrtc/IceServer.hpp b/webrtc/IceServer.hpp index ad5d93eb..103727f0 100644 --- a/webrtc/IceServer.hpp +++ b/webrtc/IceServer.hpp @@ -106,6 +106,9 @@ namespace RTC IceState state{ IceState::NEW }; std::list tuples; RTC::TransportTuple* selectedTuple{ nullptr }; + //最大不超过mtu + static constexpr size_t StunSerializeBufferSize{ 1600 }; + uint8_t StunSerializeBuffer[StunSerializeBufferSize]; }; } // namespace RTC From 272ab706b07ba619bc87f1ea2f32006a09c548c3 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 10:16:49 +0800 Subject: [PATCH 008/218] =?UTF-8?q?=E7=A1=AE=E4=BF=9D=E5=AE=89=E5=85=A8?= =?UTF-8?q?=E9=87=8A=E6=94=BE=E8=B5=84=E6=BA=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 12 ++++-------- webrtc/WebRtcTransport.cpp | 19 +++++++++++++++---- webrtc/WebRtcTransport.h | 11 ++++++++--- 3 files changed, 27 insertions(+), 15 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 3eacb1eb..055a8f0f 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1062,14 +1062,10 @@ void installWebApi() { } headerOut["Content-Type"] = "text/plain"; headerOut["Access-Control-Allow-Origin"] = "*"; - auto poller = EventPollerPool::Instance().getFirstPoller(); - auto rtc = std::make_shared(poller); - poller->async([invoker, rtc, headerOut, src]() { - rtc->attach(src); - auto sdp = rtc->GetLocalSdp(); - invoker(200, headerOut, sdp); - rtcs.emplace_back(rtc); - }); + auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); + rtc->attach(src); + invoker(200, headerOut, rtc->GetLocalSdp()); + rtcs.emplace_back(rtc); }); #endif diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 77c0c28f..5754f34f 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -2,12 +2,12 @@ #include #include "Rtcp/Rtcp.h" -WebRtcTransport::WebRtcTransport() { - dtls_transport_ = std::make_shared(EventPollerPool::Instance().getFirstPoller(), this); +WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { + dtls_transport_ = std::make_shared(poller, this); ice_server_ = std::make_shared(this, makeRandStr(4), makeRandStr(24)); } -WebRtcTransport::~WebRtcTransport() { +void WebRtcTransport::onDestory(){ dtls_transport_ = nullptr; ice_server_ = nullptr; } @@ -163,8 +163,15 @@ void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { } /////////////////////////////////////////////////////////////////////////////////// +WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &poller){ + WebRtcTransportImp::Ptr ret(new WebRtcTransportImp(poller), [](WebRtcTransportImp *ptr){ + ptr->onDestory(); + delete ptr; + }); + return ret; +} -WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) { +WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) { _socket = Socket::createSocket(poller, false); //随机端口，绑定全部网卡 _socket->bindUdpSock(0); @@ -173,6 +180,10 @@ WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) { }); } +void WebRtcTransportImp::onDestory() { + WebRtcTransport::onDestory(); +} + void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { assert(src); _src = src; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 80e0f639..7f32612a 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -11,8 +11,11 @@ class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: using Ptr = std::shared_ptr; - WebRtcTransport(); - virtual ~WebRtcTransport(); + WebRtcTransport(const EventPoller::Ptr &poller); + ~WebRtcTransport() override = default; + + /// 销毁对象 + virtual void onDestory(); /// 获取本地sdp /// \return @@ -85,7 +88,7 @@ class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_fro public: using Ptr = std::shared_ptr; - WebRtcTransportImp(const EventPoller::Ptr &poller); + static Ptr create(const EventPoller::Ptr &poller); ~WebRtcTransportImp() override = default; void attach(const RtspMediaSource::Ptr &src); @@ -97,6 +100,8 @@ protected: uint16_t getPort() const override; std::string getIP() const override; void onDtlsConnected() override; + WebRtcTransportImp(const EventPoller::Ptr &poller); + void onDestory() override; private: Socket::Ptr _socket; From 762cf227e5f9455302ecee15e35691898d784b73 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Sat, 27 Mar 2021 18:33:20 +0800 Subject: [PATCH 009/218] =?UTF-8?q?=E5=88=9D=E6=AD=A5=E6=B7=BB=E5=8A=A0sdp?= =?UTF-8?q?=E7=9B=B8=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 5 ++ webrtc/Sdp.h | 173 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 178 insertions(+) create mode 100644 webrtc/Sdp.cpp create mode 100644 webrtc/Sdp.h diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp new file mode 100644 index 00000000..484a9980 --- /dev/null +++ b/webrtc/Sdp.cpp @@ -0,0 +1,5 @@ +// +// Created by xzl on 2021/3/27. +// + +#include "Sdp.h" diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h new file mode 100644 index 00000000..6756b11c --- /dev/null +++ b/webrtc/Sdp.h @@ -0,0 +1,173 @@ +// +// Created by xzl on 2021/3/27. +// + +#ifndef ZLMEDIAKIT_SDP_H +#define ZLMEDIAKIT_SDP_H + +#include +#include +using namespace std; + +enum class RtpDirection { + invalid = -1, + //只发送 + sendonly, + //只接收 + revonly, + //同时发送接收 + sendrecv, + //禁止发送数据 + inactive +}; + +enum class DtlsRole { + invalid = -1, + //客户端 + active, + //服务端 + passive, + //既可作做客户端也可以做服务端 + actpass, +}; + +enum class SdpType { + invalid = -1, + offer, + answer +}; + +// +//v=0 +//o=- 7268199939077294076 2 IN IP4 127.0.0.1 +//s=- +//t=0 0 +//a=group:BUNDLE video +//a=msid-semantic: WMS +//m=video 9 RTP/SAVPF 96 +//c=IN IP4 0.0.0.0 +//a=rtcp:9 IN IP4 0.0.0.0 +//a=ice-ufrag:y94W +//a=ice-pwd:fuz1hZCAarezk35fruVGfdyP +//a=ice-options:trickle +//a=fingerprint:sha-256 FF:8C:29:B8:B3:2B:45:F5:21:D2:47:D5:EE:B7:F8:BB:F1:DC:95:47:7B:20:B4:59:75:0F:16:93:D0:AC:D2:73 +//a=setup:active +//a=mid:video +//a=recvonly +//a=rtcp-mux +//a=rtpmap:96 H264/90000 +//a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f + +//Session description +// v= (protocol version) +// o= (originator and session identifier) +// s= (session name) +// i=* (session information) +// u=* (URI of description) +// e=* (email address) +// p=* (phone number) +// c=* (connection information -- not required if included in +// all media) +// b=* (zero or more bandwidth information lines) +// One or more time descriptions ("t=" and "r=" lines; see below) +// z=* (time zone adjustments) +// k=* (encryption key) +// a=* (zero or more session attribute lines) +// Zero or more media descriptions +// +// Time description +// t= (time the session is active) +// r=* (zero or more repeat times) +// +// Media description, if present +// m= (media name and transport address) +// i=* (media title) +// c=* (connection information -- optional if included at +// session level) +// b=* (zero or more bandwidth information lines) +// k=* (encryption key) +// a=* (zero or more media attribute lines) + +class SdpItem { +public: + SdpItem() = default; + virtual ~SdpItem() = default; + virtual void parse(const string &str) = 0; + virtual string toString() const = 0; +}; + +class SdpTime : public SdpItem{ +public: + float start; + float end; + void parse(const string &str) override; + string toString() const override; +}; + +class SdpAttr : public SdpItem{ +public: + string name; + string value; + void parse(const string &str) override; + string toString() const override; +}; + +class SdpOrigin : public SdpItem{ +public: + // https://datatracker.ietf.org/doc/rfc4566/?include_text=1 5.2. Origin ("o=") + // o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5 + // o= + string username; + string session_id; + string session_version; + string nettype; + string addrtype; + string address; + void parse(const string &str) override; + string toString() const override; +}; + +class SdpConnection : public SdpItem { +public: + // https://datatracker.ietf.org/doc/rfc4566/?include_text=1 5.7. Connection Data ("c=") + // c=IN IP4 224.2.17.12/127 + // c= + string nettype; + string addrtype; + string address; + void parse(const string &str) override; + string toString() const override; +}; + +class RtcMedia { +public: + +}; +class RtcSdp { +public: + /////Session description（会话级别描述）//// + //v= (protocol version) + int version; + //o= (session origin information ) + SdpOrigin origin; + //s= (session name) + string session_name; + //i=* (session information) + string session_information; + //c=* (connection information -- not required if included in all media) + SdpConnection connection; + + //// Time description //// + //t= (time the session is active) + SdpTime time; + //r=* (zero or more repeat times) + int repeat; + //a=* (zero or more media attribute lines) + list attributes; + +}; + + + + +#endif //ZLMEDIAKIT_SDP_H From 8aa3ff425be5d58d5a1939ac09f4ceb94c5acf07 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 27 Mar 2021 22:23:38 +0800 Subject: [PATCH 010/218] =?UTF-8?q?=E5=AE=8C=E5=96=84sdp=E7=9B=B8=E5=85=B3?= =?UTF-8?q?=E5=AE=9A=E4=B9=89?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/Frame.h | 3 +- webrtc/Sdp.h | 284 ++++++++++++++++++++++++++++++--------- webrtc/WebRtcTransport.h | 2 +- 3 files changed, 223 insertions(+), 66 deletions(-) diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index 4988ac06..5f60a68d 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -38,7 +38,8 @@ typedef enum { TrackVideo = 0, TrackAudio, TrackTitle, - TrackMax = 3 + TrackData, + TrackMax = 4 } TrackType; /** diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 6756b11c..dfe47c54 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -5,58 +5,112 @@ #ifndef ZLMEDIAKIT_SDP_H #define ZLMEDIAKIT_SDP_H -#include #include +#include +#include "Extension/Frame.h" using namespace std; +using namespace mediakit; -enum class RtpDirection { - invalid = -1, - //只发送 - sendonly, - //只接收 - revonly, - //同时发送接收 - sendrecv, - //禁止发送数据 - inactive -}; +//https://datatracker.ietf.org/doc/rfc4566/?include_text=1 -enum class DtlsRole { - invalid = -1, - //客户端 - active, - //服务端 - passive, - //既可作做客户端也可以做服务端 - actpass, -}; - -enum class SdpType { - invalid = -1, - offer, - answer -}; - -// //v=0 -//o=- 7268199939077294076 2 IN IP4 127.0.0.1 +//o=- 2584450093346841581 2 IN IP4 127.0.0.1 //s=- //t=0 0 -//a=group:BUNDLE video -//a=msid-semantic: WMS -//m=video 9 RTP/SAVPF 96 +//a=group:BUNDLE audio video data +//a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549 +//m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 //c=IN IP4 0.0.0.0 //a=rtcp:9 IN IP4 0.0.0.0 -//a=ice-ufrag:y94W -//a=ice-pwd:fuz1hZCAarezk35fruVGfdyP -//a=ice-options:trickle -//a=fingerprint:sha-256 FF:8C:29:B8:B3:2B:45:F5:21:D2:47:D5:EE:B7:F8:BB:F1:DC:95:47:7B:20:B4:59:75:0F:16:93:D0:AC:D2:73 -//a=setup:active -//a=mid:video -//a=recvonly +//a=ice-ufrag:sXJ3 +//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV +//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 +//a=setup:actpass +//a=mid:audio +//a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level +//a=sendrecv //a=rtcp-mux -//a=rtpmap:96 H264/90000 -//a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f +//a=rtpmap:111 opus/48000/2 +//a=rtcp-fb:111 transport-cc +//a=fmtp:111 minptime=10;useinbandfec=1 +//a=rtpmap:103 ISAC/16000 +//a=rtpmap:104 ISAC/32000 +//a=rtpmap:9 G722/8000 +//a=rtpmap:0 PCMU/8000 +//a=rtpmap:8 PCMA/8000 +//a=rtpmap:106 CN/32000 +//a=rtpmap:105 CN/16000 +//a=rtpmap:13 CN/8000 +//a=rtpmap:110 telephone-event/48000 +//a=rtpmap:112 telephone-event/32000 +//a=rtpmap:113 telephone-event/16000 +//a=rtpmap:126 telephone-event/8000 +//a=ssrc:120276603 cname:iSkJ2vn5cYYubTve +//a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363 +//a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 +//a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363 +//m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125 +//c=IN IP4 0.0.0.0 +//a=rtcp:9 IN IP4 0.0.0.0 +//a=ice-ufrag:sXJ3 +//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV +//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 +//a=setup:actpass +//a=mid:video +//a=extmap:2 urn:ietf:params:rtp-hdrext:toffset +//a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +//a=extmap:4 urn:3gpp:video-orientation +//a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +//a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +//a=sendrecv +//a=rtcp-mux +//a=rtcp-rsize +//a=rtpmap:96 VP8/90000 +//a=rtcp-fb:96 ccm fir +//a=rtcp-fb:96 nack +//a=rtcp-fb:96 nack pli +//a=rtcp-fb:96 goog-remb +//a=rtcp-fb:96 transport-cc +//a=rtpmap:98 VP9/90000 +//a=rtcp-fb:98 ccm fir +//a=rtcp-fb:98 nack +//a=rtcp-fb:98 nack pli +//a=rtcp-fb:98 goog-remb +//a=rtcp-fb:98 transport-cc +//a=rtpmap:100 H264/90000 +//a=rtcp-fb:100 ccm fir +//a=rtcp-fb:100 nack +//a=rtcp-fb:100 nack pli +//a=rtcp-fb:100 goog-remb +//a=rtcp-fb:100 transport-cc +//a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f +//a=rtpmap:102 red/90000 +//a=rtpmap:127 ulpfec/90000 +//a=rtpmap:97 rtx/90000 +//a=fmtp:97 apt=96 +//a=rtpmap:99 rtx/90000 +//a=fmtp:99 apt=98 +//a=rtpmap:101 rtx/90000 +//a=fmtp:101 apt=100 +//a=rtpmap:125 rtx/90000 +//a=fmtp:125 apt=102 +//a=ssrc-group:FID 2580761338 611523443 +//a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve +//a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961 +//a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 +//a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961 +//a=ssrc:611523443 cname:iSkJ2vn5cYYubTve +//a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961 +//a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 +//a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961 +//m=application 9 DTLS/SCTP 5000 +//c=IN IP4 0.0.0.0 +//a=ice-ufrag:sXJ3 +//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV +//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 +//a=setup:actpass +//a=mid:data +//a=sctpmap:5000 webrtc-datachannel 1024 //Session description // v= (protocol version) @@ -88,33 +142,57 @@ enum class SdpType { // k=* (encryption key) // a=* (zero or more media attribute lines) +enum class RtpDirection { + invalid = -1, + //只发送 + sendonly, + //只接收 + revonly, + //同时发送接收 + sendrecv, + //禁止发送数据 + inactive +}; + +enum class DtlsRole { + invalid = -1, + //客户端 + active, + //服务端 + passive, + //既可作做客户端也可以做服务端 + actpass, +}; + +enum class SdpType { + invalid = -1, + offer, + answer +}; + class SdpItem { public: SdpItem() = default; virtual ~SdpItem() = default; virtual void parse(const string &str) = 0; virtual string toString() const = 0; + virtual const char* getKey() = 0; }; class SdpTime : public SdpItem{ public: - float start; - float end; - void parse(const string &str) override; - string toString() const override; -}; - -class SdpAttr : public SdpItem{ -public: - string name; - string value; + //5.9. Timing ("t=") + // t= + uint64_t start; + uint64_t stop; void parse(const string &str) override; string toString() const override; + const char* getKey() override { return "t";} }; class SdpOrigin : public SdpItem{ public: - // https://datatracker.ietf.org/doc/rfc4566/?include_text=1 5.2. Origin ("o=") + // 5.2. Origin ("o=") // o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5 // o= string username; @@ -125,11 +203,12 @@ public: string address; void parse(const string &str) override; string toString() const override; + const char* getKey() override { return "o";} }; class SdpConnection : public SdpItem { public: - // https://datatracker.ietf.org/doc/rfc4566/?include_text=1 5.7. Connection Data ("c=") + // 5.7. Connection Data ("c=") // c=IN IP4 224.2.17.12/127 // c= string nettype; @@ -137,12 +216,75 @@ public: string address; void parse(const string &str) override; string toString() const override; + const char* getKey() override { return "c";} +}; + +class SdpBandwidth : public SdpItem { +public: + //5.8. Bandwidth ("b=") + //b=: + + //AS、CT + string bwtype; + int bandwidth; + + void parse(const string &str) override; + string toString() const override; + const char* getKey() override { return "b";} +}; + +class SdpMedia : public SdpItem { +public: + // 5.14. Media Descriptions ("m=") + // m= ... + TrackType type; + uint16_t port; + vector proto; + vector fmt; + + void parse(const string &str) override; + string toString() const override; + const char* getKey() override { return "m";} +}; + +class SdpAttr : public SdpItem{ +public: + //5.13. Attributes ("a=") + //a= + //a=: + string name; + string value; + void parse(const string &str) override; + string toString() const override; + const char* getKey() override { return "a";} +}; + +class SdpAttrGroup : public SdpItem{ +public: + //a=group:BUNDLE line with all the 'mid' identifiers part of the + // BUNDLE group is included at the session-level. + //a=group:LS session level attribute MUST be included wth the 'mid' + // identifiers that are part of the same lip sync group. + string type; + vector mid; + void parse(const string &str) override; + string toString() const override; + const char* getKey() override { return "group";} }; class RtcMedia { public: + //m= ... + SdpMedia media; + //c= + SdpConnection connection; + //a=: + vector attributes; + bool haveAttr(const char *attr) const; + string getAttrValue(const char *attr) const; }; + class RtcSdp { public: /////Session description（会话级别描述）//// @@ -152,19 +294,33 @@ public: SdpOrigin origin; //s= (session name) string session_name; - //i=* (session information) - string session_information; - //c=* (connection information -- not required if included in all media) - SdpConnection connection; - - //// Time description //// //t= (time the session is active) SdpTime time; - //r=* (zero or more repeat times) - int repeat; - //a=* (zero or more media attribute lines) - list attributes; + //// 非必须 //// + //i=* (session information) + string information; + //u=* (URI of description) + string url; + //e=* (email address) + string email; + //p=* (phone number) + string phone; + //c=* (connection information -- not required if included in all media) + SdpConnection connection; + //b=* (zero or more bandwidth information lines) + SdpBandwidth bandwidth; + //z=* (time zone adjustments) + //z= .... + string time_zone; + //k=* (encryption key) + //k=, k=: + string crypt_key; + //r=* (zero or more repeat times) + //r= + string repeat; + //a=* (zero or more media attribute lines) + vector attributes; }; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 7f32612a..2842bee8 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -95,7 +95,7 @@ public: protected: void onWrite(const char *buf, size_t len, struct sockaddr_in *dst) override; - int getPayloadType() const ; + int getPayloadType() const override; uint32_t getSSRC() const override; uint16_t getPort() const override; std::string getIP() const override; From 3dac9e1c518c029c7d33d025634af0d995854d00 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 27 Mar 2021 23:42:44 +0800 Subject: [PATCH 011/218] =?UTF-8?q?srtp=E5=86=85=E9=83=A8=E6=97=A5?= =?UTF-8?q?=E5=BF=97=E7=9B=B8=E5=85=B3=E8=AE=BE=E7=BD=AE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/SrtpSession.cpp | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index 580c190b..a970fd7c 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -5,6 +5,7 @@ #include // std::memset(), std::memcpy() #include "logger.h" #include "Util/util.h" +#include "Util/logger.h" using namespace toolkit; namespace RTC @@ -60,6 +61,23 @@ namespace RTC srtp_err_status_t err = srtp_init(); +#if 0 + srtp_install_log_handler([](srtp_log_level_t level, + const char *msg, + void *data) { + printf("%s\n", msg); + }, nullptr); + srtp_set_debug_module("srtp", 1); + srtp_set_debug_module("hmac sha-1", 1); + srtp_set_debug_module("aes icm", 1); + srtp_set_debug_module("alloc", 1); + srtp_set_debug_module("stat test", 1); + srtp_set_debug_module("cipher", 1); + srtp_set_debug_module("auth func", 1); + srtp_set_debug_module("crypto kernel", 1); + srtp_list_debug_modules(); +#endif + if (DepLibSRTP::IsError(err)) { MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err)); } From f4746d161cc08c2d245faa254f5d33c748ba6c65 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 28 Mar 2021 09:49:34 +0800 Subject: [PATCH 012/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0sdp=E8=87=AA=E5=8A=A8?= =?UTF-8?q?=E5=8C=96=E8=A7=A3=E6=9E=90=E6=9C=BA=E5=88=B6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 228 +++++++++++++++++++++++++++++++++++++++++++++++++ webrtc/Sdp.h | 215 ++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 416 insertions(+), 27 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 484a9980..294c84a6 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -3,3 +3,231 @@ // #include "Sdp.h" + +using onCreateSdpItem = function; +static unordered_map sdpItemCreator; + +template +void registerSdpItem(){ + onCreateSdpItem func = [](const string &key, const string &value) { + auto ret = std::make_shared(); + ret->parse(value); + return ret; + }; + Item item; + InfoL << "register sdp item:" << item.getKey(); + sdpItemCreator.emplace(item.getKey(), std::move(func)); +} + +static bool registerAllItem(){ + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem >(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + return true; +} + +void RtcSdp::parse(const string &str) { + static auto flag = registerAllItem(); + RtcMedia *media = nullptr; + auto lines = split(str, "\n"); + for(auto &line : lines){ + trim(line); + if(line.size() < 3 || line[1] != '='){ + continue; + } + auto key = line.substr(0, 1); + auto value = line.substr(2); + if (key == "m") { + medias.emplace_back(RtcMedia()); + media = &medias.back(); + } + + SdpItem::Ptr item; + auto it = sdpItemCreator.find(key); + if (it != sdpItemCreator.end()) { + item = it->second(key, value); + } else { + item = std::make_shared(key); + item->parse(value); + } + if (media) { + media->items.push_back(std::move(item)); + } else { + items.push_back(std::move(item)); + } + } +} + +string RtcSdp::toString() const { + return std::string(); +} + +void SdpAttr::parse(const string &str) { + SdpItem::parse(str); + auto pos = str.find(':'); + auto key = pos == string::npos ? str : str.substr(0, pos); + auto value = pos == string::npos ? string() : str.substr(pos + 1); + auto it = sdpItemCreator.find(key); + if (it != sdpItemCreator.end()) { + detail = it->second(key, value); + } else { + detail = std::make_shared(key); + detail->parse(value); + } +} + +void test_sdp(){ + char str1[] = "v=0\n" + "o=- 380154348540553537 2 IN IP4 127.0.0.1\n" + "s=-\n" + "t=0 0\n" + "a=group:BUNDLE video\n" + "a=msid-semantic: WMS\n" + "m=video 9 RTP/SAVPF 96\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:1ZFN\n" + "a=ice-pwd:70P3H0jPlGz1fiJl5XZfXMZH\n" + "a=ice-options:trickle\n" + "a=fingerprint:sha-256 3E:10:35:6B:9A:9E:B0:55:AC:2A:88:F5:74:C1:70:32:B5:8D:88:1D:37:B0:9C:69:A6:DD:07:10:73:27:1A:16\n" + "a=setup:active\n" + "a=mid:video\n" + "a=recvonly\n" + "a=rtcp-mux\n" + "a=rtpmap:96 H264/90000\n" + "a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f"; + char str2[] = "v=0\n" + "o=- 2584450093346841581 2 IN IP4 127.0.0.1\n" + "s=-\n" + "t=0 0\n" + "a=group:BUNDLE audio video data\n" + "a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:audio\n" + "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" + "a=sendrecv\n" + "a=rtcp-mux\n" + "a=rtpmap:111 opus/48000/2\n" + "a=rtcp-fb:111 transport-cc\n" + "a=fmtp:111 minptime=10;useinbandfec=1\n" + "a=rtpmap:103 ISAC/16000\n" + "a=rtpmap:104 ISAC/32000\n" + "a=rtpmap:9 G722/8000\n" + "a=rtpmap:0 PCMU/8000\n" + "a=rtpmap:8 PCMA/8000\n" + "a=rtpmap:106 CN/32000\n" + "a=rtpmap:105 CN/16000\n" + "a=rtpmap:13 CN/8000\n" + "a=rtpmap:110 telephone-event/48000\n" + "a=rtpmap:112 telephone-event/32000\n" + "a=rtpmap:113 telephone-event/16000\n" + "a=rtpmap:126 telephone-event/8000\n" + "a=ssrc:120276603 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363\n" + "a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363\n" + "m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:video\n" + "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n" + "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\n" + "a=extmap:4 urn:3gpp:video-orientation\n" + "a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\n" + "a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\n" + "a=sendrecv\n" + "a=rtcp-mux\n" + "a=rtcp-rsize\n" + "a=rtpmap:96 VP8/90000\n" + "a=rtcp-fb:96 ccm fir\n" + "a=rtcp-fb:96 nack\n" + "a=rtcp-fb:96 nack pli\n" + "a=rtcp-fb:96 goog-remb\n" + "a=rtcp-fb:96 transport-cc\n" + "a=rtpmap:98 VP9/90000\n" + "a=rtcp-fb:98 ccm fir\n" + "a=rtcp-fb:98 nack\n" + "a=rtcp-fb:98 nack pli\n" + "a=rtcp-fb:98 goog-remb\n" + "a=rtcp-fb:98 transport-cc\n" + "a=rtpmap:100 H264/90000\n" + "a=rtcp-fb:100 ccm fir\n" + "a=rtcp-fb:100 nack\n" + "a=rtcp-fb:100 nack pli\n" + "a=rtcp-fb:100 goog-remb\n" + "a=rtcp-fb:100 transport-cc\n" + "a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\n" + "a=rtpmap:102 red/90000\n" + "a=rtpmap:127 ulpfec/90000\n" + "a=rtpmap:97 rtx/90000\n" + "a=fmtp:97 apt=96\n" + "a=rtpmap:99 rtx/90000\n" + "a=fmtp:99 apt=98\n" + "a=rtpmap:101 rtx/90000\n" + "a=fmtp:101 apt=100\n" + "a=rtpmap:125 rtx/90000\n" + "a=fmtp:125 apt=102\n" + "a=ssrc-group:FID 2580761338 611523443\n" + "a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:611523443 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=candidate:1 1 udp %u %s %u typ host\n" + "m=application 9 DTLS/SCTP 5000\n" + "c=IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:data\n" + "a=sctpmap:5000 webrtc-datachannel 1024"; + + RtcSdp sdp1; + sdp1.parse(str1); + + RtcSdp sdp2; + sdp2.parse(str2); + + InfoL << sdp1.toString(); + InfoL << sdp2.toString(); +} diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index dfe47c54..aafdae04 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -103,6 +103,7 @@ using namespace mediakit; //a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961 //a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 //a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961 +//a=candidate:1 1 udp %u %s %u typ host //m=application 9 DTLS/SCTP 5000 //c=IN IP4 0.0.0.0 //a=ice-ufrag:sXJ3 @@ -172,21 +173,40 @@ enum class SdpType { class SdpItem { public: - SdpItem() = default; + using Ptr = std::shared_ptr; + string value; virtual ~SdpItem() = default; - virtual void parse(const string &str) = 0; - virtual string toString() const = 0; + virtual void parse(const string &str) { + value = str; + } + virtual string toString() const { + return value; + } virtual const char* getKey() = 0; }; +template +class SdpString : public SdpItem{ +public: + // *=* + const char* getKey() override { static string key(1, KEY); return key.data();} +}; + +class SdpCommon : public SdpItem { +public: + string key; + SdpCommon(string key) { this->key = std::move(key); } + const char* getKey() override { return key.data();} +}; + class SdpTime : public SdpItem{ public: //5.9. Timing ("t=") // t= uint64_t start; uint64_t stop; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "t";} }; @@ -201,8 +221,8 @@ public: string nettype; string addrtype; string address; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "o";} }; @@ -214,8 +234,8 @@ public: string nettype; string addrtype; string address; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "c";} }; @@ -228,8 +248,8 @@ public: string bwtype; int bandwidth; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "b";} }; @@ -242,20 +262,20 @@ public: vector proto; vector fmt; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "m";} }; class SdpAttr : public SdpItem{ public: + using Ptr = std::shared_ptr; //5.13. Attributes ("a=") //a= //a=: - string name; - string value; + SdpItem::Ptr detail; void parse(const string &str) override; - string toString() const override; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "a";} }; @@ -267,26 +287,168 @@ public: // identifiers that are part of the same lip sync group. string type; vector mid; - void parse(const string &str) override; - string toString() const override; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "group";} }; +class SdpAttrMsidSemantic : public SdpItem { +public: + string name{"WMS"}; + string mid; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "msid-semantic";} +}; + +class SdpAttrRtcp : public SdpItem { +public: + // c=IN IP4 224.2.17.12/127 + // c= + uint16_t port; + string nettype; + string addrtype; + string address; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "rtcp";} +}; + +class SdpAttrIceUfrag : public SdpItem { +public: + //a=ice-ufrag:sXJ3 + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "ice-ufrag";} +}; + +class SdpAttrIcePwd : public SdpItem { +public: + //a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "ice-pwd";} +}; + +class SdpAttrFingerprint : public SdpItem { +public: + //a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 + string algorithm; + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "fingerprint";} +}; + +class SdpAttrSetup : public SdpItem { +public: + //a=setup:actpass + DtlsRole role{DtlsRole::invalid}; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "setup";} +}; + +class SdpAttrMid : public SdpItem { +public: + //a=mid:audio + string mid; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "mid";} +}; + +class SdpAttrExtmap : public SdpItem { +public: + //a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level + int index; + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "extmap";} +}; + +class SdpAttrRtpMap : public SdpItem { +public: + //a=rtpmap:111 opus/48000/2 + uint8_t pt; + string codec; + int sample_rate; + int channel; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "rtpmap";} +}; + +class SdpAttrRtcpFb : public SdpItem { +public: + //a=rtcp-fb:111 transport-cc + uint8_t pt; + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "rtcp-fb";} +}; + +class SdpAttrFmtp : public SdpItem { +public: + //a=rtcp-fb:111 transport-cc + uint8_t pt; + map values; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "fmtp";} +}; + +class SdpAttrSSRC : public SdpItem { +public: + //a=ssrc:120276603 cname:iSkJ2vn5cYYubTve + uint32_t ssrc; + string key; + string value; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "ssrc";} +}; + +class SdpAttrSctpMap : public SdpItem { +public: + //a=ssrc:120276603 cname:iSkJ2vn5cYYubTve + uint16_t port; + string name; + int mtu; + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "sctpmap";} +}; + +class SdpAttrCandidate : public SdpItem { +public: + //a=candidate:%s 1 udp %u %s %u typ %s + void parse(const string &str) override {SdpItem::parse(str); }; + string toString() const override { return SdpItem::toString(); }; + const char* getKey() override { return "candidate";} +}; + + class RtcMedia { public: - //m= ... - SdpMedia media; - //c= - SdpConnection connection; - //a=: - vector attributes; - + vector items; bool haveAttr(const char *attr) const; string getAttrValue(const char *attr) const; }; class RtcSdp { public: + vector items; + vector medias; + + void parse(const string &str); + string toString() const; + +#if 0 /////Session description（会话级别描述）//// //v= (protocol version) int version; @@ -319,8 +481,7 @@ public: //r=* (zero or more repeat times) //r= string repeat; - //a=* (zero or more media attribute lines) - vector attributes; +#endif }; From 81f29fea61ff29873aeca874fc04c5d45f23b90d Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 28 Mar 2021 17:32:53 +0800 Subject: [PATCH 013/218] =?UTF-8?q?=E5=9F=BA=E6=9C=AC=E5=AE=8C=E6=88=90sdp?= =?UTF-8?q?=E7=9A=84=E8=A7=A3=E6=9E=90=E5=92=8C=E7=94=9F=E6=88=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/Frame.h | 2 +- webrtc/Sdp.cpp | 535 +++++++++++++++++++++++++++++++++++++++++- webrtc/Sdp.h | 338 ++++++++++---------------- 3 files changed, 650 insertions(+), 225 deletions(-) diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index 5f60a68d..afaa115a 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -38,7 +38,7 @@ typedef enum { TrackVideo = 0, TrackAudio, TrackTitle, - TrackData, + TrackApplication, TrackMax = 4 } TrackType; diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 294c84a6..e9d6ec6c 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -3,6 +3,7 @@ // #include "Sdp.h" +#include using onCreateSdpItem = function; static unordered_map sdpItemCreator; @@ -15,10 +16,38 @@ void registerSdpItem(){ return ret; }; Item item; - InfoL << "register sdp item:" << item.getKey(); sdpItemCreator.emplace(item.getKey(), std::move(func)); } +class DirectionInterface { +public: + virtual RtpDirection getDirection() = 0; +}; + +class SdpDirectionSendonly : public SdpItem, public DirectionInterface{ +public: + const char* getKey() override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() override {return RtpDirection::sendonly;} +}; + +class SdpDirectionRecvonly : public SdpItem, public DirectionInterface{ +public: + const char* getKey() override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() override {return RtpDirection::recvonly;} +}; + +class SdpDirectionSendrecv : public SdpItem, public DirectionInterface{ +public: + const char* getKey() override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() override {return RtpDirection::sendrecv;} +}; + +class SdpDirectionInactive : public SdpItem, public DirectionInterface{ +public: + const char* getKey() override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() override {return RtpDirection::inactive;} +}; + static bool registerAllItem(){ registerSdpItem >(); registerSdpItem >(); @@ -49,9 +78,87 @@ static bool registerAllItem(){ registerSdpItem(); registerSdpItem(); registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + return true; } +TrackType getTrackType(const string &str){ + if (str == "video") { + return TrackVideo; + } + if (str == "audio") { + return TrackAudio; + } + if (str == "application") { + return TrackApplication; + } + return TrackInvalid; +} + +const char* getTrackString(TrackType type){ + switch (type) { + case TrackVideo : return "video"; + case TrackAudio : return "audio"; + case TrackApplication : return "application"; + default: return "invalid"; + } +} + +DtlsRole getDtlsRole(const string &str){ + if (str == "active") { + return DtlsRole::active; + } + if (str == "passive") { + return DtlsRole::passive; + } + if (str == "actpass") { + return DtlsRole::actpass; + } + return DtlsRole::invalid; +} + +const char* getDtlsRoleString(DtlsRole role){ + switch (role) { + case DtlsRole::active : return "active"; + case DtlsRole::passive : return "passive"; + case DtlsRole::actpass : return "actpass"; + default: return "invalid"; + } +} + +RtpDirection getRtpDirection(const string &str){ + if (str == "sendonly") { + return RtpDirection::sendonly; + } + if (str == "recvonly") { + return RtpDirection::recvonly; + } + if (str == "sendrecv") { + return RtpDirection::sendrecv; + } + if (str == "inactive") { + return RtpDirection::inactive; + } + return RtpDirection::invalid; +} + +const char* getRtpDirectionString(RtpDirection val){ + switch (val) { + case RtpDirection::sendonly : return "sendonly"; + case RtpDirection::recvonly : return "recvonly"; + case RtpDirection::sendrecv : return "sendrecv"; + case RtpDirection::inactive : return "inactive"; + default: return "invalid"; + } +} + +////////////////////////////////////////////////////////////////////////////////////////// + void RtcSdp::parse(const string &str) { static auto flag = registerAllItem(); RtcMedia *media = nullptr; @@ -85,11 +192,142 @@ void RtcSdp::parse(const string &str) { } string RtcSdp::toString() const { - return std::string(); + _StrPrinter printer; + for (auto &item : items) { + printer << item->getKey() << "=" << item->toString() << "\r\n"; + } + for (auto &media : medias) { + printer << media.toString(); + } + + return std::move(printer); +} + +////////////////////////////////////////////////////////////////////// + +string RtcMedia::toString() const { + _StrPrinter printer; + for (auto &item : items) { + printer << item->getKey() << "=" << item->toString() << "\r\n"; + } + return std::move(printer); +} + +RtpDirection RtcMedia::getDirection() const{ + for (auto &item : items) { + auto attr = dynamic_pointer_cast(item); + if (attr) { + auto dir = dynamic_pointer_cast(attr->detail); + if (dir) { + return dir->getDirection(); + } + } + } + return RtpDirection::invalid; +} + + +////////////////////////////////////////////////////////////////////////////////////////// + +#define SDP_THROW() throw std::invalid_argument(StrPrinter << "解析sdp " << getKey() << " 字段失败:" << str) +void SdpTime::parse(const string &str) { + if (sscanf(str.data(), "%" PRIu64 " %" PRIu64, &start, &stop) != 2) { + SDP_THROW(); + } +} + +string SdpTime::toString() const { + if (value.empty()) { + value = to_string(start) + " " + to_string(stop); + } + return SdpItem::toString(); +} + +void SdpOrigin::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() != 6) { + SDP_THROW(); + } + username = vec[0]; + session_id = vec[1]; + session_version = vec[2]; + nettype = vec[3]; + addrtype = vec[4]; + address = vec[6]; +} + +string SdpOrigin::toString() const { + if (value.empty()) { + value = username + " " + session_id + " " + session_version + " " + nettype + " " + addrtype + " " + address; + } + return SdpItem::toString(); +} + +void SdpConnection::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() != 3) { + SDP_THROW(); + } + nettype = vec[0]; + addrtype = vec[1]; + address = vec[2]; +} + +string SdpConnection::toString() const { + if (value.empty()) { + value = nettype + " " + addrtype + " " + address; + } + return SdpItem::toString(); +} + +void SdpBandwidth::parse(const string &str) { + auto vec = split(str, ":"); + if (vec.size() != 2) { + SDP_THROW(); + } + bwtype = vec[0]; + bandwidth = atoi(vec[1].data()); +} + +string SdpBandwidth::toString() const { + if (value.empty()) { + value = bwtype + ":" + to_string(bandwidth); + } + return SdpItem::toString(); +} + +void SdpMedia::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() < 4) { + SDP_THROW(); + } + type = getTrackType(vec[0]); + if (type == TrackInvalid) { + SDP_THROW(); + } + port = atoi(vec[1].data()); + proto = vec[2]; + for (int i = 3; i < vec.size(); ++i) { + auto pt = atoi(vec[i].data()); + if (type != TrackApplication && pt > 0xFF) { + SDP_THROW(); + } + fmts.emplace_back(pt); + } +} + +string SdpMedia::toString() const { + if (value.empty()) { + value = string(getTrackString(type)) + " " + to_string(port) + " " + proto; + for (auto fmt : fmts) { + value += ' '; + value += to_string(fmt); + } + } + return SdpItem::toString(); } void SdpAttr::parse(const string &str) { - SdpItem::parse(str); auto pos = str.find(':'); auto key = pos == string::npos ? str : str.substr(0, pos); auto value = pos == string::npos ? string() : str.substr(pos + 1); @@ -102,10 +340,290 @@ void SdpAttr::parse(const string &str) { } } +string SdpAttr::toString() const { + if (value.empty()) { + auto detail_value = detail->toString(); + if (detail_value.empty()) { + value = detail->getKey(); + } else { + value = string(detail->getKey()) + ":" + detail_value; + } + } + return SdpItem::toString(); +} + +void SdpAttrGroup::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() < 2) { + SDP_THROW(); + } + type = vec[0]; + vec.erase(vec.begin()); + mids = std::move(vec); +} + +string SdpAttrGroup::toString() const { + if (value.empty()) { + value = type; + for (auto mid : mids) { + value += ' '; + value += mid; + } + } + return SdpItem::toString(); +} + +void SdpAttrMsidSemantic::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() < 1) { + SDP_THROW(); + } + msid = vec[0]; + token = vec.size() > 1 ? vec[1] : ""; +} + +string SdpAttrMsidSemantic::toString() const { + if (value.empty()) { + if (token.empty()) { + value = string(" ") + msid; + } else { + value = string(" ") + msid + " " + token; + } + } + return SdpItem::toString(); +} + +void SdpAttrRtcp::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() != 4) { + SDP_THROW(); + } + port = atoi(vec[0].data()); + nettype = vec[1]; + addrtype = vec[2]; + address = vec[3]; +} + +string SdpAttrRtcp::toString() const { + if (value.empty()) { + value = to_string(port) + " " + nettype + " " + addrtype + " " + address; + } + return SdpItem::toString(); +} + +void SdpAttrFingerprint::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() != 2) { + SDP_THROW(); + } + algorithm = vec[0]; + hash = vec[1]; +} + +string SdpAttrFingerprint::toString() const { + if (value.empty()) { + value = algorithm + " " + hash; + } + return SdpItem::toString(); +} + +void SdpAttrSetup::parse(const string &str) { + role = getDtlsRole(str); + if (role == DtlsRole::invalid) { + SDP_THROW(); + } +} + +string SdpAttrSetup::toString() const { + if (value.empty()) { + value = getDtlsRoleString(role); + } + return SdpItem::toString(); +} + +void SdpAttrExtmap::parse(const string &str) { + char buf[128] = {0}; + if (sscanf(str.data(), "%" PRId32 " %127s", &index, buf) != 2) { + SDP_THROW(); + } + ext = buf; +} + +string SdpAttrExtmap::toString() const { + if (value.empty()) { + value = to_string(index) + " " + ext; + } + return SdpItem::toString(); +} + +void SdpAttrRtpMap::parse(const string &str) { + char buf[32] = {0}; + if (sscanf(str.data(), "%" PRId8 " %31[^/]/%" PRId32 "/%" PRId32, &pt, buf, &sample_rate, &channel) != 4) { + if (sscanf(str.data(), "%" PRId8 " %31[^/]/%" PRId32, &pt, buf, &sample_rate) != 3) { + SDP_THROW(); + } + } + codec = buf; +} + +string SdpAttrRtpMap::toString() const { + if (value.empty()) { + value = to_string(pt) + " " + codec + "/" + to_string(sample_rate); + if (channel) { + value += '/'; + value += to_string(channel); + } + } + return SdpItem::toString(); +} + +void SdpAttrRtcpFb::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() < 2) { + SDP_THROW(); + } + pt = atoi(vec[0].data()); + vec.erase(vec.begin()); + arr = std::move(vec); +} + +string SdpAttrRtcpFb::toString() const { + if (value.empty()) { + value = to_string(pt); + for (auto &item : arr) { + value += ' '; + value += item; + } + } + return SdpItem::toString(); +} + +void SdpAttrFmtp::parse(const string &str) { + auto pos = str.find(' '); + if (pos == string::npos) { + SDP_THROW(); + } + pt = atoi(str.substr(0, pos).data()); + auto vec = split(str.substr(pos + 1), ";"); + for (auto &item : vec) { + trim(item); + auto pr_vec = split(item, "="); + if (pr_vec.size() != 2) { + SDP_THROW(); + } + arr.emplace_back(std::make_pair(pr_vec[0], pr_vec[1])); + } + if (arr.empty()) { + SDP_THROW(); + } +} + +string SdpAttrFmtp::toString() const { + if (value.empty()) { + value = to_string(pt); + int i = 0; + for (auto &pr : arr) { + value += (i++ ? ';' : ' '); + value += pr.first + "=" + pr.second; + } + } + return SdpItem::toString(); +} + +void SdpAttrSSRC::parse(const string &str) { + char attr_buf[32] = {0}; + char attr_val_buf[128] = {0}; + if (3 == sscanf(str.data(), "%" PRIu32 " %31[^:]:%127[^\0]", &ssrc, attr_buf, attr_val_buf)) { + attribute = attr_buf; + attribute_value = attr_val_buf; + } else if (2 == sscanf(str.data(), "%" PRIu32 " %31s", &ssrc, attr_buf)) { + attribute = attr_buf; + } else { + SDP_THROW(); + } +} + +string SdpAttrSSRC::toString() const { + if (value.empty()) { + value = to_string(ssrc) + ' '; + value += attribute; + if (!attribute_value.empty()) { + value += ':'; + value += attribute_value; + } + } + return SdpItem::toString(); +} + +void SdpAttrSctpMap::parse(const string &str) { + char subtypes_buf[64] = {0}; + if (3 == sscanf(str.data(), "%" PRIu16 " %63[^ ] %" PRId32, &port, subtypes_buf, &streams)) { + subtypes = subtypes_buf; + } else { + SDP_THROW(); + } +} + +string SdpAttrSctpMap::toString() const { + if (value.empty()) { + value = to_string(port); + value += ' '; + value += subtypes; + value += ' '; + value += to_string(streams); + } + return SdpItem::toString(); +} + +void SdpAttrCandidate::parse(const string &str) { + char transport_buf[16] = {0}; + char address_buf[32] = {0}; + char type_buf[16] = {0}; + + if (7 != sscanf(str.data(), "%" PRIu32 " %" PRIu32 " %15[^ ] %" PRIu32 " %31[^ ] %" PRIu16 " typ %15[^ \0]", + &foundation, &component, transport_buf, &priority, address_buf, &port, type_buf)) { + SDP_THROW(); + } + transport = transport_buf; + address = address_buf; + type = type_buf; + auto pos = str.find(type); + if (pos != string::npos) { + auto remain = str.substr(pos + type.size()); + trim(remain); + if (!remain.empty()) { + auto vec = split(remain, " "); + string key; + for (auto &item : vec) { + if (key.empty()) { + key = item; + } else { + arr.emplace_back(std::make_pair(std::move(key), std::move(item))); + } + } + } + } +} + +string SdpAttrCandidate::toString() const { + if (value.empty()) { + value = to_string(foundation) + " " + to_string(component) + " " + transport + " " + to_string(priority) + + " " + address + " " + to_string(port) + " typ " + type; + for (auto &pr : arr) { + value += ' '; + value += pr.first; + value += ' '; + value += pr.second; + } + } + return SdpItem::toString(); +} + void test_sdp(){ char str1[] = "v=0\n" "o=- 380154348540553537 2 IN IP4 127.0.0.1\n" "s=-\n" + "b=CT:1900\n" "t=0 0\n" "a=group:BUNDLE video\n" "a=msid-semantic: WMS\n" @@ -212,7 +730,7 @@ void test_sdp(){ "a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" "a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" "a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=candidate:1 1 udp %u %s %u typ host\n" + "a=candidate:3575467457 1 udp 2113937151 10.15.83.23 57857 typ host generation 0 ufrag 6R0z network-cost 999\n" "m=application 9 DTLS/SCTP 5000\n" "c=IN IP4 0.0.0.0\n" "a=ice-ufrag:sXJ3\n" @@ -220,7 +738,8 @@ void test_sdp(){ "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" "a=setup:actpass\n" "a=mid:data\n" - "a=sctpmap:5000 webrtc-datachannel 1024"; + "a=sctpmap:5000 webrtc-datachannel 1024\n" + "a=sctp-port:5000"; RtcSdp sdp1; sdp1.parse(str1); @@ -228,6 +747,12 @@ void test_sdp(){ RtcSdp sdp2; sdp2.parse(str2); + for (auto media : sdp1.medias) { + InfoL << getRtpDirectionString(media.getDirection()); + } + for (auto media : sdp2.medias) { + InfoL << getRtpDirectionString(media.getDirection()); + } InfoL << sdp1.toString(); InfoL << sdp2.toString(); } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index aafdae04..c77e09c4 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -12,107 +12,6 @@ using namespace std; using namespace mediakit; //https://datatracker.ietf.org/doc/rfc4566/?include_text=1 - -//v=0 -//o=- 2584450093346841581 2 IN IP4 127.0.0.1 -//s=- -//t=0 0 -//a=group:BUNDLE audio video data -//a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549 -//m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 -//c=IN IP4 0.0.0.0 -//a=rtcp:9 IN IP4 0.0.0.0 -//a=ice-ufrag:sXJ3 -//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV -//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 -//a=setup:actpass -//a=mid:audio -//a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level -//a=sendrecv -//a=rtcp-mux -//a=rtpmap:111 opus/48000/2 -//a=rtcp-fb:111 transport-cc -//a=fmtp:111 minptime=10;useinbandfec=1 -//a=rtpmap:103 ISAC/16000 -//a=rtpmap:104 ISAC/32000 -//a=rtpmap:9 G722/8000 -//a=rtpmap:0 PCMU/8000 -//a=rtpmap:8 PCMA/8000 -//a=rtpmap:106 CN/32000 -//a=rtpmap:105 CN/16000 -//a=rtpmap:13 CN/8000 -//a=rtpmap:110 telephone-event/48000 -//a=rtpmap:112 telephone-event/32000 -//a=rtpmap:113 telephone-event/16000 -//a=rtpmap:126 telephone-event/8000 -//a=ssrc:120276603 cname:iSkJ2vn5cYYubTve -//a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363 -//a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 -//a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363 -//m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125 -//c=IN IP4 0.0.0.0 -//a=rtcp:9 IN IP4 0.0.0.0 -//a=ice-ufrag:sXJ3 -//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV -//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 -//a=setup:actpass -//a=mid:video -//a=extmap:2 urn:ietf:params:rtp-hdrext:toffset -//a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time -//a=extmap:4 urn:3gpp:video-orientation -//a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 -//a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay -//a=sendrecv -//a=rtcp-mux -//a=rtcp-rsize -//a=rtpmap:96 VP8/90000 -//a=rtcp-fb:96 ccm fir -//a=rtcp-fb:96 nack -//a=rtcp-fb:96 nack pli -//a=rtcp-fb:96 goog-remb -//a=rtcp-fb:96 transport-cc -//a=rtpmap:98 VP9/90000 -//a=rtcp-fb:98 ccm fir -//a=rtcp-fb:98 nack -//a=rtcp-fb:98 nack pli -//a=rtcp-fb:98 goog-remb -//a=rtcp-fb:98 transport-cc -//a=rtpmap:100 H264/90000 -//a=rtcp-fb:100 ccm fir -//a=rtcp-fb:100 nack -//a=rtcp-fb:100 nack pli -//a=rtcp-fb:100 goog-remb -//a=rtcp-fb:100 transport-cc -//a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f -//a=rtpmap:102 red/90000 -//a=rtpmap:127 ulpfec/90000 -//a=rtpmap:97 rtx/90000 -//a=fmtp:97 apt=96 -//a=rtpmap:99 rtx/90000 -//a=fmtp:99 apt=98 -//a=rtpmap:101 rtx/90000 -//a=fmtp:101 apt=100 -//a=rtpmap:125 rtx/90000 -//a=fmtp:125 apt=102 -//a=ssrc-group:FID 2580761338 611523443 -//a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve -//a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961 -//a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 -//a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961 -//a=ssrc:611523443 cname:iSkJ2vn5cYYubTve -//a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961 -//a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549 -//a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961 -//a=candidate:1 1 udp %u %s %u typ host -//m=application 9 DTLS/SCTP 5000 -//c=IN IP4 0.0.0.0 -//a=ice-ufrag:sXJ3 -//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV -//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 -//a=setup:actpass -//a=mid:data -//a=sctpmap:5000 webrtc-datachannel 1024 - //Session description // v= (protocol version) // o= (originator and session identifier) @@ -148,7 +47,7 @@ enum class RtpDirection { //只发送 sendonly, //只接收 - revonly, + recvonly, //同时发送接收 sendrecv, //禁止发送数据 @@ -171,10 +70,16 @@ enum class SdpType { answer }; +TrackType getTrackType(const string &str); +const char* getTrackString(TrackType type); +DtlsRole getDtlsRole(const string &str); +const char* getDtlsRoleString(DtlsRole role); +RtpDirection getRtpDirection(const string &str); +const char* getRtpDirectionString(RtpDirection val); + class SdpItem { public: using Ptr = std::shared_ptr; - string value; virtual ~SdpItem() = default; virtual void parse(const string &str) { value = str; @@ -183,6 +88,9 @@ public: return value; } virtual const char* getKey() = 0; + +protected: + mutable string value; }; template @@ -203,10 +111,10 @@ class SdpTime : public SdpItem{ public: //5.9. Timing ("t=") // t= - uint64_t start; - uint64_t stop; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + uint64_t start {0}; + uint64_t stop {0}; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "t";} }; @@ -215,14 +123,14 @@ public: // 5.2. Origin ("o=") // o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5 // o= - string username; + string username {"-"}; string session_id; string session_version; - string nettype; - string addrtype; - string address; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string nettype {"IN"}; + string addrtype {"IP4"}; + string address {"0.0.0.0"}; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "o";} }; @@ -231,11 +139,11 @@ public: // 5.7. Connection Data ("c=") // c=IN IP4 224.2.17.12/127 // c= - string nettype; - string addrtype; - string address; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string nettype {"IN"}; + string addrtype {"IP4"}; + string address {"0.0.0.0"}; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "c";} }; @@ -245,11 +153,11 @@ public: //b=: //AS、CT - string bwtype; - int bandwidth; + string bwtype {"AS"}; + uint32_t bandwidth {0}; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "b";} }; @@ -259,11 +167,11 @@ public: // m= ... TrackType type; uint16_t port; - vector proto; - vector fmt; + string proto; + vector fmts; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "m";} }; @@ -275,7 +183,7 @@ public: //a=: SdpItem::Ptr detail; void parse(const string &str) override; - string toString() const override { return SdpItem::toString(); }; + string toString() const override; const char* getKey() override { return "a";} }; @@ -285,50 +193,63 @@ public: // BUNDLE group is included at the session-level. //a=group:LS session level attribute MUST be included wth the 'mid' // identifiers that are part of the same lip sync group. - string type; - vector mid; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string type {"BUNDLE"}; + vector mids; + void parse(const string &str) override ; + string toString() const override ; const char* getKey() override { return "group";} }; class SdpAttrMsidSemantic : public SdpItem { public: - string name{"WMS"}; - string mid; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + //https://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02#section-3 + //3. The Msid-Semantic Attribute + // + // In order to fully reproduce the semantics of the SDP and SSRC + // grouping frameworks, a session-level attribute is defined for + // signalling the semantics associated with an msid grouping. + // + // This OPTIONAL attribute gives the message ID and its group semantic. + // a=msid-semantic: examplefoo LS + // + // + // The ABNF of msid-semantic is: + // + // msid-semantic-attr = "msid-semantic:" " " msid token + // token = + // + // The semantic field may hold values from the IANA registries + // "Semantics for the "ssrc-group" SDP Attribute" and "Semantics for the + // "group" SDP Attribute". + //a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549 + string msid{"WMS"}; + string token; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "msid-semantic";} }; class SdpAttrRtcp : public SdpItem { public: - // c=IN IP4 224.2.17.12/127 - // c= + // a=rtcp:9 IN IP4 0.0.0.0 uint16_t port; - string nettype; - string addrtype; - string address; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string nettype {"IN"}; + string addrtype {"IP4"}; + string address {"0.0.0.0"}; + void parse(const string &str) override;; + string toString() const override; const char* getKey() override { return "rtcp";} }; class SdpAttrIceUfrag : public SdpItem { public: //a=ice-ufrag:sXJ3 - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "ice-ufrag";} }; class SdpAttrIcePwd : public SdpItem { public: //a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "ice-pwd";} }; @@ -336,27 +257,24 @@ class SdpAttrFingerprint : public SdpItem { public: //a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79 string algorithm; - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string hash; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "fingerprint";} }; class SdpAttrSetup : public SdpItem { public: //a=setup:actpass - DtlsRole role{DtlsRole::invalid}; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + DtlsRole role{DtlsRole::actpass}; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "setup";} }; class SdpAttrMid : public SdpItem { public: //a=mid:audio - string mid; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; const char* getKey() override { return "mid";} }; @@ -364,9 +282,9 @@ class SdpAttrExtmap : public SdpItem { public: //a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level int index; - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string ext; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "extmap";} }; @@ -376,68 +294,85 @@ public: uint8_t pt; string codec; int sample_rate; - int channel; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + int channel {0}; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "rtpmap";} }; class SdpAttrRtcpFb : public SdpItem { public: - //a=rtcp-fb:111 transport-cc + //a=rtcp-fb:98 nack pli uint8_t pt; - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + vector arr; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "rtcp-fb";} }; class SdpAttrFmtp : public SdpItem { public: - //a=rtcp-fb:111 transport-cc + //fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f uint8_t pt; - map values; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + vector > arr; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "fmtp";} }; class SdpAttrSSRC : public SdpItem { public: //a=ssrc:120276603 cname:iSkJ2vn5cYYubTve + //a=ssrc: + //a=ssrc: : uint32_t ssrc; - string key; - string value; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string attribute; + string attribute_value; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "ssrc";} }; class SdpAttrSctpMap : public SdpItem { public: - //a=ssrc:120276603 cname:iSkJ2vn5cYYubTve + //https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-05 + //a=sctpmap:5000 webrtc-datachannel 1024 + //a=sctpmap: sctpmap-number media-subtypes [streams] uint16_t port; - string name; - int mtu; - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + string subtypes; + int streams; + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "sctpmap";} }; class SdpAttrCandidate : public SdpItem { public: - //a=candidate:%s 1 udp %u %s %u typ %s - void parse(const string &str) override {SdpItem::parse(str); }; - string toString() const override { return SdpItem::toString(); }; + //https://tools.ietf.org/html/rfc5245 + //15.1. "candidate" Attribute + //a=candidate:4 1 udp 2 192.168.1.7 58107 typ host + //a=candidate:
typ + uint32_t foundation; + uint32_t component; + string transport {"udp"}; + uint32_t priority; + string address; + uint16_t port; + string type; + vector > arr; + + void parse(const string &str) override; + string toString() const override; const char* getKey() override { return "candidate";} }; - class RtcMedia { public: vector items; + string toString() const; bool haveAttr(const char *attr) const; string getAttrValue(const char *attr) const; + RtpDirection getDirection() const; }; class RtcSdp { @@ -447,41 +382,6 @@ public: void parse(const string &str); string toString() const; - -#if 0 - /////Session description（会话级别描述）//// - //v= (protocol version) - int version; - //o= (session origin information ) - SdpOrigin origin; - //s= (session name) - string session_name; - //t= (time the session is active) - SdpTime time; - - //// 非必须 //// - //i=* (session information) - string information; - //u=* (URI of description) - string url; - //e=* (email address) - string email; - //p=* (phone number) - string phone; - //c=* (connection information -- not required if included in all media) - SdpConnection connection; - //b=* (zero or more bandwidth information lines) - SdpBandwidth bandwidth; - //z=* (time zone adjustments) - //z= .... - string time_zone; - //k=* (encryption key) - //k=, k=: - string crypt_key; - //r=* (zero or more repeat times) - //r= - string repeat; -#endif }; From 6fa385f3a1add95b92afcd0bd665412ef6a0c859 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 28 Mar 2021 23:31:21 +0800 Subject: [PATCH 014/218] =?UTF-8?q?sdp=20=E6=94=AF=E6=8C=81ssrc-group?= =?UTF-8?q?=E5=B1=9E=E6=80=A7?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 57 +++++++++++++++++++++++++++++------ webrtc/Sdp.h | 81 +++++++++++++++++++++++++++++++++++--------------- 2 files changed, 105 insertions(+), 33 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index e9d6ec6c..e122eafa 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -21,31 +21,31 @@ void registerSdpItem(){ class DirectionInterface { public: - virtual RtpDirection getDirection() = 0; + virtual RtpDirection getDirection() const = 0; }; class SdpDirectionSendonly : public SdpItem, public DirectionInterface{ public: - const char* getKey() override { return getRtpDirectionString(getDirection());} - RtpDirection getDirection() override {return RtpDirection::sendonly;} + const char* getKey() const override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() const override {return RtpDirection::sendonly;} }; class SdpDirectionRecvonly : public SdpItem, public DirectionInterface{ public: - const char* getKey() override { return getRtpDirectionString(getDirection());} - RtpDirection getDirection() override {return RtpDirection::recvonly;} + const char* getKey() const override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() const override {return RtpDirection::recvonly;} }; class SdpDirectionSendrecv : public SdpItem, public DirectionInterface{ public: - const char* getKey() override { return getRtpDirectionString(getDirection());} - RtpDirection getDirection() override {return RtpDirection::sendrecv;} + const char* getKey() const override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() const override {return RtpDirection::sendrecv;} }; class SdpDirectionInactive : public SdpItem, public DirectionInterface{ public: - const char* getKey() override { return getRtpDirectionString(getDirection());} - RtpDirection getDirection() override {return RtpDirection::inactive;} + const char* getKey() const override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() const override {return RtpDirection::inactive;} }; static bool registerAllItem(){ @@ -69,6 +69,7 @@ static bool registerAllItem(){ registerSdpItem(); registerSdpItem(); registerSdpItem(); + registerSdpItem(); registerSdpItem(); registerSdpItem(); registerSdpItem(); @@ -77,6 +78,7 @@ static bool registerAllItem(){ registerSdpItem(); registerSdpItem(); registerSdpItem(); + registerSdpItem(); registerSdpItem(); registerSdpItem(); registerSdpItem(); @@ -230,6 +232,8 @@ RtpDirection RtcMedia::getDirection() const{ ////////////////////////////////////////////////////////////////////////////////////////// #define SDP_THROW() throw std::invalid_argument(StrPrinter << "解析sdp " << getKey() << " 字段失败:" << str) +#define SDP_THROW2() throw std::invalid_argument(StrPrinter << "生成sdp " << getKey() << " 字段失败") + void SdpTime::parse(const string &str) { if (sscanf(str.data(), "%" PRIu64 " %" PRIu64, &start, &stop) != 2) { SDP_THROW(); @@ -555,6 +559,41 @@ string SdpAttrSSRC::toString() const { return SdpItem::toString(); } +void SdpAttrSSRCGroup::parse(const string &str) { + auto vec = split(str, " "); + if (vec.size() == 3) { + if (vec[0] != "FID") { + SDP_THROW(); + } + type = std::move(vec[0]); + u.fid.rtp_ssrc = atoi(vec[1].data()); + u.fid.rtx_ssrc = atoi(vec[2].data()); + } else if (vec.size() == 4) { + if (vec[0] != "SIM") { + SDP_THROW(); + } + type = std::move(vec[0]); + u.sim.rtp_ssrc_low = atoi(vec[1].data()); + u.sim.rtp_ssrc_mid = atoi(vec[2].data()); + u.sim.rtp_ssrc_high = atoi(vec[3].data()); + } else { + SDP_THROW(); + } +} + +string SdpAttrSSRCGroup::toString() const { + if (value.empty()) { + if (type == "FID") { + value = type + " " + to_string(u.fid.rtp_ssrc) + " " + to_string(u.fid.rtx_ssrc); + } else if (type == "SIM") { + value = type + " " + to_string(u.sim.rtp_ssrc_low) + " " + to_string(u.sim.rtp_ssrc_mid) + " " + to_string(u.sim.rtp_ssrc_high); + } else { + SDP_THROW2(); + } + } + return SdpItem::toString(); +} + void SdpAttrSctpMap::parse(const string &str) { char subtypes_buf[64] = {0}; if (3 == sscanf(str.data(), "%" PRIu16 " %63[^ ] %" PRId32, &port, subtypes_buf, &streams)) { diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index c77e09c4..e10ad091 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -87,7 +87,7 @@ public: virtual string toString() const { return value; } - virtual const char* getKey() = 0; + virtual const char* getKey() const = 0; protected: mutable string value; @@ -97,14 +97,14 @@ template class SdpString : public SdpItem{ public: // *=* - const char* getKey() override { static string key(1, KEY); return key.data();} + const char* getKey() const override { static string key(1, KEY); return key.data();} }; class SdpCommon : public SdpItem { public: string key; SdpCommon(string key) { this->key = std::move(key); } - const char* getKey() override { return key.data();} + const char* getKey() const override { return key.data();} }; class SdpTime : public SdpItem{ @@ -115,7 +115,7 @@ public: uint64_t stop {0}; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "t";} + const char* getKey() const override { return "t";} }; class SdpOrigin : public SdpItem{ @@ -131,7 +131,7 @@ public: string address {"0.0.0.0"}; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "o";} + const char* getKey() const override { return "o";} }; class SdpConnection : public SdpItem { @@ -144,7 +144,7 @@ public: string address {"0.0.0.0"}; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "c";} + const char* getKey() const override { return "c";} }; class SdpBandwidth : public SdpItem { @@ -158,7 +158,7 @@ public: void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "b";} + const char* getKey() const override { return "b";} }; class SdpMedia : public SdpItem { @@ -172,7 +172,7 @@ public: void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "m";} + const char* getKey() const override { return "m";} }; class SdpAttr : public SdpItem{ @@ -184,7 +184,7 @@ public: SdpItem::Ptr detail; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "a";} + const char* getKey() const override { return "a";} }; class SdpAttrGroup : public SdpItem{ @@ -197,7 +197,7 @@ public: vector mids; void parse(const string &str) override ; string toString() const override ; - const char* getKey() override { return "group";} + const char* getKey() const override { return "group";} }; class SdpAttrMsidSemantic : public SdpItem { @@ -226,7 +226,7 @@ public: string token; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "msid-semantic";} + const char* getKey() const override { return "msid-semantic";} }; class SdpAttrRtcp : public SdpItem { @@ -238,19 +238,25 @@ public: string address {"0.0.0.0"}; void parse(const string &str) override;; string toString() const override; - const char* getKey() override { return "rtcp";} + const char* getKey() const override { return "rtcp";} }; class SdpAttrIceUfrag : public SdpItem { public: //a=ice-ufrag:sXJ3 - const char* getKey() override { return "ice-ufrag";} + const char* getKey() const override { return "ice-ufrag";} }; class SdpAttrIcePwd : public SdpItem { public: //a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV - const char* getKey() override { return "ice-pwd";} + const char* getKey() const override { return "ice-pwd";} +}; + +class SdpAttrIceOption : public SdpItem { +public: + //a=ice-options:trickle + const char* getKey() const override { return "ice-options";} }; class SdpAttrFingerprint : public SdpItem { @@ -260,7 +266,7 @@ public: string hash; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "fingerprint";} + const char* getKey() const override { return "fingerprint";} }; class SdpAttrSetup : public SdpItem { @@ -269,13 +275,13 @@ public: DtlsRole role{DtlsRole::actpass}; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "setup";} + const char* getKey() const override { return "setup";} }; class SdpAttrMid : public SdpItem { public: //a=mid:audio - const char* getKey() override { return "mid";} + const char* getKey() const override { return "mid";} }; class SdpAttrExtmap : public SdpItem { @@ -285,7 +291,7 @@ public: string ext; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "extmap";} + const char* getKey() const override { return "extmap";} }; class SdpAttrRtpMap : public SdpItem { @@ -297,7 +303,7 @@ public: int channel {0}; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "rtpmap";} + const char* getKey() const override { return "rtpmap";} }; class SdpAttrRtcpFb : public SdpItem { @@ -307,7 +313,7 @@ public: vector arr; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "rtcp-fb";} + const char* getKey() const override { return "rtcp-fb";} }; class SdpAttrFmtp : public SdpItem { @@ -317,7 +323,7 @@ public: vector > arr; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "fmtp";} + const char* getKey() const override { return "fmtp";} }; class SdpAttrSSRC : public SdpItem { @@ -330,7 +336,34 @@ public: string attribute_value; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "ssrc";} + const char* getKey() const override { return "ssrc";} +}; + +class SdpAttrSSRCGroup : public SdpItem { +public: + //a=ssrc-group 定义参考 RFC 5576 ，用于描述多个 ssrc 之间的关联，常见的有两种： + + //a=ssrc-group:FID 2430709021 3715850271 + // FID (Flow Identification) 最初用在 FEC 的关联中，WebRTC 中通常用于关联一组常规 RTP stream 和 重传 RTP stream 。 + + //a=ssrc-group:SIM 360918977 360918978 360918980 + // 在 Chrome 独有的 SDP munging 风格的 simulcast 中使用，将三组编码质量由低到高的 MediaStreamTrack 关联在一起。 + string type{"FID"}; + union { + struct { + uint32_t rtp_ssrc; + uint32_t rtx_ssrc; + } fid; + struct { + uint32_t rtp_ssrc_low; + uint32_t rtp_ssrc_mid; + uint32_t rtp_ssrc_high; + } sim; + } u; + + void parse(const string &str) override; + string toString() const override; + const char* getKey() const override { return "ssrc-group";} }; class SdpAttrSctpMap : public SdpItem { @@ -343,7 +376,7 @@ public: int streams; void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "sctpmap";} + const char* getKey() const override { return "sctpmap";} }; class SdpAttrCandidate : public SdpItem { @@ -363,7 +396,7 @@ public: void parse(const string &str) override; string toString() const override; - const char* getKey() override { return "candidate";} + const char* getKey() const override { return "candidate";} }; class RtcMedia { From 7722564ad8af177e7c18c9512374185ccc7c9f13 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 28 Mar 2021 23:36:11 +0800 Subject: [PATCH 015/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E6=B3=A8=E9=87=8A?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index e10ad091..5afd294e 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -167,6 +167,10 @@ public: // m= ... TrackType type; uint16_t port; + //RTP/AVP：应用场景为视频/音频的 RTP 协议。参考 RFC 3551 + //RTP/SAVP：应用场景为视频/音频的 SRTP 协议。参考 RFC 3711 + //RTP/AVPF: 应用场景为视频/音频的 RTP 协议，支持 RTCP-based Feedback。参考 RFC 4585 + //RTP/SAVPF: 应用场景为视频/音频的 SRTP 协议，支持 RTCP-based Feedback。参考 RFC 5124 string proto; vector fmts; From e670099192f964d6857baa15e9a850b3efe3174b Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 28 Mar 2021 23:52:26 +0800 Subject: [PATCH 016/218] =?UTF-8?q?=E5=AE=8C=E5=96=84sdp=E7=9B=B8=E5=85=B3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.h | 27 ++++++++++++++++++++++----- webrtc/WebRtcTransport.cpp | 1 + 2 files changed, 23 insertions(+), 5 deletions(-) diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 5afd294e..8af83f04 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -12,6 +12,8 @@ using namespace std; using namespace mediakit; //https://datatracker.ietf.org/doc/rfc4566/?include_text=1 +//https://blog.csdn.net/aggresss/article/details/109850434 +//https://aggresss.blog.csdn.net/article/details/106436703 //Session description // v= (protocol version) // o= (originator and session identifier) @@ -290,6 +292,7 @@ public: class SdpAttrExtmap : public SdpItem { public: + //https://aggresss.blog.csdn.net/article/details/106436703 //a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level int index; string ext; @@ -312,7 +315,12 @@ public: class SdpAttrRtcpFb : public SdpItem { public: - //a=rtcp-fb:98 nack pli + //a=rtcp-fb:98 nack pli + //a=rtcp-fb:120 nack 支持 nack 重传，nack (Negative-Acknowledgment) 。 + //a=rtcp-fb:120 nack pli 支持 nack 关键帧重传，PLI (Picture Loss Indication) 。 + //a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli 是用于重传时的关键帧请求。 + //a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 + //a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。 uint8_t pt; vector arr; void parse(const string &str) override; @@ -332,9 +340,20 @@ public: class SdpAttrSSRC : public SdpItem { public: - //a=ssrc:120276603 cname:iSkJ2vn5cYYubTve + //a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7 + //a=ssrc:3245185839 msid:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9 0cf7e597-36a2-4480-9796-69bf0955eef5 + //a=ssrc:3245185839 mslabel:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9 + //a=ssrc:3245185839 label:0cf7e597-36a2-4480-9796-69bf0955eef5 //a=ssrc: //a=ssrc: : + //cname 是必须的，msid/mslabel/label 这三个属性都是 WebRTC 自创的，或者说 Google 自创的，可以参考 https://tools.ietf.org/html/draft-ietf-mmusic-msid-17， + // 理解它们三者的关系需要先了解三个概念：RTP stream / MediaStreamTrack / MediaStream ： + //一个 a=ssrc 代表一个 RTP stream ； + //一个 MediaStreamTrack 通常包含一个或多个 RTP stream，例如一个视频 MediaStreamTrack 中通常包含两个 RTP stream，一个用于常规传输，一个用于 nack 重传； + //一个 MediaStream 通常包含一个或多个 MediaStreamTrack ，例如 simulcast 场景下，一个 MediaStream 通常会包含三个不同编码质量的 MediaStreamTrack ； + //这种标记方式并不被 Firefox 认可，在 Firefox 生成的 SDP 中一个 a=ssrc 通常只有一行，例如： + //a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7 + uint32_t ssrc; string attribute; string attribute_value; @@ -345,11 +364,9 @@ public: class SdpAttrSSRCGroup : public SdpItem { public: - //a=ssrc-group 定义参考 RFC 5576 ，用于描述多个 ssrc 之间的关联，常见的有两种： - + //a=ssrc-group 定义参考 RFC 5576(https://tools.ietf.org/html/rfc5576) ，用于描述多个 ssrc 之间的关联，常见的有两种： //a=ssrc-group:FID 2430709021 3715850271 // FID (Flow Identification) 最初用在 FEC 的关联中，WebRTC 中通常用于关联一组常规 RTP stream 和 重传 RTP stream 。 - //a=ssrc-group:SIM 360918977 360918978 360918980 // 在 Chrome 独有的 SDP munging 风格的 simulcast 中使用，将三组编码质量由低到高的 MediaStreamTrack 关联在一起。 string type{"FID"}; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 5754f34f..c32e6850 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -93,6 +93,7 @@ std::string WebRtcTransport::GetLocalSdp() { "a=mid:video\r\n" "a=sendonly\r\n" "a=rtcp-mux\r\n" + "a=ice-lite\r\n" "a=ice-ufrag:%s\r\n" "a=ice-pwd:%s\r\n" "a=ice-options:trickle\r\n" From 1228d07ff3d6666e4a469726968f757d807f0e23 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 29 Mar 2021 00:12:20 +0800 Subject: [PATCH 017/218] =?UTF-8?q?extmap=E6=94=AF=E6=8C=81=E6=96=B9?= =?UTF-8?q?=E5=90=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 17 +++++++++++++---- webrtc/Sdp.h | 3 ++- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index e122eafa..e0888dfd 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -447,15 +447,24 @@ string SdpAttrSetup::toString() const { void SdpAttrExtmap::parse(const string &str) { char buf[128] = {0}; - if (sscanf(str.data(), "%" PRId32 " %127s", &index, buf) != 2) { - SDP_THROW(); + char direction_buf[32] = {0}; + if (sscanf(str.data(), "%" PRId32 "/%31[^ ] %127s", &index, direction_buf, buf) != 3) { + if (sscanf(str.data(), "%" PRId32 " %127s", &index, buf) != 2) { + SDP_THROW(); + } + } else { + direction = getRtpDirection(direction_buf); } ext = buf; } string SdpAttrExtmap::toString() const { if (value.empty()) { - value = to_string(index) + " " + ext; + if(direction == RtpDirection::invalid){ + value = to_string(index) + " " + ext; + } else { + value = to_string(index) + "/" + getRtpDirectionString(direction) + " " + ext; + } } return SdpItem::toString(); } @@ -693,7 +702,7 @@ void test_sdp(){ "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" "a=setup:actpass\n" "a=mid:audio\n" - "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" + "a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" "a=sendrecv\n" "a=rtcp-mux\n" "a=rtpmap:111 opus/48000/2\n" diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 8af83f04..9c0d2740 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -293,8 +293,9 @@ public: class SdpAttrExtmap : public SdpItem { public: //https://aggresss.blog.csdn.net/article/details/106436703 - //a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level + //a=extmap:1[/sendonly] urn:ietf:params:rtp-hdrext:ssrc-audio-level int index; + RtpDirection direction{RtpDirection::invalid}; string ext; void parse(const string &str) override; string toString() const override; From 7726caa7f28fd615d64eb195dd6d26356ea39fb3 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 09:53:54 +0800 Subject: [PATCH 018/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dbug=E4=B8=8E=E7=BC=96?= =?UTF-8?q?=E8=AF=91=E8=AD=A6=E5=91=8A?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index e0888dfd..38875bcc 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -235,7 +235,7 @@ RtpDirection RtcMedia::getDirection() const{ #define SDP_THROW2() throw std::invalid_argument(StrPrinter << "生成sdp " << getKey() << " 字段失败") void SdpTime::parse(const string &str) { - if (sscanf(str.data(), "%" PRIu64 " %" PRIu64, &start, &stop) != 2) { + if (sscanf(str.data(), "%" SCNu64 " %" SCNu64, &start, &stop) != 2) { SDP_THROW(); } } @@ -257,7 +257,7 @@ void SdpOrigin::parse(const string &str) { session_version = vec[2]; nettype = vec[3]; addrtype = vec[4]; - address = vec[6]; + address = vec[5]; } string SdpOrigin::toString() const { @@ -311,7 +311,7 @@ void SdpMedia::parse(const string &str) { } port = atoi(vec[1].data()); proto = vec[2]; - for (int i = 3; i < vec.size(); ++i) { + for (size_t i = 3; i < vec.size(); ++i) { auto pt = atoi(vec[i].data()); if (type != TrackApplication && pt > 0xFF) { SDP_THROW(); @@ -448,8 +448,8 @@ string SdpAttrSetup::toString() const { void SdpAttrExtmap::parse(const string &str) { char buf[128] = {0}; char direction_buf[32] = {0}; - if (sscanf(str.data(), "%" PRId32 "/%31[^ ] %127s", &index, direction_buf, buf) != 3) { - if (sscanf(str.data(), "%" PRId32 " %127s", &index, buf) != 2) { + if (sscanf(str.data(), "%" SCNd32 "/%31[^ ] %127s", &index, direction_buf, buf) != 3) { + if (sscanf(str.data(), "%" SCNd32 " %127s", &index, buf) != 2) { SDP_THROW(); } } else { @@ -471,8 +471,8 @@ string SdpAttrExtmap::toString() const { void SdpAttrRtpMap::parse(const string &str) { char buf[32] = {0}; - if (sscanf(str.data(), "%" PRId8 " %31[^/]/%" PRId32 "/%" PRId32, &pt, buf, &sample_rate, &channel) != 4) { - if (sscanf(str.data(), "%" PRId8 " %31[^/]/%" PRId32, &pt, buf, &sample_rate) != 3) { + if (sscanf(str.data(), "%" SCNu8 " %31[^/]/%" SCNd32 "/%" SCNd32, &pt, buf, &sample_rate, &channel) != 4) { + if (sscanf(str.data(), "%" SCNu8 " %31[^/]/%" SCNd32, &pt, buf, &sample_rate) != 3) { SDP_THROW(); } } @@ -543,13 +543,14 @@ string SdpAttrFmtp::toString() const { return SdpItem::toString(); } -void SdpAttrSSRC::parse(const string &str) { +void SdpAttrSSRC::parse(const string &str_in) { + auto str = str_in + '\n'; char attr_buf[32] = {0}; char attr_val_buf[128] = {0}; - if (3 == sscanf(str.data(), "%" PRIu32 " %31[^:]:%127[^\0]", &ssrc, attr_buf, attr_val_buf)) { + if (3 == sscanf(str.data(), "%" SCNu32 " %31[^:]:%127[^\n]", &ssrc, attr_buf, attr_val_buf)) { attribute = attr_buf; attribute_value = attr_val_buf; - } else if (2 == sscanf(str.data(), "%" PRIu32 " %31s", &ssrc, attr_buf)) { + } else if (2 == sscanf(str.data(), "%" SCNu32 " %31s[^\n]", &ssrc, attr_buf)) { attribute = attr_buf; } else { SDP_THROW(); @@ -605,7 +606,7 @@ string SdpAttrSSRCGroup::toString() const { void SdpAttrSctpMap::parse(const string &str) { char subtypes_buf[64] = {0}; - if (3 == sscanf(str.data(), "%" PRIu16 " %63[^ ] %" PRId32, &port, subtypes_buf, &streams)) { + if (3 == sscanf(str.data(), "%" SCNu16 " %63[^ ] %" SCNd32, &port, subtypes_buf, &streams)) { subtypes = subtypes_buf; } else { SDP_THROW(); @@ -628,7 +629,7 @@ void SdpAttrCandidate::parse(const string &str) { char address_buf[32] = {0}; char type_buf[16] = {0}; - if (7 != sscanf(str.data(), "%" PRIu32 " %" PRIu32 " %15[^ ] %" PRIu32 " %31[^ ] %" PRIu16 " typ %15[^ \0]", + if (7 != sscanf(str.data(), "%" SCNu32 " %" SCNu32 " %15[^ ] %" SCNu32 " %31[^ ] %" SCNu16 " typ %15[^ ]", &foundation, &component, transport_buf, &priority, address_buf, &port, type_buf)) { SDP_THROW(); } From 2643269581a1ce7616e0add9598d31cba138d13d Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 10:54:18 +0800 Subject: [PATCH 019/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E9=83=A8=E5=88=86?= =?UTF-8?q?=E5=AF=B9=E8=B1=A1=E5=AE=9A=E4=B9=89?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 16 +++++------ webrtc/Sdp.h | 74 ++++++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 79 insertions(+), 11 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 38875bcc..dae5d776 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -161,9 +161,9 @@ const char* getRtpDirectionString(RtpDirection val){ ////////////////////////////////////////////////////////////////////////////////////////// -void RtcSdp::parse(const string &str) { +void RtcSessionSdp::parse(const string &str) { static auto flag = registerAllItem(); - RtcMedia *media = nullptr; + RtcMediaSdp *media = nullptr; auto lines = split(str, "\n"); for(auto &line : lines){ trim(line); @@ -173,7 +173,7 @@ void RtcSdp::parse(const string &str) { auto key = line.substr(0, 1); auto value = line.substr(2); if (key == "m") { - medias.emplace_back(RtcMedia()); + medias.emplace_back(RtcMediaSdp()); media = &medias.back(); } @@ -193,7 +193,7 @@ void RtcSdp::parse(const string &str) { } } -string RtcSdp::toString() const { +string RtcSessionSdp::toString() const { _StrPrinter printer; for (auto &item : items) { printer << item->getKey() << "=" << item->toString() << "\r\n"; @@ -207,7 +207,7 @@ string RtcSdp::toString() const { ////////////////////////////////////////////////////////////////////// -string RtcMedia::toString() const { +string RtcMediaSdp::toString() const { _StrPrinter printer; for (auto &item : items) { printer << item->getKey() << "=" << item->toString() << "\r\n"; @@ -215,7 +215,7 @@ string RtcMedia::toString() const { return std::move(printer); } -RtpDirection RtcMedia::getDirection() const{ +RtpDirection RtcMediaSdp::getDirection() const{ for (auto &item : items) { auto attr = dynamic_pointer_cast(item); if (attr) { @@ -790,10 +790,10 @@ void test_sdp(){ "a=sctpmap:5000 webrtc-datachannel 1024\n" "a=sctp-port:5000"; - RtcSdp sdp1; + RtcSessionSdp sdp1; sdp1.parse(str1); - RtcSdp sdp2; + RtcSessionSdp sdp2; sdp2.parse(str2); for (auto media : sdp1.medias) { diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 9c0d2740..325c11c0 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -421,7 +421,7 @@ public: const char* getKey() const override { return "candidate";} }; -class RtcMedia { +class RtcMediaSdp { public: vector items; string toString() const; @@ -430,15 +430,83 @@ public: RtpDirection getDirection() const; }; -class RtcSdp { +class RtcSessionSdp { public: vector items; - vector medias; + vector medias; void parse(const string &str); string toString() const; }; +////////////////////////////////////////////////////////////////// + +//ssrc类型 +enum class RtcSSRCType { + rtp = 0, + rtx, + sim_low, + sim_mid, + ssrc_high +}; + +//ssrc相关信息 +class RtcSSRC{ +public: + RtcSSRCType type; + string cname; + string msid; + string mslabel; + string label; +}; + +//rtc传输编码方案 +class RtcPlan{ +public: + uint8_t pt; + string codec; + uint32_t sample_rate; + //音频时有效 + uint32_t channel = 0; + vector > fmtp; + vector rtcp_fb; +}; + +//rtc 媒体描述 +class RtcCodec{ +public: + TrackType type; + string mid; + uint16_t port; + string proto; + + //////// rtp //////// + vector plan; + SdpConnection rtp_addr; + RtpDirection direction; + RtcSSRC ssrc; + + //////// rtx - rtcp //////// + bool rtcp_mux; + bool rtcp_rsize; + uint32_t rtx_ssrc; + SdpAttrRtcp rtcp_addr; + + //////// ice //////// + bool ice_trickle; + bool ice_lite; + bool ice_renomination; + string ice_ufrag; + string ice_pwd; + std::vector candidate; + + //////// dtls //////// + DtlsRole role; + SdpAttrFingerprint fingerprint; + + //////// extmap //////// + vector extmap; +}; From b714dfdddaf469b0465fd2bd6f4b1f45ab128b73 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 11:39:58 +0800 Subject: [PATCH 020/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E8=BE=85=E5=8A=A9?= =?UTF-8?q?=E6=96=B9=E6=B3=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 122 +++++++++++++++++++++++++++++++++++++------------ webrtc/Sdp.h | 66 +++++++++++++++++++++----- 2 files changed, 148 insertions(+), 40 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index dae5d776..d4a1d8f3 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -161,9 +161,99 @@ const char* getRtpDirectionString(RtpDirection val){ ////////////////////////////////////////////////////////////////////////////////////////// +string RtcSdpBase::toString() const { + _StrPrinter printer; + for (auto &item : items) { + printer << item->getKey() << "=" << item->toString() << "\r\n"; + } + return std::move(printer); +} + +RtpDirection RtcSdpBase::getDirection() const{ + for (auto &item : items) { + auto attr = dynamic_pointer_cast(item); + if (attr) { + auto dir = dynamic_pointer_cast(attr->detail); + if (dir) { + return dir->getDirection(); + } + } + } + return RtpDirection::invalid; +} + +SdpItem::Ptr RtcSdpBase::getItem(char key, const char *attr_key) const { + for (auto item : items) { + if (item->getKey()[0] == key) { + if (!attr_key) { + return item; + } + auto attr = dynamic_pointer_cast(item); + if (attr && attr->detail->getKey() == attr_key) { + return item; + } + } + } + return SdpItem::Ptr(); +} + +int RtcSdpBase::getVersion() const { + return atoi(getStringItem('v').data()); +} + +SdpOrigin RtcSdpBase::getOrigin() const { + return getItemClass('o'); +} + +string RtcSdpBase::getSessionName() const { + return getStringItem('s'); +} + +string RtcSdpBase::getSessionInfo() const { + return getStringItem('i'); +} + +SdpTime RtcSdpBase::getSessionTime() const{ + return getItemClass('t'); +} + +SdpConnection RtcSdpBase::getConnection() const { + return getItemClass('c'); +} + +SdpBandwidth RtcSdpBase::getBandwidth() const { + return getItemClass('b'); +} + +string RtcSdpBase::getUri() const { + return getStringItem('u'); +} + +string RtcSdpBase::getEmail() const { + return getStringItem('e'); +} + +string RtcSdpBase::getPhone() const { + return getStringItem('p'); +} + +string RtcSdpBase::getTimeZone() const { + return getStringItem('z'); +} + +string RtcSdpBase::getEncryptKey() const { + return getStringItem('k'); +} + +string RtcSdpBase::getRepeatTimes() const { + return getStringItem('r'); +} + +////////////////////////////////////////////////////////////////////// + void RtcSessionSdp::parse(const string &str) { static auto flag = registerAllItem(); - RtcMediaSdp *media = nullptr; + RtcSdpBase *media = nullptr; auto lines = split(str, "\n"); for(auto &line : lines){ trim(line); @@ -173,7 +263,7 @@ void RtcSessionSdp::parse(const string &str) { auto key = line.substr(0, 1); auto value = line.substr(2); if (key == "m") { - medias.emplace_back(RtcMediaSdp()); + medias.emplace_back(RtcSdpBase()); media = &medias.back(); } @@ -195,9 +285,7 @@ void RtcSessionSdp::parse(const string &str) { string RtcSessionSdp::toString() const { _StrPrinter printer; - for (auto &item : items) { - printer << item->getKey() << "=" << item->toString() << "\r\n"; - } + printer << RtcSdpBase::toString(); for (auto &media : medias) { printer << media.toString(); } @@ -205,30 +293,6 @@ string RtcSessionSdp::toString() const { return std::move(printer); } -////////////////////////////////////////////////////////////////////// - -string RtcMediaSdp::toString() const { - _StrPrinter printer; - for (auto &item : items) { - printer << item->getKey() << "=" << item->toString() << "\r\n"; - } - return std::move(printer); -} - -RtpDirection RtcMediaSdp::getDirection() const{ - for (auto &item : items) { - auto attr = dynamic_pointer_cast(item); - if (attr) { - auto dir = dynamic_pointer_cast(attr->detail); - if (dir) { - return dir->getDirection(); - } - } - } - return RtpDirection::invalid; -} - - ////////////////////////////////////////////////////////////////////////////////////////// #define SDP_THROW() throw std::invalid_argument(StrPrinter << "解析sdp " << getKey() << " 字段失败:" << str) diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 325c11c0..381f51c3 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -421,22 +421,55 @@ public: const char* getKey() const override { return "candidate";} }; -class RtcMediaSdp { +class RtcSdpBase { public: vector items; - string toString() const; - bool haveAttr(const char *attr) const; - string getAttrValue(const char *attr) const; + +public: + virtual string toString() const; + + int getVersion() const; + SdpOrigin getOrigin() const; + string getSessionName() const; + string getSessionInfo() const; + SdpTime getSessionTime() const; + SdpConnection getConnection() const; + SdpBandwidth getBandwidth() const; + + string getUri() const; + string getEmail() const; + string getPhone() const; + string getTimeZone() const; + string getEncryptKey() const; + string getRepeatTimes() const; RtpDirection getDirection() const; + +private: + SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const; + + template + cls getItemClass(char key, const char *attr_key = nullptr) const{ + auto item = dynamic_pointer_cast(getItem(key, attr_key)); + if (!item) { + return cls(); + } + return *item; + } + + string getStringItem(char key, const char *attr_key = nullptr) const{ + auto item = getItem(key, attr_key); + if (!item) { + return ""; + } + return item->toString(); + } }; -class RtcSessionSdp { +class RtcSessionSdp : public RtcSdpBase{ public: - vector items; - vector medias; - + vector medias; void parse(const string &str); - string toString() const; + string toString() const override; }; ////////////////////////////////////////////////////////////////// @@ -468,12 +501,12 @@ public: uint32_t sample_rate; //音频时有效 uint32_t channel = 0; - vector > fmtp; vector rtcp_fb; + vector > fmtp; }; //rtc 媒体描述 -class RtcCodec{ +class RtcMedia{ public: TrackType type; string mid; @@ -508,6 +541,17 @@ public: vector extmap; }; +class RtcSession{ +public: + int version; + SdpOrigin origin; + string session_name; + string session_info; + SdpConnection connection; + SdpBandwidth bandwidth; + set group_bundle; + vector media; +}; #endif //ZLMEDIAKIT_SDP_H From ba06def4d8113d2c046998e0eac8fd513a4067e6 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 12:00:42 +0800 Subject: [PATCH 021/218] =?UTF-8?q?=E5=BC=80=E5=A7=8B=E6=8F=90=E5=8F=96?= =?UTF-8?q?=E6=B1=87=E6=80=BBsdp=E4=BF=A1=E6=81=AF?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 31 ++++++++++++++++++++++++++++++- webrtc/Sdp.h | 8 +++++--- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index d4a1d8f3..4db1f14c 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -190,7 +190,7 @@ SdpItem::Ptr RtcSdpBase::getItem(char key, const char *attr_key) const { } auto attr = dynamic_pointer_cast(item); if (attr && attr->detail->getKey() == attr_key) { - return item; + return attr->detail; } } } @@ -869,3 +869,32 @@ void test_sdp(){ InfoL << sdp1.toString(); InfoL << sdp2.toString(); } + +void RtcSession::loadFrom(const string &str) { + RtcSessionSdp sdp; + sdp.parse(str); + + version = sdp.getVersion(); + origin = sdp.getOrigin(); + session_name = sdp.getSessionName(); + session_info = sdp.getSessionInfo(); + connection = sdp.getConnection(); + bandwidth = sdp.getBandwidth(); + auto group = sdp.getItemClass('a', "group"); + auto mids = group.mids; + + for (auto &media : sdp.medias) { + auto mline = media.getItemClass('m'); + switch (mline.type) { + case TrackVideo: + case TrackAudio: + case TrackApplication: + break; + default: throw std::invalid_argument(StrPrinter << "不识别的media类型:" << mline.toString()); + } + RtcMedia rtc_media; + rtc_media.type = mline.type; + rtc_media.mid = media.getStringItem('a', "mid"); + } + +} diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 381f51c3..ec38fb05 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -444,9 +444,6 @@ public: string getRepeatTimes() const; RtpDirection getDirection() const; -private: - SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const; - template cls getItemClass(char key, const char *attr_key = nullptr) const{ auto item = dynamic_pointer_cast(getItem(key, attr_key)); @@ -463,6 +460,9 @@ private: } return item->toString(); } + +private: + SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const; }; class RtcSessionSdp : public RtcSdpBase{ @@ -551,6 +551,8 @@ public: SdpBandwidth bandwidth; set group_bundle; vector media; + + void loadFrom(const string &sdp); }; From 9d2498a694e24175ad6a947894cbb393f6eaf487 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 12:28:47 +0800 Subject: [PATCH 022/218] =?UTF-8?q?candidate=20foundation=E6=94=B9?= =?UTF-8?q?=E4=B8=BAstring=E7=B1=BB=E5=9E=8B?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 16 ++++++++-------- server/main.cpp | 2 ++ webrtc/Sdp.cpp | 8 +++++--- webrtc/Sdp.h | 2 +- 4 files changed, 16 insertions(+), 12 deletions(-) diff --git a/conf/config.ini b/conf/config.ini index 59a7f06b..a2d66b04 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -140,14 +140,14 @@ maxReqSize=4096 #404网页内容，用户可以自定义404网页 notFound=404 Not Found

您访问的资源不存在！

ZLMediaKit-4.0
#http服务器监听端口 -port=80 +port=0 #http文件服务器根目录 #可以为相对(相对于本可执行程序目录)或绝对路径 rootPath=./www #http文件服务器读文件缓存大小，单位BYTE，调整该参数可以优化文件io性能 sendBufSize=65536 #https服务器监听端口 -sslport=443 +sslport=20443 #是否显示文件夹菜单，开启后可以浏览文件夹 dirMenu=1 @@ -187,9 +187,9 @@ keepAliveSecond=15 #在接收rtmp推流时，是否重新生成时间戳(很多推流器的时间戳着实很烂) modifyStamp=0 #rtmp服务器监听端口 -port=1935 +port=8935 #rtmps服务器监听地址 -sslport=19350 +sslport=0 [rtp] #音频mtu大小，该参数限制rtp最大字节数，推荐不要超过1400 @@ -202,7 +202,7 @@ videoMtuSize=1400 #导出调试数据(包括rtp/ps/h264)至该目录,置空则关闭数据导出 dumpDir= #udp和tcp代理服务器，支持rtp(必须是ts或ps类型)代理 -port=10000 +port=0 #rtp超时时间，单位秒 timeoutSec=15 @@ -221,13 +221,13 @@ handshakeSecond=15 #或者tcp发送缓存超过这个时间，则会断开连接，单位秒 keepAliveSecond=15 #rtsp服务器监听地址 -port=554 +port=8554 #rtsps服务器监听地址 -sslport=322 +sslport=0 [shell] #调试telnet服务器接受最大bufffer大小 maxReqSize=1024 #调试telnet服务器监听端口 -port=9000 +port=0 diff --git a/server/main.cpp b/server/main.cpp index 75264ff3..e70ed504 100644 --- a/server/main.cpp +++ b/server/main.cpp @@ -209,6 +209,7 @@ static void inline listen_shell_input(){ //全局变量，在WebApi中用于保存配置文件用 string g_ini_file; +extern void test_sdp(); int start_main(int argc,char *argv[]) { { CMD_main cmd_main; @@ -266,6 +267,7 @@ int start_main(int argc,char *argv[]) { }); } + test_sdp(); uint16_t shellPort = mINI::Instance()[Shell::kPort]; uint16_t rtspPort = mINI::Instance()[Rtsp::kPort]; uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort]; diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 4db1f14c..04766eda 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -689,14 +689,16 @@ string SdpAttrSctpMap::toString() const { } void SdpAttrCandidate::parse(const string &str) { + char foundation_buf[32] = {0}; char transport_buf[16] = {0}; char address_buf[32] = {0}; char type_buf[16] = {0}; - if (7 != sscanf(str.data(), "%" SCNu32 " %" SCNu32 " %15[^ ] %" SCNu32 " %31[^ ] %" SCNu16 " typ %15[^ ]", - &foundation, &component, transport_buf, &priority, address_buf, &port, type_buf)) { + if (7 != sscanf(str.data(), "%31[^ ] %" SCNu32 " %15[^ ] %" SCNu32 " %31[^ ] %" SCNu16 " typ %15[^ ]", + foundation_buf, &component, transport_buf, &priority, address_buf, &port, type_buf)) { SDP_THROW(); } + foundation = foundation_buf; transport = transport_buf; address = address_buf; type = type_buf; @@ -720,7 +722,7 @@ void SdpAttrCandidate::parse(const string &str) { string SdpAttrCandidate::toString() const { if (value.empty()) { - value = to_string(foundation) + " " + to_string(component) + " " + transport + " " + to_string(priority) + + value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) + " " + address + " " + to_string(port) + " typ " + type; for (auto &pr : arr) { value += ' '; diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index ec38fb05..fd56ec26 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -407,7 +407,7 @@ public: //15.1. "candidate" Attribute //a=candidate:4 1 udp 2 192.168.1.7 58107 typ host //a=candidate:
typ - uint32_t foundation; + string foundation; uint32_t component; string transport {"udp"}; uint32_t priority; From 62036d0f5d7f3962fb1231b412b75cd2f274cf6d Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Mon, 29 Mar 2021 18:31:56 +0800 Subject: [PATCH 023/218] =?UTF-8?q?=E5=AE=8C=E5=96=84sdp=E6=8F=90=E5=8F=96?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 11 +++++++++++ webrtc/Sdp.h | 6 +++--- 2 files changed, 14 insertions(+), 3 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 04766eda..b30e6039 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -897,6 +897,17 @@ void RtcSession::loadFrom(const string &str) { RtcMedia rtc_media; rtc_media.type = mline.type; rtc_media.mid = media.getStringItem('a', "mid"); + rtc_media.proto = mline.proto; + rtc_media.type = mline.type; + rtc_media.port = mline.port; + rtc_media.direction = media.getDirection(); + rtc_media.rtp_addr = media.getItemClass('c'); + rtc_media.rtcp_addr = media.getItemClass('a',"rtcp"); + rtc_media.ice_ufrag = media.getStringItem('a',"ice-ufrag"); + rtc_media.ice_ufrag = media.getStringItem('a',"ice-pwd"); + rtc_media.fingerprint = media.getItemClass('a', "fingerprint"); + rtc_media.role = media.getItemClass('a',"setup").role; +// rtc_media.rtcp_mux = } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index fd56ec26..78ee3b6d 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -494,7 +494,7 @@ public: }; //rtc传输编码方案 -class RtcPlan{ +class RtcCodecPlan{ public: uint8_t pt; string codec; @@ -514,10 +514,10 @@ public: string proto; //////// rtp //////// - vector plan; + RtcSSRC ssrc; SdpConnection rtp_addr; RtpDirection direction; - RtcSSRC ssrc; + vector plan; //////// rtx - rtcp //////// bool rtcp_mux; From 87c53dab92d54d146b0e3d91125f4282b8cb504e Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 29 Mar 2021 23:03:55 +0800 Subject: [PATCH 024/218] =?UTF-8?q?=E5=AE=8C=E6=88=90sdp=E7=9A=84=E5=88=86?= =?UTF-8?q?=E6=9E=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 173 ++++++++++++++++++++++++++++++++++++++++++------- webrtc/Sdp.h | 62 ++++++++++++++---- 2 files changed, 198 insertions(+), 37 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index b30e6039..0cfa1892 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -189,7 +189,7 @@ SdpItem::Ptr RtcSdpBase::getItem(char key, const char *attr_key) const { return item; } auto attr = dynamic_pointer_cast(item); - if (attr && attr->detail->getKey() == attr_key) { + if (attr && !strcmp(attr->detail->getKey() , attr_key)) { return attr->detail; } } @@ -554,23 +554,18 @@ string SdpAttrRtpMap::toString() const { return SdpItem::toString(); } -void SdpAttrRtcpFb::parse(const string &str) { - auto vec = split(str, " "); - if (vec.size() < 2) { +void SdpAttrRtcpFb::parse(const string &str_in) { + auto str = str_in + "\n"; + char rtcp_type_buf[32] = {0}; + if (2 != sscanf(str.data(), "%" SCNu8 " %31[^\n]", &pt, rtcp_type_buf)) { SDP_THROW(); } - pt = atoi(vec[0].data()); - vec.erase(vec.begin()); - arr = std::move(vec); + rtcp_type = rtcp_type_buf; } string SdpAttrRtcpFb::toString() const { if (value.empty()) { - value = to_string(pt); - for (auto &item : arr) { - value += ' '; - value += item; - } + value = to_string(pt) + " " + rtcp_type; } return SdpItem::toString(); } @@ -870,6 +865,13 @@ void test_sdp(){ } InfoL << sdp1.toString(); InfoL << sdp2.toString(); + + RtcSession session1; + session1.loadFrom(str1); + + RtcSession session2; + session2.loadFrom(str2); + InfoL; } void RtcSession::loadFrom(const string &str) { @@ -882,32 +884,153 @@ void RtcSession::loadFrom(const string &str) { session_info = sdp.getSessionInfo(); connection = sdp.getConnection(); bandwidth = sdp.getBandwidth(); - auto group = sdp.getItemClass('a', "group"); - auto mids = group.mids; - + msid_semantic = sdp.getItemClass('a', "msid-semantic"); for (auto &media : sdp.medias) { auto mline = media.getItemClass('m'); switch (mline.type) { case TrackVideo: case TrackAudio: - case TrackApplication: - break; + case TrackApplication: break; default: throw std::invalid_argument(StrPrinter << "不识别的media类型:" << mline.toString()); } - RtcMedia rtc_media; + this->media.emplace_back(); + auto &rtc_media = this->media.back(); rtc_media.type = mline.type; rtc_media.mid = media.getStringItem('a', "mid"); rtc_media.proto = mline.proto; rtc_media.type = mline.type; rtc_media.port = mline.port; - rtc_media.direction = media.getDirection(); - rtc_media.rtp_addr = media.getItemClass('c'); - rtc_media.rtcp_addr = media.getItemClass('a',"rtcp"); - rtc_media.ice_ufrag = media.getStringItem('a',"ice-ufrag"); - rtc_media.ice_ufrag = media.getStringItem('a',"ice-pwd"); + rtc_media.addr = media.getItemClass('c'); + rtc_media.ice_ufrag = media.getStringItem('a', "ice-ufrag"); + rtc_media.ice_pwd = media.getStringItem('a', "ice-pwd"); + rtc_media.role = media.getItemClass('a', "setup").role; rtc_media.fingerprint = media.getItemClass('a', "fingerprint"); - rtc_media.role = media.getItemClass('a',"setup").role; -// rtc_media.rtcp_mux = + rtc_media.ice_trickle = media.getItem('a', "ice-trickle").operator bool(); + rtc_media.ice_lite = media.getItem('a', "ice-lite").operator bool(); + rtc_media.ice_renomination = media.getItem('a', "ice-renomination").operator bool(); + rtc_media.candidate = media.getAllItem('a', "candidate"); + + if (mline.type == TrackType::TrackApplication) { + rtc_media.sctp_port = atoi(media.getStringItem('a', "sctp-port").data()); + rtc_media.sctpmap = media.getItemClass('a', "sctpmap"); + continue; + } + rtc_media.rtcp_addr = media.getItemClass('a', "rtcp"); + rtc_media.direction = media.getDirection(); + rtc_media.extmap = media.getAllItem('a', "extmap"); + rtc_media.rtcp_mux = media.getItem('a', "rtcp-mux").operator bool(); + rtc_media.rtcp_rsize = media.getItem('a', "rtcp-rsize").operator bool(); + + map rtc_ssrc_map; + for (auto &ssrc : media.getAllItem('a', "ssrc")) { + auto &rtc_ssrc = rtc_ssrc_map[ssrc.ssrc]; + rtc_ssrc.ssrc = ssrc.ssrc; + if (ssrc.attribute == "cname") { + rtc_ssrc.cname = ssrc.attribute_value; + continue; + } + if (ssrc.attribute == "msid") { + rtc_ssrc.msid = ssrc.attribute_value; + continue; + } + if (ssrc.attribute == "mslabel") { + rtc_ssrc.mslabel = ssrc.attribute_value; + continue; + } + if (ssrc.attribute == "label") { + rtc_ssrc.label = ssrc.attribute_value; + continue; + } + } + + uint32_t ssrc_rtp = 0, ssrc_rtx = 0, ssrc_rtp_low = 0, ssrc_rtp_mid = 0, ssrc_rtp_high = 0; + auto ssrc_groups = media.getAllItem('a', "ssrc-group"); + for (auto &group : ssrc_groups) { + if (group.isFID()) { + ssrc_rtp = group.u.fid.rtp_ssrc; + ssrc_rtx = group.u.fid.rtx_ssrc; + rtc_media.rtx = true; + } else if (group.isSIM()) { + rtc_media.simulcast = true; + ssrc_rtp_low = group.u.sim.rtp_ssrc_low; + ssrc_rtp_mid = group.u.sim.rtp_ssrc_mid; + ssrc_rtp_high = group.u.sim.rtp_ssrc_high; + } + } + + if (!ssrc_rtp) { + //没有指定ssrc-group字段，那么只有一个ssrc + if (rtc_ssrc_map.size() > 1) { + throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却有多个"); + } + ssrc_rtp = rtc_ssrc_map.begin()->second.ssrc; + } + for (auto &pr : rtc_ssrc_map) { + auto &rtc_ssrc = pr.second; + if (rtc_ssrc.ssrc == ssrc_rtp) { + rtc_media.rtp_ssrc = rtc_ssrc; + } + if (rtc_ssrc.ssrc == ssrc_rtx) { + rtc_media.rtx_ssrc = rtc_ssrc; + } + if (rtc_ssrc.ssrc == ssrc_rtp_low) { + rtc_media.rtp_ssrc_low = rtc_ssrc; + } + if (rtc_ssrc.ssrc == ssrc_rtp_mid) { + rtc_media.rtp_ssrc_mid = rtc_ssrc; + } + if (rtc_ssrc.ssrc == ssrc_rtp_high) { + rtc_media.rtp_ssrc_high = rtc_ssrc; + } + } + + auto rtpmap_arr = media.getAllItem('a', "rtpmap"); + auto rtcpfb_arr = media.getAllItem('a', "rtcp-fb"); + auto fmtp_aar = media.getAllItem('a', "fmtp"); + //方便根据pt查找rtpmap,一个pt必有一条 + map rtpmap_map; + //方便根据pt查找rtcp-fb,一个pt可能有多条或0条 + multimap rtcpfb_map; + //方便根据pt查找fmtp，一个pt最多一条 + map fmtp_map; + + for (auto &rtpmap : rtpmap_arr) { + if (!rtpmap_map.emplace(rtpmap.pt, rtpmap).second) { + //添加失败，有多条 + throw std::invalid_argument(StrPrinter << "该pt存在多条a=rtpmap:" << rtpmap.pt); + } + } + for (auto &rtpfb : rtcpfb_arr) { + rtcpfb_map.emplace(rtpfb.pt, rtpfb); + } + for (auto &fmtp : fmtp_aar) { + if (!fmtp_map.emplace(fmtp.pt, fmtp).second) { + //添加失败，有多条 + throw std::invalid_argument(StrPrinter << "该pt存在多条a=fmtp:" << fmtp.pt); + } + } + for (auto &pt : mline.fmts) { + //遍历所有编码方案的pt + rtc_media.plan.emplace_back(); + auto &plan = rtc_media.plan.back(); + auto rtpmap_it = rtpmap_map.find(pt); + if (rtpmap_it == rtpmap_map.end()) { + throw std::invalid_argument(StrPrinter << "该pt不存在相对于的a=rtpmap:" << pt); + } + plan.pt = rtpmap_it->second.pt; + plan.codec = rtpmap_it->second.codec; + plan.sample_rate = rtpmap_it->second.sample_rate; + plan.channel = rtpmap_it->second.channel; + auto fmtp_it = fmtp_map.find(pt); + if (fmtp_it != fmtp_map.end()) { + plan.fmtp = fmtp_it->second.arr; + } + for (auto rtpfb_it = rtcpfb_map.find(pt); + rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) { + plan.rtcp_fb.emplace_back(rtpfb_it->second.rtcp_type); + } + } } + group = sdp.getItemClass('a', "group"); } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 78ee3b6d..ed2ddad5 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -294,7 +294,7 @@ class SdpAttrExtmap : public SdpItem { public: //https://aggresss.blog.csdn.net/article/details/106436703 //a=extmap:1[/sendonly] urn:ietf:params:rtp-hdrext:ssrc-audio-level - int index; + uint32_t index; RtpDirection direction{RtpDirection::invalid}; string ext; void parse(const string &str) override; @@ -307,8 +307,8 @@ public: //a=rtpmap:111 opus/48000/2 uint8_t pt; string codec; - int sample_rate; - int channel {0}; + uint32_t sample_rate; + uint32_t channel {0}; void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "rtpmap";} @@ -323,7 +323,7 @@ public: //a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 //a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。 uint8_t pt; - vector arr; + string rtcp_type; void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "rtcp-fb";} @@ -383,6 +383,8 @@ public: } sim; } u; + bool isFID() const { return type == "FID"; } + bool isSIM() const { return type == "SIM"; } void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "ssrc-group";} @@ -395,7 +397,7 @@ public: //a=sctpmap: sctpmap-number media-subtypes [streams] uint16_t port; string subtypes; - int streams; + uint32_t streams; void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "sctpmap";} @@ -426,6 +428,7 @@ public: vector items; public: + virtual ~RtcSdpBase() = default; virtual string toString() const; int getVersion() const; @@ -461,8 +464,31 @@ public: return item->toString(); } -private: SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const; + + template + vector getAllItem(char key, const char *attr_key = nullptr) const { + vector ret; + for (auto item : items) { + if (item->getKey()[0] == key) { + if (!attr_key) { + auto c = dynamic_pointer_cast(item); + if (c) { + ret.emplace_back(*c); + } + } else { + auto attr = dynamic_pointer_cast(item); + if (attr && !strcmp(attr->detail->getKey(), attr_key)) { + auto c = dynamic_pointer_cast(attr->detail); + if (c) { + ret.emplace_back(*c); + } + } + } + } + } + return ret; + } }; class RtcSessionSdp : public RtcSdpBase{ @@ -486,7 +512,7 @@ enum class RtcSSRCType { //ssrc相关信息 class RtcSSRC{ public: - RtcSSRCType type; + uint32_t ssrc {0}; string cname; string msid; string mslabel; @@ -514,15 +540,22 @@ public: string proto; //////// rtp //////// - RtcSSRC ssrc; - SdpConnection rtp_addr; + RtcSSRC rtp_ssrc; + // for simulcast + bool simulcast{false}; + RtcSSRC rtp_ssrc_low; + RtcSSRC rtp_ssrc_mid; + RtcSSRC rtp_ssrc_high; + + SdpConnection addr; RtpDirection direction; vector plan; //////// rtx - rtcp //////// + bool rtx{false}; bool rtcp_mux; bool rtcp_rsize; - uint32_t rtx_ssrc; + RtcSSRC rtx_ssrc; SdpAttrRtcp rtcp_addr; //////// ice //////// @@ -539,18 +572,23 @@ public: //////// extmap //////// vector extmap; + + //////// sctp //////////// + SdpAttrSctpMap sctpmap; + uint32_t sctp_port {0}; }; class RtcSession{ public: - int version; + uint32_t version; SdpOrigin origin; string session_name; string session_info; SdpConnection connection; SdpBandwidth bandwidth; - set group_bundle; + SdpAttrMsidSemantic msid_semantic; vector media; + SdpAttrGroup group; void loadFrom(const string &sdp); }; From a1b2aa9abb3cdf9445e82f58e9b9b406d46c509d Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 29 Mar 2021 23:55:29 +0800 Subject: [PATCH 025/218] =?UTF-8?q?=E6=95=B4=E7=90=86=E4=B8=8E=E6=A0=A1?= =?UTF-8?q?=E9=AA=8C?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 53 ++++++++++++++++++++++++++++++++++++++++++++++++-- webrtc/Sdp.h | 43 +++++++++++++++++++++++----------------- 2 files changed, 76 insertions(+), 20 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 0cfa1892..10074f7d 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -949,9 +949,7 @@ void RtcSession::loadFrom(const string &str) { if (group.isFID()) { ssrc_rtp = group.u.fid.rtp_ssrc; ssrc_rtx = group.u.fid.rtx_ssrc; - rtc_media.rtx = true; } else if (group.isSIM()) { - rtc_media.simulcast = true; ssrc_rtp_low = group.u.sim.rtp_ssrc_low; ssrc_rtp_mid = group.u.sim.rtp_ssrc_mid; ssrc_rtp_high = group.u.sim.rtp_ssrc_high; @@ -1033,4 +1031,55 @@ void RtcSession::loadFrom(const string &str) { } group = sdp.getItemClass('a', "group"); + checkValid(); } + +string RtcCodecPlan::getFmtp(const char *key) const{ + for (auto &item : fmtp) { + if (item.first == key) { + return item.second; + } + } + return ""; +} + +const RtcCodecPlan *RtcMedia::getPlan(uint8_t pt) const{ + for (auto &item : plan) { + if (item.pt == pt) { + return &item; + } + } + return nullptr; +} + +void RtcMedia::checkValid() const{ + switch (direction) { + case RtpDirection::sendonly: + case RtpDirection::sendrecv: { + if (rtp_ssrc.empty()) { + throw std::invalid_argument("发送rtp但是未指定rtp ssrc"); + } + break; + } + default: break; + } + for (auto &item : plan) { + if (item.codec == "rtx") { + if (rtx_ssrc.empty()) { + throw std::invalid_argument("指定开启rtx但是未指定rtx ssrc"); + } + auto apt = atoi(item.getFmtp("apt").data()); + if (!getPlan(apt)) { + throw std::invalid_argument("找不到rtx关联的plan信息"); + } + } + } + //todo 校验更多信息 +} + +void RtcSession::checkValid() const{ + for (auto &item : media) { + item.checkValid(); + } + //todo 校验更多信息 +} \ No newline at end of file diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index ed2ddad5..60f8c6b8 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -517,6 +517,8 @@ public: string msid; string mslabel; string label; + + bool empty() const {return ssrc == 0;} }; //rtc传输编码方案 @@ -527,47 +529,48 @@ public: uint32_t sample_rate; //音频时有效 uint32_t channel = 0; + //rtcp反馈 vector rtcp_fb; vector > fmtp; + + string getFmtp(const char *key) const; }; //rtc 媒体描述 class RtcMedia{ public: - TrackType type; + TrackType type{TrackType::TrackInvalid}; string mid; - uint16_t port; + uint16_t port{0}; + SdpConnection addr; string proto; + RtpDirection direction{RtpDirection::invalid}; + vector plan; //////// rtp //////// RtcSSRC rtp_ssrc; - // for simulcast - bool simulcast{false}; + RtcSSRC rtx_ssrc; + + //////// simulcast //////// RtcSSRC rtp_ssrc_low; RtcSSRC rtp_ssrc_mid; RtcSSRC rtp_ssrc_high; - SdpConnection addr; - RtpDirection direction; - vector plan; - - //////// rtx - rtcp //////// - bool rtx{false}; - bool rtcp_mux; - bool rtcp_rsize; - RtcSSRC rtx_ssrc; + //////// rtcp //////// + bool rtcp_mux{false}; + bool rtcp_rsize{false}; SdpAttrRtcp rtcp_addr; //////// ice //////// - bool ice_trickle; - bool ice_lite; - bool ice_renomination; + bool ice_trickle{false}; + bool ice_lite{false}; + bool ice_renomination{false}; string ice_ufrag; string ice_pwd; std::vector candidate; //////// dtls //////// - DtlsRole role; + DtlsRole role{DtlsRole::invalid}; SdpAttrFingerprint fingerprint; //////// extmap //////// @@ -575,7 +578,10 @@ public: //////// sctp //////////// SdpAttrSctpMap sctpmap; - uint32_t sctp_port {0}; + uint32_t sctp_port{0}; + + void checkValid() const; + const RtcCodecPlan *getPlan(uint8_t pt) const; }; class RtcSession{ @@ -591,6 +597,7 @@ public: SdpAttrGroup group; void loadFrom(const string &sdp); + void checkValid() const; }; From daef7051adead95d87eebd10816ed60022d288bd Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 09:53:58 +0800 Subject: [PATCH 026/218] =?UTF-8?q?sdp=E5=AE=8C=E6=95=B4=E6=80=A7=E6=A3=80?= =?UTF-8?q?=E6=9F=A5=EF=BC=8C=E4=B8=8E=E5=BF=BD=E7=95=A5=E5=A4=A7=E5=B0=8F?= =?UTF-8?q?=E5=86=99=E6=8B=BC=E5=86=99?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- 3rdpart/assert.h | 17 ++++--- webrtc/Sdp.cpp | 129 ++++++++++++++++++++++++----------------------- webrtc/Sdp.h | 18 +++++-- 3 files changed, 90 insertions(+), 74 deletions(-) diff --git a/3rdpart/assert.h b/3rdpart/assert.h index c3b4b1c2..e5699be1 100644 --- a/3rdpart/assert.h +++ b/3rdpart/assert.h @@ -12,19 +12,20 @@ #define ZLMEDIAKIT_ASSERT_H #include + +#ifdef __cplusplus +extern "C" { +#endif +extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line); +#ifdef __cplusplus +} +#endif + #ifndef NDEBUG #ifdef assert #undef assert #endif//assert - #ifdef __cplusplus - extern "C" { - #endif - extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line); - #ifdef __cplusplus - } - #endif - #define assert(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); #else #define assert(e) ((void)0) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 10074f7d..9b00fd4c 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -3,10 +3,14 @@ // #include "Sdp.h" +#include "assert.h" +#include "Common/Parser.h" #include +#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); + using onCreateSdpItem = function; -static unordered_map sdpItemCreator; +static map sdpItemCreator; template void registerSdpItem(){ @@ -89,17 +93,15 @@ static bool registerAllItem(){ return true; } -TrackType getTrackType(const string &str){ - if (str == "video") { - return TrackVideo; - } - if (str == "audio") { - return TrackAudio; - } - if (str == "application") { - return TrackApplication; - } - return TrackInvalid; +static map track_str_map = { + {"video", TrackVideo}, + {"audio", TrackAudio}, + {"application", TrackApplication} +}; + +TrackType getTrackType(const string &str) { + auto it = track_str_map.find(str); + return it == track_str_map.end() ? TrackInvalid : it->second; } const char* getTrackString(TrackType type){ @@ -111,17 +113,15 @@ const char* getTrackString(TrackType type){ } } -DtlsRole getDtlsRole(const string &str){ - if (str == "active") { - return DtlsRole::active; - } - if (str == "passive") { - return DtlsRole::passive; - } - if (str == "actpass") { - return DtlsRole::actpass; - } - return DtlsRole::invalid; +static map dtls_role_map = { + {"active", DtlsRole::active}, + {"passive", DtlsRole::passive}, + {"actpass", DtlsRole::actpass} +}; + +DtlsRole getDtlsRole(const string &str) { + auto it = dtls_role_map.find(str); + return it == dtls_role_map.end() ? DtlsRole::invalid : it->second; } const char* getDtlsRoleString(DtlsRole role){ @@ -133,20 +133,16 @@ const char* getDtlsRoleString(DtlsRole role){ } } -RtpDirection getRtpDirection(const string &str){ - if (str == "sendonly") { - return RtpDirection::sendonly; - } - if (str == "recvonly") { - return RtpDirection::recvonly; - } - if (str == "sendrecv") { - return RtpDirection::sendrecv; - } - if (str == "inactive") { - return RtpDirection::inactive; - } - return RtpDirection::invalid; +static map direction_map = { + {"sendonly", RtpDirection::sendonly}, + {"recvonly", RtpDirection::recvonly}, + {"sendrecv", RtpDirection::sendrecv}, + {"inactive", RtpDirection::inactive} +}; + +RtpDirection getRtpDirection(const string &str) { + auto it = direction_map.find(str); + return it == direction_map.end() ? RtpDirection::invalid : it->second; } const char* getRtpDirectionString(RtpDirection val){ @@ -182,14 +178,15 @@ RtpDirection RtcSdpBase::getDirection() const{ return RtpDirection::invalid; } -SdpItem::Ptr RtcSdpBase::getItem(char key, const char *attr_key) const { +SdpItem::Ptr RtcSdpBase::getItem(char key_c, const char *attr_key) const { for (auto item : items) { - if (item->getKey()[0] == key) { + string key(1, key_c); + if (strcasecmp(item->getKey(), key.data()) == 0) { if (!attr_key) { return item; } auto attr = dynamic_pointer_cast(item); - if (attr && !strcmp(attr->detail->getKey() , attr_key)) { + if (attr && !strcasecmp(attr->detail->getKey() , attr_key)) { return attr->detail; } } @@ -884,6 +881,7 @@ void RtcSession::loadFrom(const string &str) { session_info = sdp.getSessionInfo(); connection = sdp.getConnection(); bandwidth = sdp.getBandwidth(); + time = sdp.getSessionTime(); msid_semantic = sdp.getItemClass('a', "msid-semantic"); for (auto &media : sdp.medias) { auto mline = media.getItemClass('m'); @@ -1036,7 +1034,7 @@ void RtcSession::loadFrom(const string &str) { string RtcCodecPlan::getFmtp(const char *key) const{ for (auto &item : fmtp) { - if (item.first == key) { + if (strcasecmp(item.first.data(), key) == 0) { return item.second; } } @@ -1052,34 +1050,41 @@ const RtcCodecPlan *RtcMedia::getPlan(uint8_t pt) const{ return nullptr; } -void RtcMedia::checkValid() const{ - switch (direction) { - case RtpDirection::sendonly: - case RtpDirection::sendrecv: { - if (rtp_ssrc.empty()) { - throw std::invalid_argument("发送rtp但是未指定rtp ssrc"); - } - break; - } - default: break; - } +const RtcCodecPlan *RtcMedia::getPlan(const char *codec) const{ for (auto &item : plan) { - if (item.codec == "rtx") { - if (rtx_ssrc.empty()) { - throw std::invalid_argument("指定开启rtx但是未指定rtx ssrc"); - } - auto apt = atoi(item.getFmtp("apt").data()); - if (!getPlan(apt)) { - throw std::invalid_argument("找不到rtx关联的plan信息"); - } + if (strcasecmp(item.codec.data(), codec) == 0) { + return &item; } } - //todo 校验更多信息 + return nullptr; +} + +void RtcMedia::checkValid() const{ + CHECK(type != TrackInvalid); + CHECK(!mid.empty()); + CHECK(!proto.empty()); + CHECK(direction != RtpDirection::invalid || type == TrackApplication); + CHECK(!plan.empty() || type == TrackApplication ); + bool send_rtp = (direction == RtpDirection::sendonly || direction == RtpDirection::sendrecv); + CHECK(!rtp_ssrc.empty() || !send_rtp); + auto rtx_plan = getPlan("rtx"); + if (rtx_plan) { + //开启rtx后必须指定rtx_ssrc + CHECK(!rtx_ssrc.empty() || !send_rtp); + auto apt = atoi(rtx_plan->getFmtp("apt").data()); + //开启rtx后必须指定其关联的其他的plan + CHECK(getPlan(apt)); + } } void RtcSession::checkValid() const{ + CHECK(version == 0); + CHECK(!origin.empty()); + CHECK(!session_name.empty()); + CHECK(!msid_semantic.empty()); + CHECK(!media.empty()); + CHECK(group.mids.size() <= media.size()); for (auto &item : media) { item.checkValid(); } - //todo 校验更多信息 } \ No newline at end of file diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 60f8c6b8..32c92cb6 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -134,6 +134,10 @@ public: void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "o";} + bool empty() const { + return username.empty() || session_id.empty() || session_version.empty() + || nettype.empty() || addrtype.empty() || address.empty(); + } }; class SdpConnection : public SdpItem { @@ -233,6 +237,9 @@ public: void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "msid-semantic";} + bool empty() const { + return msid.empty(); + } }; class SdpAttrRtcp : public SdpItem { @@ -467,10 +474,11 @@ public: SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const; template - vector getAllItem(char key, const char *attr_key = nullptr) const { + vector getAllItem(char key_c, const char *attr_key = nullptr) const { vector ret; for (auto item : items) { - if (item->getKey()[0] == key) { + string key(1, key_c); + if (strcasecmp(item->getKey(), key.data()) == 0) { if (!attr_key) { auto c = dynamic_pointer_cast(item); if (c) { @@ -478,7 +486,7 @@ public: } } else { auto attr = dynamic_pointer_cast(item); - if (attr && !strcmp(attr->detail->getKey(), attr_key)) { + if (attr && !strcasecmp(attr->detail->getKey(), attr_key)) { auto c = dynamic_pointer_cast(attr->detail); if (c) { ret.emplace_back(*c); @@ -518,7 +526,7 @@ public: string mslabel; string label; - bool empty() const {return ssrc == 0;} + bool empty() const {return ssrc == 0 && cname.empty();} }; //rtc传输编码方案 @@ -582,6 +590,7 @@ public: void checkValid() const; const RtcCodecPlan *getPlan(uint8_t pt) const; + const RtcCodecPlan *getPlan(const char *codec) const; }; class RtcSession{ @@ -590,6 +599,7 @@ public: SdpOrigin origin; string session_name; string session_info; + SdpTime time; SdpConnection connection; SdpBandwidth bandwidth; SdpAttrMsidSemantic msid_semantic; From a3d0d3b7ada143fdfaee6c767791f792209b5de2 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 09:58:53 +0800 Subject: [PATCH 027/218] =?UTF-8?q?=E5=BF=BD=E7=95=A5=E5=A4=A7=E5=B0=8F?= =?UTF-8?q?=E5=86=99=E6=8B=BC=E5=86=99?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 9b00fd4c..e0b46367 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -259,7 +259,7 @@ void RtcSessionSdp::parse(const string &str) { } auto key = line.substr(0, 1); auto value = line.substr(2); - if (key == "m") { + if (!strcasecmp(key.data(), "m")) { medias.emplace_back(RtcSdpBase()); media = &medias.back(); } @@ -923,19 +923,19 @@ void RtcSession::loadFrom(const string &str) { for (auto &ssrc : media.getAllItem('a', "ssrc")) { auto &rtc_ssrc = rtc_ssrc_map[ssrc.ssrc]; rtc_ssrc.ssrc = ssrc.ssrc; - if (ssrc.attribute == "cname") { + if (!strcasecmp(ssrc.attribute.data(), "cname")) { rtc_ssrc.cname = ssrc.attribute_value; continue; } - if (ssrc.attribute == "msid") { + if (!strcasecmp(ssrc.attribute.data(), "msid")) { rtc_ssrc.msid = ssrc.attribute_value; continue; } - if (ssrc.attribute == "mslabel") { + if (!strcasecmp(ssrc.attribute.data(), "mslabel")) { rtc_ssrc.mslabel = ssrc.attribute_value; continue; } - if (ssrc.attribute == "label") { + if (!strcasecmp(ssrc.attribute.data(), "label")) { rtc_ssrc.label = ssrc.attribute_value; continue; } From a7a94f08857194aa80d8eb5efdb617f078d6b3a5 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 10:59:15 +0800 Subject: [PATCH 028/218] =?UTF-8?q?sdp=E7=9B=B8=E5=85=B3=E7=9A=84=E4=BB=A3?= =?UTF-8?q?=E7=A0=81=E7=A7=BB=E5=8A=A8=E5=88=B0=E4=B8=BB=E7=9B=AE=E5=BD=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/AAC.cpp | 2 +- src/Extension/Frame.cpp | 70 ++++++++++++++++++++++++++++++++--------- src/Extension/Frame.h | 19 +++++++++++ src/Extension/G711.cpp | 2 +- src/Extension/H264.cpp | 2 +- src/Extension/H265.cpp | 2 +- src/Extension/L16.cpp | 2 +- src/Extension/Opus.cpp | 2 +- webrtc/Sdp.cpp | 35 +++++++-------------- webrtc/Sdp.h | 3 +- 10 files changed, 93 insertions(+), 46 deletions(-) diff --git a/src/Extension/AAC.cpp b/src/Extension/AAC.cpp index d29ccc59..84308355 100644 --- a/src/Extension/AAC.cpp +++ b/src/Extension/AAC.cpp @@ -205,7 +205,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << " MPEG4-GENERIC/" << sample_rate << "/" << channels << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n"; string configStr; char buf[4] = {0}; diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index 39e04a45..aa005491 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -11,7 +11,7 @@ #include "Frame.h" #include "H264.h" #include "H265.h" - +#include "Common/Parser.h" using namespace std; using namespace toolkit; @@ -106,22 +106,10 @@ Frame::Ptr Frame::getCacheAbleFrame(const Frame::Ptr &frame){ return std::make_shared(frame); } -#define SWITCH_CASE(codec_id) case codec_id : return #codec_id -const char *getCodecName(CodecId codecId) { - switch (codecId) { - SWITCH_CASE(CodecH264); - SWITCH_CASE(CodecH265); - SWITCH_CASE(CodecAAC); - SWITCH_CASE(CodecG711A); - SWITCH_CASE(CodecG711U); - SWITCH_CASE(CodecOpus); - SWITCH_CASE(CodecL16); - default : return "unknown codec"; - } -} - TrackType getTrackType(CodecId codecId){ switch (codecId){ + case CodecVP8: + case CodecVP9: case CodecH264: case CodecH265: return TrackVideo; case CodecAAC: @@ -133,6 +121,58 @@ TrackType getTrackType(CodecId codecId){ } } +const char* getCodecName(CodecId codec){ + switch (codec) { + case CodecH264 : return "H264"; + case CodecH265 : return "H265"; + case CodecAAC : return "mpeg4-generic"; + case CodecG711A : return "PCMA"; + case CodecG711U : return "PCMU"; + case CodecOpus : return "opus"; + case CodecVP8 : return "VP8"; + case CodecVP9 : return "VP9"; + case CodecL16 : return "L16"; + default: return "invalid"; + } +} + +static map codec_map = { + {"H264", CodecH264}, + {"H265", CodecH265}, + {"mpeg4-generic", CodecAAC}, + {"PCMA", CodecG711A}, + {"PCMU", CodecG711U}, + {"opus", CodecOpus}, + {"VP8", CodecVP8}, + {"VP9", CodecVP9}, + {"L16", CodecL16} +}; + +CodecId getCodecId(const string &str){ + auto it = codec_map.find(str); + return it == codec_map.end() ? CodecInvalid : it->second; +} + +static map track_str_map = { + {"video", TrackVideo}, + {"audio", TrackAudio}, + {"application", TrackApplication} +}; + +TrackType getTrackType(const string &str) { + auto it = track_str_map.find(str); + return it == track_str_map.end() ? TrackInvalid : it->second; +} + +const char* getTrackString(TrackType type){ + switch (type) { + case TrackVideo : return "video"; + case TrackAudio : return "audio"; + case TrackApplication : return "application"; + default: return "invalid"; + } +} + const char *CodecInfo::getCodecName() { return mediakit::getCodecName(getCodecId()); } diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index afaa115a..1ad4734d 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -30,6 +30,8 @@ typedef enum { CodecG711U, CodecOpus, CodecL16, + CodecVP8, + CodecVP9, CodecMax = 0x7FFF } CodecId; @@ -42,6 +44,23 @@ typedef enum { TrackMax = 4 } TrackType; +/** + * 字符串转媒体类型转 + */ +TrackType getTrackType(const string &str); + +/** + * 媒体类型转字符串 + */ +const char* getTrackString(TrackType type); + +/** + * 根据SDP中描述获取codec_id + * @param str + * @return + */ +CodecId getCodecId(const string &str); + /** * 获取编码器名称 */ diff --git a/src/Extension/G711.cpp b/src/Extension/G711.cpp index da86d5a2..eab141ce 100644 --- a/src/Extension/G711.cpp +++ b/src/Extension/G711.cpp @@ -33,7 +33,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "/" << channels << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n"; _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n"; } diff --git a/src/Extension/H264.cpp b/src/Extension/H264.cpp index b7bd1e29..04c638e1 100644 --- a/src/Extension/H264.cpp +++ b/src/Extension/H264.cpp @@ -235,7 +235,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << " H264/" << 90000 << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << 90000 << "\r\n"; _printer << "a=fmtp:" << payload_type << " packetization-mode=1; profile-level-id="; char strTemp[1024]; diff --git a/src/Extension/H265.cpp b/src/Extension/H265.cpp index 48132a1d..27f12d1b 100644 --- a/src/Extension/H265.cpp +++ b/src/Extension/H265.cpp @@ -252,7 +252,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << " H265/" << 90000 << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << 90000 << "\r\n"; _printer << "a=fmtp:" << payload_type << " "; _printer << "sprop-vps="; _printer << encodeBase64(strVPS) << "; "; diff --git a/src/Extension/L16.cpp b/src/Extension/L16.cpp index e6c778ac..8194dfa3 100644 --- a/src/Extension/L16.cpp +++ b/src/Extension/L16.cpp @@ -33,7 +33,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << " L16/" << sample_rate << "/" << channels << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n"; _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n"; } diff --git a/src/Extension/Opus.cpp b/src/Extension/Opus.cpp index 275354cb..03e107a9 100644 --- a/src/Extension/Opus.cpp +++ b/src/Extension/Opus.cpp @@ -31,7 +31,7 @@ public: if (bitrate) { _printer << "b=AS:" << bitrate << "\r\n"; } - _printer << "a=rtpmap:" << payload_type << " opus/" << sample_rate << "/" << channels << "\r\n"; + _printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n"; _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n"; } diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index e0b46367..d24d5a06 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -93,26 +93,6 @@ static bool registerAllItem(){ return true; } -static map track_str_map = { - {"video", TrackVideo}, - {"audio", TrackAudio}, - {"application", TrackApplication} -}; - -TrackType getTrackType(const string &str) { - auto it = track_str_map.find(str); - return it == track_str_map.end() ? TrackInvalid : it->second; -} - -const char* getTrackString(TrackType type){ - switch (type) { - case TrackVideo : return "video"; - case TrackAudio : return "audio"; - case TrackApplication : return "application"; - default: return "invalid"; - } -} - static map dtls_role_map = { {"active", DtlsRole::active}, {"passive", DtlsRole::passive}, @@ -1059,6 +1039,18 @@ const RtcCodecPlan *RtcMedia::getPlan(const char *codec) const{ return nullptr; } +const RtcCodecPlan *RtcMedia::getRelatedRtxPlan(uint8_t pt) const{ + for (auto &item : plan) { + if (strcasecmp(item.codec.data(), "rtx") == 0) { + auto apt = atoi(item.getFmtp("apt").data()); + if (pt == apt) { + return &item; + } + } + } + return nullptr; +} + void RtcMedia::checkValid() const{ CHECK(type != TrackInvalid); CHECK(!mid.empty()); @@ -1071,9 +1063,6 @@ void RtcMedia::checkValid() const{ if (rtx_plan) { //开启rtx后必须指定rtx_ssrc CHECK(!rtx_ssrc.empty() || !send_rtp); - auto apt = atoi(rtx_plan->getFmtp("apt").data()); - //开启rtx后必须指定其关联的其他的plan - CHECK(getPlan(apt)); } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 32c92cb6..51163e01 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -72,8 +72,6 @@ enum class SdpType { answer }; -TrackType getTrackType(const string &str); -const char* getTrackString(TrackType type); DtlsRole getDtlsRole(const string &str); const char* getDtlsRoleString(DtlsRole role); RtpDirection getRtpDirection(const string &str); @@ -591,6 +589,7 @@ public: void checkValid() const; const RtcCodecPlan *getPlan(uint8_t pt) const; const RtcCodecPlan *getPlan(const char *codec) const; + const RtcCodecPlan *getRelatedRtxPlan(uint8_t pt) const; }; class RtcSession{ From 2065b6fea8845f9ac08f1ca6b2fc14c0e69aed83 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 11:51:19 +0800 Subject: [PATCH 029/218] =?UTF-8?q?=E5=8E=BB=E6=8E=89=E6=9C=80=E5=A4=A7?= =?UTF-8?q?=E6=9E=9A=E4=B8=BE=E5=AE=9A=E4=B9=89?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/Frame.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index 1ad4734d..d7c4d8f6 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -32,7 +32,7 @@ typedef enum { CodecL16, CodecVP8, CodecVP9, - CodecMax = 0x7FFF + CodecMax } CodecId; typedef enum { @@ -41,7 +41,7 @@ typedef enum { TrackAudio, TrackTitle, TrackApplication, - TrackMax = 4 + TrackMax } TrackType; /** From 6d16ae91e6eb43a3f1e6d16b9e76d0a59686ab46 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 11:51:39 +0800 Subject: [PATCH 030/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtc=E9=85=8D=E7=BD=AE?= =?UTF-8?q?=E7=B1=BB?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 79 ++++++++++++++++++++++++++++++++++++++++++++++++++ webrtc/Sdp.h | 44 ++++++++++++++++++++++++++++ 2 files changed, 123 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index d24d5a06..3b2cfc9d 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -493,6 +493,7 @@ void SdpAttrExtmap::parse(const string &str) { if (sscanf(str.data(), "%" SCNd32 " %127s", &index, buf) != 2) { SDP_THROW(); } + direction = RtpDirection::sendrecv; } else { direction = getRtpDirection(direction_buf); } @@ -1076,4 +1077,82 @@ void RtcSession::checkValid() const{ for (auto &item : media) { item.checkValid(); } +} + +void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ + enable = true; + rtcp_mux = true; + rtcp_rsize = false; + group_bundle = true; + unified_plan = false; + support_rtx = true; + support_red = false; + support_ulpfec = false; + support_simulcast = false; + ice_lite = true; + ice_trickle = true; + ice_renomination = false; + switch (type) { + case TrackVideo: { + preferred_codec = {CodecAAC, CodecOpus, CodecG711U, CodecG711A}; + rtcp_fb = {"transport-cc"}; + extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"}; + break; + } + case TrackAudio: { + preferred_codec = {CodecH264, CodecH265}; + rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"}; + extmap = {"2 urn:ietf:params:rtp-hdrext:toffset", + "3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", + "4 urn:3gpp:video-orientation", + "5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", + "6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"}; + break; + } + case TrackApplication: { + enable = false; + break; + } + default: break; + } +} + +void RtcConfigure::setDefaultSetting(string ice_ufrag, + string ice_pwd, + DtlsRole role, + RtpDirection direction, + const SdpAttrFingerprint &fingerprint) { + + video.setDefaultSetting(TrackVideo); + audio.setDefaultSetting(TrackAudio); + application.setDefaultSetting(TrackApplication); + + video.ice_ufrag = audio.ice_ufrag = application.ice_ufrag = ice_ufrag; + video.ice_pwd = audio.ice_pwd = application.ice_pwd = ice_ufrag; + video.role = audio.role = application.role = role; + video.direction = audio.direction = application.direction = direction; + video.fingerprint = audio.fingerprint = application.fingerprint = fingerprint; +} + +void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType type) { + switch (type) { + case TrackAudio: { + audio.candidate.emplace_back(candidate); + break; + } + case TrackVideo: { + video.candidate.emplace_back(candidate); + break; + } + case TrackApplication: { + application.candidate.emplace_back(candidate); + break; + } + default: { + audio.candidate.emplace_back(candidate); + video.candidate.emplace_back(candidate); + application.candidate.emplace_back(candidate); + break; + } + } } \ No newline at end of file diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 51163e01..efca5783 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -609,5 +609,49 @@ public: void checkValid() const; }; +class RtcConfigure { +public: + class RtcTrackConfigure { + public: + bool enable; + bool rtcp_mux; + bool rtcp_rsize; + bool group_bundle; + bool unified_plan; + bool support_rtx; + bool support_red; + bool support_ulpfec; + + bool support_simulcast; + bool ice_lite; + bool ice_trickle; + bool ice_renomination; + string ice_ufrag; + string ice_pwd; + + DtlsRole role{DtlsRole::invalid}; + RtpDirection direction{RtpDirection::invalid}; + SdpAttrFingerprint fingerprint; + + vector rtcp_fb; + vector preferred_codec; + vector extmap; + vector candidate; + + void setDefaultSetting(TrackType type); + }; + + RtcTrackConfigure video; + RtcTrackConfigure audio; + RtcTrackConfigure application; + + void setDefaultSetting(string ice_ufrag, + string ice_pwd, + DtlsRole role, + RtpDirection direction, + const SdpAttrFingerprint &fingerprint); + void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid); +}; + #endif //ZLMEDIAKIT_SDP_H From c6a5471a2379aec047fac6f79b1a5184c4efe622 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 13:09:02 +0800 Subject: [PATCH 031/218] =?UTF-8?q?=E5=AE=8C=E6=88=90sdp=E8=BD=ACstring?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 180 +++++++++++++++++++++++++++++++++++++++++++++++-- webrtc/Sdp.h | 19 +++++- 2 files changed, 194 insertions(+), 5 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 3b2cfc9d..b0aae5b5 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -52,6 +52,18 @@ public: RtpDirection getDirection() const override {return RtpDirection::inactive;} }; +class DirectionInterfaceImp : public SdpItem, public DirectionInterface{ +public: + DirectionInterfaceImp(RtpDirection direct){ + direction = direct; + } + const char* getKey() const override { return getRtpDirectionString(getDirection());} + RtpDirection getDirection() const override {return direction;} + +private: + RtpDirection direction; +}; + static bool registerAllItem(){ registerSdpItem >(); registerSdpItem >(); @@ -502,7 +514,7 @@ void SdpAttrExtmap::parse(const string &str) { string SdpAttrExtmap::toString() const { if (value.empty()) { - if(direction == RtpDirection::invalid){ + if(direction == RtpDirection::invalid || direction == RtpDirection::sendrecv){ value = to_string(index) + " " + ext; } else { value = to_string(index) + "/" + getRtpDirectionString(direction) + " " + ext; @@ -630,9 +642,9 @@ void SdpAttrSSRCGroup::parse(const string &str) { string SdpAttrSSRCGroup::toString() const { if (value.empty()) { - if (type == "FID") { + if (isFID()) { value = type + " " + to_string(u.fid.rtp_ssrc) + " " + to_string(u.fid.rtx_ssrc); - } else if (type == "SIM") { + } else if (isSIM()) { value = type + " " + to_string(u.sim.rtp_ssrc_low) + " " + to_string(u.sim.rtp_ssrc_mid) + " " + to_string(u.sim.rtp_ssrc_high); } else { SDP_THROW2(); @@ -849,7 +861,8 @@ void test_sdp(){ RtcSession session2; session2.loadFrom(str2); - InfoL; + DebugL << session1.toString(); + DebugL << session2.toString(); } void RtcSession::loadFrom(const string &str) { @@ -1013,6 +1026,165 @@ void RtcSession::loadFrom(const string &str) { checkValid(); } +std::shared_ptr wrapSdpAttr(SdpItem::Ptr item){ + auto ret = std::make_shared(); + ret->detail = std::move(item); + return ret; +} + +string RtcSession::toString() const{ + checkValid(); + RtcSessionSdp sdp; + sdp.items.emplace_back(std::make_shared >(to_string(version))); + sdp.items.emplace_back(std::make_shared(origin)); + sdp.items.emplace_back(std::make_shared >(session_name)); + if (!session_info.empty()) { + sdp.items.emplace_back(std::make_shared >(session_info)); + } + sdp.items.emplace_back(std::make_shared(time)); + sdp.items.emplace_back(std::make_shared(connection)); + if (!bandwidth.empty()) { + sdp.items.emplace_back(std::make_shared(bandwidth)); + } + sdp.items.emplace_back(wrapSdpAttr(std::make_shared(msid_semantic))); + sdp.items.emplace_back(wrapSdpAttr(std::make_shared(group))); + for (auto &m : media) { + sdp.medias.emplace_back(); + auto &sdp_media = sdp.medias.back(); + auto mline = std::make_shared(); + mline->type = m.type; + mline->port = m.port; + mline->proto = m.proto; + for (auto &p : m.plan) { + mline->fmts.emplace_back(p.pt); + } + if (m.type == TrackApplication) { + mline->fmts.emplace_back(m.sctp_port); + } + sdp_media.items.emplace_back(std::move(mline)); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.mid))); + sdp_media.items.emplace_back(std::make_shared(m.addr)); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.ice_ufrag))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.ice_pwd))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.fingerprint))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.role))); + if (m.ice_trickle) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-trickle"))); + } + if (m.ice_lite) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-lite"))); + } + if (m.ice_renomination) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-renomination"))); + } + for (auto &ext : m.extmap) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(ext))); + } + if (m.direction != RtpDirection::invalid) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.direction))); + } + if (m.rtcp_addr.port) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.rtcp_addr))); + } + if (m.rtcp_mux) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("rtcp-mux"))); + } + if (m.rtcp_rsize) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("rtcp-rsize"))); + } + + if(m.type != TrackApplication) { + for (auto &p : m.plan) { + auto rtp_map = std::make_shared(); + rtp_map->pt = p.pt; + rtp_map->codec = p.codec; + rtp_map->sample_rate = p.sample_rate; + rtp_map->channel = p.channel; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(rtp_map))); + + for (auto &fb : p.rtcp_fb) { + auto rtcp_fb = std::make_shared(); + rtcp_fb->pt = p.pt; + rtcp_fb->rtcp_type = fb; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(rtcp_fb))); + } + + if (!p.fmtp.empty()) { + auto fmtp = std::make_shared(); + fmtp->pt = p.pt; + fmtp->arr = p.fmtp; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(fmtp))); + } + } + + if (!m.rtp_ssrc.empty() && !m.rtx_ssrc.empty()) { + auto group = std::make_shared(); + group->type = "FID"; + group->u.fid.rtp_ssrc = m.rtp_ssrc.ssrc; + group->u.fid.rtx_ssrc = m.rtx_ssrc.ssrc; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); + } + + static auto addSSRCItem = [](const RtcSSRC &rtp_ssrc, vector &items) { + SdpAttrSSRC ssrc; + ssrc.ssrc = rtp_ssrc.ssrc; + + ssrc.attribute = "cname"; + ssrc.attribute_value = rtp_ssrc.cname; + items.emplace_back(wrapSdpAttr(std::make_shared(ssrc))); + + if (!rtp_ssrc.msid.empty()) { + ssrc.attribute = "msid"; + ssrc.attribute_value = rtp_ssrc.msid; + items.emplace_back(wrapSdpAttr(std::make_shared(ssrc))); + } + + if (!rtp_ssrc.mslabel.empty()) { + ssrc.attribute = "mslabel"; + ssrc.attribute_value = rtp_ssrc.mslabel; + items.emplace_back(wrapSdpAttr(std::make_shared(ssrc))); + } + + if (!rtp_ssrc.label.empty()) { + ssrc.attribute = "label"; + ssrc.attribute_value = rtp_ssrc.label; + items.emplace_back(wrapSdpAttr(std::make_shared(ssrc))); + } + }; + if (!m.rtp_ssrc.empty()) { + addSSRCItem(m.rtp_ssrc, sdp_media.items); + } + if (!m.rtx_ssrc.empty()) { + addSSRCItem(m.rtx_ssrc, sdp_media.items); + } + + bool enable_sim = false; + if (!m.rtp_ssrc_low.empty() && !m.rtp_ssrc_mid.empty() && !m.rtp_ssrc_high.empty()) { + auto group = std::make_shared(); + group->type = "SIM"; + group->u.sim.rtp_ssrc_low = m.rtp_ssrc_low.ssrc; + group->u.sim.rtp_ssrc_mid = m.rtp_ssrc_mid.ssrc; + group->u.sim.rtp_ssrc_high = m.rtp_ssrc_high.ssrc; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); + enable_sim = true; + } + if (enable_sim) { + addSSRCItem(m.rtp_ssrc_low, sdp_media.items); + addSSRCItem(m.rtp_ssrc_mid, sdp_media.items); + addSSRCItem(m.rtp_ssrc_high, sdp_media.items); + } + } else { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.sctpmap))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("sctp-port", to_string(m.sctp_port)))); + } + + for (auto &cand : m.candidate) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(cand))); + } + } + return sdp.toString(); +} + string RtcCodecPlan::getFmtp(const char *key) const{ for (auto &item : fmtp) { if (strcasecmp(item.first.data(), key) == 0) { diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index efca5783..5cf2f0c8 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -96,6 +96,8 @@ protected: template class SdpString : public SdpItem{ public: + SdpString() = default; + SdpString(string val) {value == std::move(val);} // *=* const char* getKey() const override { static string key(1, KEY); return key.data();} }; @@ -104,6 +106,11 @@ class SdpCommon : public SdpItem { public: string key; SdpCommon(string key) { this->key = std::move(key); } + SdpCommon(string key, string val) { + this->key = std::move(key); + this->value = std::move(val); + } + const char* getKey() const override { return key.data();} }; @@ -163,6 +170,7 @@ public: void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "b";} + bool empty() const {return bandwidth == 0;} }; class SdpMedia : public SdpItem { @@ -243,7 +251,7 @@ public: class SdpAttrRtcp : public SdpItem { public: // a=rtcp:9 IN IP4 0.0.0.0 - uint16_t port; + uint16_t port{0}; string nettype {"IN"}; string addrtype {"IP4"}; string address {"0.0.0.0"}; @@ -254,12 +262,16 @@ public: class SdpAttrIceUfrag : public SdpItem { public: + SdpAttrIceUfrag() = default; + SdpAttrIceUfrag(string str) {value = std::move(str);} //a=ice-ufrag:sXJ3 const char* getKey() const override { return "ice-ufrag";} }; class SdpAttrIcePwd : public SdpItem { public: + SdpAttrIcePwd() = default; + SdpAttrIcePwd(string str) {value = std::move(str);} //a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV const char* getKey() const override { return "ice-pwd";} }; @@ -283,6 +295,8 @@ public: class SdpAttrSetup : public SdpItem { public: //a=setup:actpass + SdpAttrSetup() = default; + SdpAttrSetup(DtlsRole r) { role = r; } DtlsRole role{DtlsRole::actpass}; void parse(const string &str) override; string toString() const override; @@ -291,6 +305,8 @@ public: class SdpAttrMid : public SdpItem { public: + SdpAttrMid() = default; + SdpAttrMid(string val) { value = std::move(val); } //a=mid:audio const char* getKey() const override { return "mid";} }; @@ -607,6 +623,7 @@ public: void loadFrom(const string &sdp); void checkValid() const; + string toString() const; }; class RtcConfigure { From e296ebae6e8782dc75f2ac69d0bc92b2b924cd55 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 13:11:59 +0800 Subject: [PATCH 032/218] bug fixed --- webrtc/Sdp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 5cf2f0c8..8a373d82 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -97,7 +97,7 @@ template class SdpString : public SdpItem{ public: SdpString() = default; - SdpString(string val) {value == std::move(val);} + SdpString(string val) {value = std::move(val);} // *=* const char* getKey() const override { static string key(1, KEY); return key.data();} }; From 45b8f83131060a36a6f511e87f53c9271611370e Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 13:53:45 +0800 Subject: [PATCH 033/218] =?UTF-8?q?=E5=88=A0=E9=99=A4=E6=B5=8B=E8=AF=95?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/main.cpp | 2 - webrtc/Sdp.cpp | 159 +++--------------------------------------------- webrtc/Sdp.h | 1 + 3 files changed, 10 insertions(+), 152 deletions(-) diff --git a/server/main.cpp b/server/main.cpp index e70ed504..75264ff3 100644 --- a/server/main.cpp +++ b/server/main.cpp @@ -209,7 +209,6 @@ static void inline listen_shell_input(){ //全局变量，在WebApi中用于保存配置文件用 string g_ini_file; -extern void test_sdp(); int start_main(int argc,char *argv[]) { { CMD_main cmd_main; @@ -267,7 +266,6 @@ int start_main(int argc,char *argv[]) { }); } - test_sdp(); uint16_t shellPort = mINI::Instance()[Shell::kPort]; uint16_t rtspPort = mINI::Instance()[Rtsp::kPort]; uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort]; diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index b0aae5b5..a6b548fd 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -719,152 +719,6 @@ string SdpAttrCandidate::toString() const { return SdpItem::toString(); } -void test_sdp(){ - char str1[] = "v=0\n" - "o=- 380154348540553537 2 IN IP4 127.0.0.1\n" - "s=-\n" - "b=CT:1900\n" - "t=0 0\n" - "a=group:BUNDLE video\n" - "a=msid-semantic: WMS\n" - "m=video 9 RTP/SAVPF 96\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:1ZFN\n" - "a=ice-pwd:70P3H0jPlGz1fiJl5XZfXMZH\n" - "a=ice-options:trickle\n" - "a=fingerprint:sha-256 3E:10:35:6B:9A:9E:B0:55:AC:2A:88:F5:74:C1:70:32:B5:8D:88:1D:37:B0:9C:69:A6:DD:07:10:73:27:1A:16\n" - "a=setup:active\n" - "a=mid:video\n" - "a=recvonly\n" - "a=rtcp-mux\n" - "a=rtpmap:96 H264/90000\n" - "a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f"; - char str2[] = "v=0\n" - "o=- 2584450093346841581 2 IN IP4 127.0.0.1\n" - "s=-\n" - "t=0 0\n" - "a=group:BUNDLE audio video data\n" - "a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:audio\n" - "a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" - "a=sendrecv\n" - "a=rtcp-mux\n" - "a=rtpmap:111 opus/48000/2\n" - "a=rtcp-fb:111 transport-cc\n" - "a=fmtp:111 minptime=10;useinbandfec=1\n" - "a=rtpmap:103 ISAC/16000\n" - "a=rtpmap:104 ISAC/32000\n" - "a=rtpmap:9 G722/8000\n" - "a=rtpmap:0 PCMU/8000\n" - "a=rtpmap:8 PCMA/8000\n" - "a=rtpmap:106 CN/32000\n" - "a=rtpmap:105 CN/16000\n" - "a=rtpmap:13 CN/8000\n" - "a=rtpmap:110 telephone-event/48000\n" - "a=rtpmap:112 telephone-event/32000\n" - "a=rtpmap:113 telephone-event/16000\n" - "a=rtpmap:126 telephone-event/8000\n" - "a=ssrc:120276603 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363\n" - "a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363\n" - "m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:video\n" - "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n" - "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\n" - "a=extmap:4 urn:3gpp:video-orientation\n" - "a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\n" - "a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\n" - "a=sendrecv\n" - "a=rtcp-mux\n" - "a=rtcp-rsize\n" - "a=rtpmap:96 VP8/90000\n" - "a=rtcp-fb:96 ccm fir\n" - "a=rtcp-fb:96 nack\n" - "a=rtcp-fb:96 nack pli\n" - "a=rtcp-fb:96 goog-remb\n" - "a=rtcp-fb:96 transport-cc\n" - "a=rtpmap:98 VP9/90000\n" - "a=rtcp-fb:98 ccm fir\n" - "a=rtcp-fb:98 nack\n" - "a=rtcp-fb:98 nack pli\n" - "a=rtcp-fb:98 goog-remb\n" - "a=rtcp-fb:98 transport-cc\n" - "a=rtpmap:100 H264/90000\n" - "a=rtcp-fb:100 ccm fir\n" - "a=rtcp-fb:100 nack\n" - "a=rtcp-fb:100 nack pli\n" - "a=rtcp-fb:100 goog-remb\n" - "a=rtcp-fb:100 transport-cc\n" - "a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\n" - "a=rtpmap:102 red/90000\n" - "a=rtpmap:127 ulpfec/90000\n" - "a=rtpmap:97 rtx/90000\n" - "a=fmtp:97 apt=96\n" - "a=rtpmap:99 rtx/90000\n" - "a=fmtp:99 apt=98\n" - "a=rtpmap:101 rtx/90000\n" - "a=fmtp:101 apt=100\n" - "a=rtpmap:125 rtx/90000\n" - "a=fmtp:125 apt=102\n" - "a=ssrc-group:FID 2580761338 611523443\n" - "a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:611523443 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=candidate:3575467457 1 udp 2113937151 10.15.83.23 57857 typ host generation 0 ufrag 6R0z network-cost 999\n" - "m=application 9 DTLS/SCTP 5000\n" - "c=IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:data\n" - "a=sctpmap:5000 webrtc-datachannel 1024\n" - "a=sctp-port:5000"; - - RtcSessionSdp sdp1; - sdp1.parse(str1); - - RtcSessionSdp sdp2; - sdp2.parse(str2); - - for (auto media : sdp1.medias) { - InfoL << getRtpDirectionString(media.getDirection()); - } - for (auto media : sdp2.medias) { - InfoL << getRtpDirectionString(media.getDirection()); - } - InfoL << sdp1.toString(); - InfoL << sdp2.toString(); - - RtcSession session1; - session1.loadFrom(str1); - - RtcSession session2; - session2.loadFrom(str2); - DebugL << session1.toString(); - DebugL << session2.toString(); -} - void RtcSession::loadFrom(const string &str) { RtcSessionSdp sdp; sdp.parse(str); @@ -1294,7 +1148,6 @@ void RtcConfigure::setDefaultSetting(string ice_ufrag, DtlsRole role, RtpDirection direction, const SdpAttrFingerprint &fingerprint) { - video.setDefaultSetting(TrackVideo); audio.setDefaultSetting(TrackAudio); application.setDefaultSetting(TrackApplication); @@ -1321,9 +1174,15 @@ void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType typ break; } default: { - audio.candidate.emplace_back(candidate); - video.candidate.emplace_back(candidate); - application.candidate.emplace_back(candidate); + if (audio.group_bundle) { + audio.candidate.emplace_back(candidate); + } + if (video.group_bundle) { + video.candidate.emplace_back(candidate); + } + if (application.group_bundle) { + application.candidate.emplace_back(candidate); + } break; } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 8a373d82..bfdf01d4 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -431,6 +431,7 @@ public: //a=candidate:4 1 udp 2 192.168.1.7 58107 typ host //a=candidate:
typ string foundation; + //传输媒体的类型,1代表RTP;2代表 RTCP。 uint32_t component; string transport {"udp"}; uint32_t priority; From 978917b4657e31fd04c05d5a623106c54e57ef17 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 30 Mar 2021 18:34:17 +0800 Subject: [PATCH 034/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E6=8E=A5=E5=8F=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 15 +++++++++++++++ webrtc/Sdp.h | 3 +++ 2 files changed, 18 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index a6b548fd..c8cd55ac 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1186,4 +1186,19 @@ void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType typ break; } } +} + +shared_ptr RtcConfigure::createOffer(){ + shared_ptr ret = std::make_shared(); + ret->version = 0; + ret->origin.parse("- 0 0 IN IP4 0.0.0.0"); + ret->session_name = "zlmediakit_webrtc_session"; + ret->session_info = "zlmediakit_webrtc_session"; + ret->connection.parse("IN IP4 0.0.0.0"); + ret->msid_semantic.parse("WMS *"); + return nullptr; +} + +shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ + return nullptr; } \ No newline at end of file diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index bfdf01d4..d6860930 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -669,6 +669,9 @@ public: RtpDirection direction, const SdpAttrFingerprint &fingerprint); void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid); + + shared_ptr createOffer(); + shared_ptr createAnswer(const RtcSession &offer); }; From 7e5cb3339531594980d7152432a98e8c4f89aa01 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 31 Mar 2021 17:15:26 +0800 Subject: [PATCH 035/218] =?UTF-8?q?=E5=BC=80=E5=A7=8B=E5=AF=B9=E6=8E=A5js?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 2 +- server/WebApi.cpp | 35 +- server/WebApi.h | 17 + server/main.cpp | 149 + webrtc/Sdp.cpp | 5 +- webrtc/Sdp.h | 25 + webrtc/WebRtcTransport.cpp | 12 +- webrtc/WebRtcTransport.h | 9 +- www/webrtc/ZLMRTCClient.js | 7466 ++++++++++++++++++++++++++++++++ www/webrtc/ZLMRTCClient.js.map | 1 + www/webrtc/index.html | 127 +- www/webrtc/js/adapter.js | 2475 ----------- www/webrtc/js/common.js | 12 - www/webrtc/js/main.js | 134 - 14 files changed, 7807 insertions(+), 2662 deletions(-) create mode 100644 www/webrtc/ZLMRTCClient.js create mode 100644 www/webrtc/ZLMRTCClient.js.map delete mode 100644 www/webrtc/js/adapter.js delete mode 100644 www/webrtc/js/common.js delete mode 100644 www/webrtc/js/main.js diff --git a/conf/config.ini b/conf/config.ini index a2d66b04..46d57ce3 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -136,7 +136,7 @@ charSet=utf-8 #http链接超时时间 keepAliveSecond=30 #http请求体最大字节数，如果post的body太大，则不适合缓存body在内存 -maxReqSize=4096 +maxReqSize=40960 #404网页内容，用户可以自定义404网页 notFound=404 Not Found

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ZLMediaKit-4.0
#http服务器监听端口 diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 055a8f0f..22c2feea 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -127,6 +127,26 @@ static HttpApi toApi(const function &cb) { }); } +static HttpApi toApi(const function &cb) { + return [cb](const Parser &parser, const HttpSession::HttpResponseInvoker &invoker, SockInfo &sender) { + GET_CONFIG(string, charSet, Http::kCharSet); + HttpSession::KeyValue headerOut; + headerOut["Content-Type"] = string("application/json; charset=") + charSet; + + Json::Value val; + val["code"] = API::Success; + + cb(sender, parser.getHeader(), headerOut, parser, val, invoker); + }; +} + +static HttpApi toApi(const function &cb) { + return toApi([cb](API_ARGS_STRING_ASYNC) { + cb(API_ARGS_VALUE); + invoker(200, headerOut, val.toStyledString()); + }); +} + void api_regist(const string &api_path, const function &func) { s_map_api.emplace(api_path, toApi(func)); } @@ -143,6 +163,14 @@ void api_regist(const string &api_path, const function &func){ + s_map_api.emplace(api_path, toApi(func)); +} + +void api_regist(const string &api_path, const function &func){ + s_map_api.emplace(api_path, toApi(func)); +} + //获取HTTP请求中url参数、content参数 static ApiArgsType getAllArgs(const Parser &parser) { ApiArgsType allArgs; @@ -1054,9 +1082,9 @@ void installWebApi() { #ifdef ENABLE_WEBRTC static list rtcs; - api_regist("/webrtc",[](API_ARGS_MAP_ASYNC){ + api_regist("/index/api/webrtc",[](API_ARGS_STRING){ CHECK_ARGS("app", "stream"); - auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, DEFAULT_VHOST, allArgs["app"], allArgs["stream"])); + auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, DEFAULT_VHOST, allArgs.getUrlArgs()["app"], allArgs.getUrlArgs()["stream"])); if (!src) { throw ApiRetException("流不存在", API::NotFound); } @@ -1064,7 +1092,8 @@ void installWebApi() { headerOut["Access-Control-Allow-Origin"] = "*"; auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); rtc->attach(src); - invoker(200, headerOut, rtc->GetLocalSdp()); + val["sdp"] = rtc->getAnswerSdp(allArgs.Content()); + val["type"] = "answer"; rtcs.emplace_back(rtc); }); #endif diff --git a/server/WebApi.h b/server/WebApi.h index 286fde41..1fa1d17c 100755 --- a/server/WebApi.h +++ b/server/WebApi.h @@ -85,6 +85,8 @@ using ApiArgsType = map; #define API_ARGS_MAP_ASYNC API_ARGS_MAP, const HttpSession::HttpResponseInvoker &invoker #define API_ARGS_JSON SockInfo &sender, HttpSession::KeyValue &headerIn, HttpSession::KeyValue &headerOut, Json::Value &allArgs, Json::Value &val #define API_ARGS_JSON_ASYNC API_ARGS_JSON, const HttpSession::HttpResponseInvoker &invoker +#define API_ARGS_STRING SockInfo &sender, HttpSession::KeyValue &headerIn, HttpSession::KeyValue &headerOut, const Parser &allArgs, Json::Value &val +#define API_ARGS_STRING_ASYNC API_ARGS_STRING, const HttpSession::HttpResponseInvoker &invoker #define API_ARGS_VALUE sender, headerIn, headerOut, allArgs, val //注册http请求参数是map类型的http api @@ -97,6 +99,11 @@ void api_regist(const string &api_path, const function &fun //注册http请求参数是Json::Value类型，但是可以异步回复的的http api void api_regist(const string &api_path, const function &func); +//注册http请求参数是http原始请求信息的http api +void api_regist(const string &api_path, const function &func); +//注册http请求参数是http原始请求信息的异步回复的http api +void api_regist(const string &api_path, const function &func); + template bool checkArgs(Args &&args, First &&first) { return !args[first].empty(); @@ -107,6 +114,16 @@ bool checkArgs(Args &&args, First &&first, KeyTypes &&...keys) { return !args[first].empty() && checkArgs(std::forward(args), std::forward(keys)...); } +template +bool checkArgs(const Parser &args, First &&first) { + return !args.getUrlArgs()[first].empty(); +} + +template +bool checkArgs(const Parser &args, First &&first, KeyTypes &&...keys) { + return !args.getUrlArgs()[first].empty() && checkArgs(args, std::forward(keys)...); +} + //检查http参数是否为空的宏 #define CHECK_ARGS(...) \ if(!checkArgs(allArgs,##__VA_ARGS__)){ \ diff --git a/server/main.cpp b/server/main.cpp index 75264ff3..fdba471d 100644 --- a/server/main.cpp +++ b/server/main.cpp @@ -25,6 +25,7 @@ #include "Rtp/RtpServer.h" #include "WebApi.h" #include "WebHook.h" +#include "../webrtc/Sdp.h" #if defined(ENABLE_VERSION) #include "Version.h" @@ -209,6 +210,153 @@ static void inline listen_shell_input(){ //全局变量，在WebApi中用于保存配置文件用 string g_ini_file; + +void test_sdp(){ + char str1[] = "v=0\n" + "o=- 380154348540553537 2 IN IP4 127.0.0.1\n" + "s=-\n" + "b=CT:1900\n" + "t=0 0\n" + "a=group:BUNDLE video\n" + "a=msid-semantic: WMS\n" + "m=video 9 RTP/SAVPF 96\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:1ZFN\n" + "a=ice-pwd:70P3H0jPlGz1fiJl5XZfXMZH\n" + "a=ice-options:trickle\n" + "a=fingerprint:sha-256 3E:10:35:6B:9A:9E:B0:55:AC:2A:88:F5:74:C1:70:32:B5:8D:88:1D:37:B0:9C:69:A6:DD:07:10:73:27:1A:16\n" + "a=setup:active\n" + "a=mid:video\n" + "a=recvonly\n" + "a=rtcp-mux\n" + "a=rtpmap:96 H264/90000\n" + "a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f"; + char str2[] = "v=0\n" + "o=- 2584450093346841581 2 IN IP4 127.0.0.1\n" + "s=-\n" + "t=0 0\n" + "a=group:BUNDLE audio video data\n" + "a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:audio\n" + "a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" + "a=sendrecv\n" + "a=rtcp-mux\n" + "a=rtpmap:111 opus/48000/2\n" + "a=rtcp-fb:111 transport-cc\n" + "a=fmtp:111 minptime=10;useinbandfec=1\n" + "a=rtpmap:103 ISAC/16000\n" + "a=rtpmap:104 ISAC/32000\n" + "a=rtpmap:9 G722/8000\n" + "a=rtpmap:0 PCMU/8000\n" + "a=rtpmap:8 PCMA/8000\n" + "a=rtpmap:106 CN/32000\n" + "a=rtpmap:105 CN/16000\n" + "a=rtpmap:13 CN/8000\n" + "a=rtpmap:110 telephone-event/48000\n" + "a=rtpmap:112 telephone-event/32000\n" + "a=rtpmap:113 telephone-event/16000\n" + "a=rtpmap:126 telephone-event/8000\n" + "a=ssrc:120276603 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363\n" + "a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363\n" + "m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125\n" + "c=IN IP4 0.0.0.0\n" + "a=rtcp:9 IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:video\n" + "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n" + "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\n" + "a=extmap:4 urn:3gpp:video-orientation\n" + "a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\n" + "a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\n" + "a=sendrecv\n" + "a=rtcp-mux\n" + "a=rtcp-rsize\n" + "a=rtpmap:96 VP8/90000\n" + "a=rtcp-fb:96 ccm fir\n" + "a=rtcp-fb:96 nack\n" + "a=rtcp-fb:96 nack pli\n" + "a=rtcp-fb:96 goog-remb\n" + "a=rtcp-fb:96 transport-cc\n" + "a=rtpmap:98 VP9/90000\n" + "a=rtcp-fb:98 ccm fir\n" + "a=rtcp-fb:98 nack\n" + "a=rtcp-fb:98 nack pli\n" + "a=rtcp-fb:98 goog-remb\n" + "a=rtcp-fb:98 transport-cc\n" + "a=rtpmap:100 H264/90000\n" + "a=rtcp-fb:100 ccm fir\n" + "a=rtcp-fb:100 nack\n" + "a=rtcp-fb:100 nack pli\n" + "a=rtcp-fb:100 goog-remb\n" + "a=rtcp-fb:100 transport-cc\n" + "a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\n" + "a=rtpmap:102 red/90000\n" + "a=rtpmap:127 ulpfec/90000\n" + "a=rtpmap:97 rtx/90000\n" + "a=fmtp:97 apt=96\n" + "a=rtpmap:99 rtx/90000\n" + "a=fmtp:99 apt=98\n" + "a=rtpmap:101 rtx/90000\n" + "a=fmtp:101 apt=100\n" + "a=rtpmap:125 rtx/90000\n" + "a=fmtp:125 apt=102\n" + "a=ssrc-group:FID 2580761338 611523443\n" + "a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:611523443 cname:iSkJ2vn5cYYubTve\n" + "a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" + "a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n" + "a=candidate:3575467457 1 udp 2113937151 10.15.83.23 57857 typ host generation 0 ufrag 6R0z network-cost 999\n" + "m=application 9 DTLS/SCTP 5000\n" + "c=IN IP4 0.0.0.0\n" + "a=ice-ufrag:sXJ3\n" + "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" + "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" + "a=setup:actpass\n" + "a=mid:data\n" + "a=sctpmap:5000 webrtc-datachannel 1024\n" + "a=sctp-port:5000"; + + RtcSessionSdp sdp1; + sdp1.parse(str1); + + RtcSessionSdp sdp2; + sdp2.parse(str2); + + for (auto media : sdp1.medias) { + InfoL << getRtpDirectionString(media.getDirection()); + } + for (auto media : sdp2.medias) { + InfoL << getRtpDirectionString(media.getDirection()); + } + InfoL << sdp1.toString(); + InfoL << sdp2.toString(); + + RtcSession session1; + session1.loadFrom(str1); + + RtcSession session2; + session2.loadFrom(str2); + DebugL << session1.toString(); + DebugL << session2.toString(); +} + int start_main(int argc,char *argv[]) { { CMD_main cmd_main; @@ -266,6 +414,7 @@ int start_main(int argc,char *argv[]) { }); } +// test_sdp(); uint16_t shellPort = mINI::Instance()[Shell::kPort]; uint16_t rtspPort = mINI::Instance()[Rtsp::kPort]; uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort]; diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index c8cd55ac..5f40da33 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -101,7 +101,10 @@ static bool registerAllItem(){ registerSdpItem(); registerSdpItem(); registerSdpItem(); - + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); + registerSdpItem(); return true; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index d6860930..a3ef62bc 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -445,6 +445,28 @@ public: const char* getKey() const override { return "candidate";} }; +class SdpAttrMsid : public SdpItem{ +public: + const char* getKey() const override { return "msid";} +}; + +class SdpAttrExtmapAllowMixed : public SdpItem{ +public: + const char* getKey() const override { return "extmap-allow-mixed";} +}; + +class SdpAttrSimulcast : public SdpItem{ +public: + //todo + const char* getKey() const override { return "simulcast";} +}; + +class SdpAttrRid : public SdpItem{ +public: + //todo + const char* getKey() const override { return "rid";} +}; + class RtcSdpBase { public: vector items; @@ -575,6 +597,7 @@ public: RtcSSRC rtx_ssrc; //////// simulcast //////// + bool simulcast{false}; RtcSSRC rtp_ssrc_low; RtcSSRC rtp_ssrc_mid; RtcSSRC rtp_ssrc_high; @@ -611,6 +634,8 @@ public: class RtcSession{ public: + using Ptr = std::shared_ptr; + uint32_t version; SdpOrigin origin; string session_name; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index c32e6850..c96c1e73 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -62,7 +62,17 @@ void WebRtcTransport::onWrite(const char *buf, size_t len){ onWrite(buf, len, (struct sockaddr_in *)tuple); } -std::string WebRtcTransport::GetLocalSdp() { +std::string WebRtcTransport::getAnswerSdp(const string &offer){ + InfoL << offer; + _offer_sdp = std::make_shared(); + _offer_sdp->loadFrom(offer); + + RtcConfigure configure; + _answer_sdp = configure.createAnswer(*_offer_sdp); + return _answer_sdp->toString(); +} + +std::string WebRtcTransport::getOfferSdp() { RTC::DtlsTransport::Fingerprint remote_fingerprint; remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm("sha-256"); remote_fingerprint.value = ""; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 2842bee8..29ca7c86 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -7,6 +7,7 @@ #include "IceServer.hpp" #include "SrtpSession.hpp" #include "StunPacket.hpp" +#include "Sdp.h" class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: @@ -17,9 +18,9 @@ public: /// 销毁对象 virtual void onDestory(); - /// 获取本地sdp - /// \return - std::string GetLocalSdp(); + std::string getAnswerSdp(const string &offer); + + std::string getOfferSdp(); /// 收到udp数据 /// \param buf @@ -76,6 +77,8 @@ private: std::shared_ptr ice_server_; std::shared_ptr dtls_transport_; std::shared_ptr srtp_session_; + RtcSession::Ptr _offer_sdp; + RtcSession::Ptr _answer_sdp; }; #include "Poller/EventPoller.h" diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js new file mode 100644 index 00000000..233ed60b --- /dev/null +++ b/www/webrtc/ZLMRTCClient.js @@ -0,0 +1,7466 @@ +var ZLMRTCClient = (function (exports) { + 'use strict'; + + const Events$1 = { + WEBRTC_NOT_SUPPORT: 'WEBRTC_NOT_SUPPORT', + WEBRTC_ICE_CANDIDATE_ERROR: 'WEBRTC_ICE_CANDIDATE_ERROR', + WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED: 'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED', + WEBRTC_ON_REMOTE_STREAMS: 'WEBRTC_ON_REMOTE_STREAMS', + WEBRTC_ON_LOCAL_STREAM: 'WEBRTC_ON_LOCAL_STREAM' + }; + + const VERSION = '1.0.1'; + const BUILD_DATE = 'Wed Mar 31 2021 14:49:19 GMT+0800 (China Standard Time)'; + + // Copyright (C) <2018> Intel Corporation + // + // SPDX-License-Identifier: Apache-2.0 + // eslint-disable-next-line require-jsdoc + function isFirefox() { + return window.navigator.userAgent.match('Firefox') !== null; + } // eslint-disable-next-line require-jsdoc + + function isChrome() { + return window.navigator.userAgent.match('Chrome') !== null; + } // eslint-disable-next-line require-jsdoc + + function isEdge() { + return window.navigator.userAgent.match(/Edge\/(\d+).(\d+)$/) !== null; + } // eslint-disable-next-line require-jsdoc + + // Copyright (C) <2018> Intel Corporation + /** + * @class AudioSourceInfo + * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'. + * @memberOf Owt.Base + * @readonly + * @enum {string} + */ + + const AudioSourceInfo = { + MIC: 'mic', + SCREENCAST: 'screen-cast', + FILE: 'file', + MIXED: 'mixed' + }; + /** + * @class VideoSourceInfo + * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'. + * @memberOf Owt.Base + * @readonly + * @enum {string} + */ + + const VideoSourceInfo = { + CAMERA: 'camera', + SCREENCAST: 'screen-cast', + FILE: 'file', + MIXED: 'mixed' + }; + /** + * @class TrackKind + * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks. + * @memberOf Owt.Base + * @readonly + * @enum {string} + */ + + const TrackKind = { + /** + * Audio tracks. + * @type string + */ + AUDIO: 'audio', + + /** + * Video tracks. + * @type string + */ + VIDEO: 'video', + + /** + * Both audio and video tracks. + * @type string + */ + AUDIO_AND_VIDEO: 'av' + }; + /** + * @class Resolution + * @memberOf Owt.Base + * @classDesc The Resolution defines the size of a rectangle. + * @constructor + * @param {number} width + * @param {number} height + */ + + class Resolution { + // eslint-disable-next-line require-jsdoc + constructor(width, height) { + /** + * @member {number} width + * @instance + * @memberof Owt.Base.Resolution + */ + this.width = width; + /** + * @member {number} height + * @instance + * @memberof Owt.Base.Resolution + */ + + this.height = height; + } + + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + let logDisabled_ = true; + let deprecationWarnings_ = true; + + /** + * Extract browser version out of the provided user agent string. + * + * @param {!string} uastring userAgent string. + * @param {!string} expr Regular expression used as match criteria. + * @param {!number} pos position in the version string to be returned. + * @return {!number} browser version. + */ + function extractVersion(uastring, expr, pos) { + const match = uastring.match(expr); + return match && match.length >= pos && parseInt(match[pos], 10); + } + + // Wraps the peerconnection event eventNameToWrap in a function + // which returns the modified event object (or false to prevent + // the event). + function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) { + if (!window.RTCPeerConnection) { + return; + } + const proto = window.RTCPeerConnection.prototype; + const nativeAddEventListener = proto.addEventListener; + proto.addEventListener = function(nativeEventName, cb) { + if (nativeEventName !== eventNameToWrap) { + return nativeAddEventListener.apply(this, arguments); + } + const wrappedCallback = (e) => { + const modifiedEvent = wrapper(e); + if (modifiedEvent) { + if (cb.handleEvent) { + cb.handleEvent(modifiedEvent); + } else { + cb(modifiedEvent); + } + } + }; + this._eventMap = this._eventMap || {}; + if (!this._eventMap[eventNameToWrap]) { + this._eventMap[eventNameToWrap] = new Map(); + } + this._eventMap[eventNameToWrap].set(cb, wrappedCallback); + return nativeAddEventListener.apply(this, [nativeEventName, + wrappedCallback]); + }; + + const nativeRemoveEventListener = proto.removeEventListener; + proto.removeEventListener = function(nativeEventName, cb) { + if (nativeEventName !== eventNameToWrap || !this._eventMap + || !this._eventMap[eventNameToWrap]) { + return nativeRemoveEventListener.apply(this, arguments); + } + if (!this._eventMap[eventNameToWrap].has(cb)) { + return nativeRemoveEventListener.apply(this, arguments); + } + const unwrappedCb = this._eventMap[eventNameToWrap].get(cb); + this._eventMap[eventNameToWrap].delete(cb); + if (this._eventMap[eventNameToWrap].size === 0) { + delete this._eventMap[eventNameToWrap]; + } + if (Object.keys(this._eventMap).length === 0) { + delete this._eventMap; + } + return nativeRemoveEventListener.apply(this, [nativeEventName, + unwrappedCb]); + }; + + Object.defineProperty(proto, 'on' + eventNameToWrap, { + get() { + return this['_on' + eventNameToWrap]; + }, + set(cb) { + if (this['_on' + eventNameToWrap]) { + this.removeEventListener(eventNameToWrap, + this['_on' + eventNameToWrap]); + delete this['_on' + eventNameToWrap]; + } + if (cb) { + this.addEventListener(eventNameToWrap, + this['_on' + eventNameToWrap] = cb); + } + }, + enumerable: true, + configurable: true + }); + } + + function disableLog(bool) { + if (typeof bool !== 'boolean') { + return new Error('Argument type: ' + typeof bool + + '. Please use a boolean.'); + } + logDisabled_ = bool; + return (bool) ? 'adapter.js logging disabled' : + 'adapter.js logging enabled'; + } + + /** + * Disable or enable deprecation warnings + * @param {!boolean} bool set to true to disable warnings. + */ + function disableWarnings(bool) { + if (typeof bool !== 'boolean') { + return new Error('Argument type: ' + typeof bool + + '. Please use a boolean.'); + } + deprecationWarnings_ = !bool; + return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled'); + } + + function log$1() { + if (typeof window === 'object') { + if (logDisabled_) { + return; + } + if (typeof console !== 'undefined' && typeof console.log === 'function') { + console.log.apply(console, arguments); + } + } + } + + /** + * Shows a deprecation warning suggesting the modern and spec-compatible API. + */ + function deprecated(oldMethod, newMethod) { + if (!deprecationWarnings_) { + return; + } + console.warn(oldMethod + ' is deprecated, please use ' + newMethod + + ' instead.'); + } + + /** + * Browser detector. + * + * @return {object} result containing browser and version + * properties. + */ + function detectBrowser(window) { + // Returned result object. + const result = {browser: null, version: null}; + + // Fail early if it's not a browser + if (typeof window === 'undefined' || !window.navigator) { + result.browser = 'Not a browser.'; + return result; + } + + const {navigator} = window; + + if (navigator.mozGetUserMedia) { // Firefox. + result.browser = 'firefox'; + result.version = extractVersion(navigator.userAgent, + /Firefox\/(\d+)\./, 1); + } else if (navigator.webkitGetUserMedia || + (window.isSecureContext === false && window.webkitRTCPeerConnection && + !window.RTCIceGatherer)) { + // Chrome, Chromium, Webview, Opera. + // Version matches Chrome/WebRTC version. + // Chrome 74 removed webkitGetUserMedia on http as well so we need the + // more complicated fallback to webkitRTCPeerConnection. + result.browser = 'chrome'; + result.version = extractVersion(navigator.userAgent, + /Chrom(e|ium)\/(\d+)\./, 2); + } else if (navigator.mediaDevices && + navigator.userAgent.match(/Edge\/(\d+).(\d+)$/)) { // Edge. + result.browser = 'edge'; + result.version = extractVersion(navigator.userAgent, + /Edge\/(\d+).(\d+)$/, 2); + } else if (window.RTCPeerConnection && + navigator.userAgent.match(/AppleWebKit\/(\d+)\./)) { // Safari. + result.browser = 'safari'; + result.version = extractVersion(navigator.userAgent, + /AppleWebKit\/(\d+)\./, 1); + result.supportsUnifiedPlan = window.RTCRtpTransceiver && + 'currentDirection' in window.RTCRtpTransceiver.prototype; + } else { // Default fallthrough: not supported. + result.browser = 'Not a supported browser.'; + return result; + } + + return result; + } + + /** + * Checks if something is an object. + * + * @param {*} val The something you want to check. + * @return true if val is an object, false otherwise. + */ + function isObject$1(val) { + return Object.prototype.toString.call(val) === '[object Object]'; + } + + /** + * Remove all empty objects and undefined values + * from a nested object -- an enhanced and vanilla version + * of Lodash's `compact`. + */ + function compactObject(data) { + if (!isObject$1(data)) { + return data; + } + + return Object.keys(data).reduce(function(accumulator, key) { + const isObj = isObject$1(data[key]); + const value = isObj ? compactObject(data[key]) : data[key]; + const isEmptyObject = isObj && !Object.keys(value).length; + if (value === undefined || isEmptyObject) { + return accumulator; + } + return Object.assign(accumulator, {[key]: value}); + }, {}); + } + + /* iterates the stats graph recursively. */ + function walkStats(stats, base, resultSet) { + if (!base || resultSet.has(base.id)) { + return; + } + resultSet.set(base.id, base); + Object.keys(base).forEach(name => { + if (name.endsWith('Id')) { + walkStats(stats, stats.get(base[name]), resultSet); + } else if (name.endsWith('Ids')) { + base[name].forEach(id => { + walkStats(stats, stats.get(id), resultSet); + }); + } + }); + } + + /* filter getStats for a sender/receiver track. */ + function filterStats(result, track, outbound) { + const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp'; + const filteredResult = new Map(); + if (track === null) { + return filteredResult; + } + const trackStats = []; + result.forEach(value => { + if (value.type === 'track' && + value.trackIdentifier === track.id) { + trackStats.push(value); + } + }); + trackStats.forEach(trackStat => { + result.forEach(stats => { + if (stats.type === streamStatsType && stats.trackId === trackStat.id) { + walkStats(result, stats, filteredResult); + } + }); + }); + return filteredResult; + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + const logging = log$1; + + function shimGetUserMedia$3(window, browserDetails) { + const navigator = window && window.navigator; + + if (!navigator.mediaDevices) { + return; + } + + const constraintsToChrome_ = function(c) { + if (typeof c !== 'object' || c.mandatory || c.optional) { + return c; + } + const cc = {}; + Object.keys(c).forEach(key => { + if (key === 'require' || key === 'advanced' || key === 'mediaSource') { + return; + } + const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]}; + if (r.exact !== undefined && typeof r.exact === 'number') { + r.min = r.max = r.exact; + } + const oldname_ = function(prefix, name) { + if (prefix) { + return prefix + name.charAt(0).toUpperCase() + name.slice(1); + } + return (name === 'deviceId') ? 'sourceId' : name; + }; + if (r.ideal !== undefined) { + cc.optional = cc.optional || []; + let oc = {}; + if (typeof r.ideal === 'number') { + oc[oldname_('min', key)] = r.ideal; + cc.optional.push(oc); + oc = {}; + oc[oldname_('max', key)] = r.ideal; + cc.optional.push(oc); + } else { + oc[oldname_('', key)] = r.ideal; + cc.optional.push(oc); + } + } + if (r.exact !== undefined && typeof r.exact !== 'number') { + cc.mandatory = cc.mandatory || {}; + cc.mandatory[oldname_('', key)] = r.exact; + } else { + ['min', 'max'].forEach(mix => { + if (r[mix] !== undefined) { + cc.mandatory = cc.mandatory || {}; + cc.mandatory[oldname_(mix, key)] = r[mix]; + } + }); + } + }); + if (c.advanced) { + cc.optional = (cc.optional || []).concat(c.advanced); + } + return cc; + }; + + const shimConstraints_ = function(constraints, func) { + if (browserDetails.version >= 61) { + return func(constraints); + } + constraints = JSON.parse(JSON.stringify(constraints)); + if (constraints && typeof constraints.audio === 'object') { + const remap = function(obj, a, b) { + if (a in obj && !(b in obj)) { + obj[b] = obj[a]; + delete obj[a]; + } + }; + constraints = JSON.parse(JSON.stringify(constraints)); + remap(constraints.audio, 'autoGainControl', 'googAutoGainControl'); + remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression'); + constraints.audio = constraintsToChrome_(constraints.audio); + } + if (constraints && typeof constraints.video === 'object') { + // Shim facingMode for mobile & surface pro. + let face = constraints.video.facingMode; + face = face && ((typeof face === 'object') ? face : {ideal: face}); + const getSupportedFacingModeLies = browserDetails.version < 66; + + if ((face && (face.exact === 'user' || face.exact === 'environment' || + face.ideal === 'user' || face.ideal === 'environment')) && + !(navigator.mediaDevices.getSupportedConstraints && + navigator.mediaDevices.getSupportedConstraints().facingMode && + !getSupportedFacingModeLies)) { + delete constraints.video.facingMode; + let matches; + if (face.exact === 'environment' || face.ideal === 'environment') { + matches = ['back', 'rear']; + } else if (face.exact === 'user' || face.ideal === 'user') { + matches = ['front']; + } + if (matches) { + // Look for matches in label, or use last cam for back (typical). + return navigator.mediaDevices.enumerateDevices() + .then(devices => { + devices = devices.filter(d => d.kind === 'videoinput'); + let dev = devices.find(d => matches.some(match => + d.label.toLowerCase().includes(match))); + if (!dev && devices.length && matches.includes('back')) { + dev = devices[devices.length - 1]; // more likely the back cam + } + if (dev) { + constraints.video.deviceId = face.exact ? {exact: dev.deviceId} : + {ideal: dev.deviceId}; + } + constraints.video = constraintsToChrome_(constraints.video); + logging('chrome: ' + JSON.stringify(constraints)); + return func(constraints); + }); + } + } + constraints.video = constraintsToChrome_(constraints.video); + } + logging('chrome: ' + JSON.stringify(constraints)); + return func(constraints); + }; + + const shimError_ = function(e) { + if (browserDetails.version >= 64) { + return e; + } + return { + name: { + PermissionDeniedError: 'NotAllowedError', + PermissionDismissedError: 'NotAllowedError', + InvalidStateError: 'NotAllowedError', + DevicesNotFoundError: 'NotFoundError', + ConstraintNotSatisfiedError: 'OverconstrainedError', + TrackStartError: 'NotReadableError', + MediaDeviceFailedDueToShutdown: 'NotAllowedError', + MediaDeviceKillSwitchOn: 'NotAllowedError', + TabCaptureError: 'AbortError', + ScreenCaptureError: 'AbortError', + DeviceCaptureError: 'AbortError' + }[e.name] || e.name, + message: e.message, + constraint: e.constraint || e.constraintName, + toString() { + return this.name + (this.message && ': ') + this.message; + } + }; + }; + + const getUserMedia_ = function(constraints, onSuccess, onError) { + shimConstraints_(constraints, c => { + navigator.webkitGetUserMedia(c, onSuccess, e => { + if (onError) { + onError(shimError_(e)); + } + }); + }); + }; + navigator.getUserMedia = getUserMedia_.bind(navigator); + + // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia + // function which returns a Promise, it does not accept spec-style + // constraints. + if (navigator.mediaDevices.getUserMedia) { + const origGetUserMedia = navigator.mediaDevices.getUserMedia. + bind(navigator.mediaDevices); + navigator.mediaDevices.getUserMedia = function(cs) { + return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => { + if (c.audio && !stream.getAudioTracks().length || + c.video && !stream.getVideoTracks().length) { + stream.getTracks().forEach(track => { + track.stop(); + }); + throw new DOMException('', 'NotFoundError'); + } + return stream; + }, e => Promise.reject(shimError_(e)))); + }; + } + } + + /* + * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + function shimGetDisplayMedia$2(window, getSourceId) { + if (window.navigator.mediaDevices && + 'getDisplayMedia' in window.navigator.mediaDevices) { + return; + } + if (!(window.navigator.mediaDevices)) { + return; + } + // getSourceId is a function that returns a promise resolving with + // the sourceId of the screen/window/tab to be shared. + if (typeof getSourceId !== 'function') { + console.error('shimGetDisplayMedia: getSourceId argument is not ' + + 'a function'); + return; + } + window.navigator.mediaDevices.getDisplayMedia = + function getDisplayMedia(constraints) { + return getSourceId(constraints) + .then(sourceId => { + const widthSpecified = constraints.video && constraints.video.width; + const heightSpecified = constraints.video && + constraints.video.height; + const frameRateSpecified = constraints.video && + constraints.video.frameRate; + constraints.video = { + mandatory: { + chromeMediaSource: 'desktop', + chromeMediaSourceId: sourceId, + maxFrameRate: frameRateSpecified || 3 + } + }; + if (widthSpecified) { + constraints.video.mandatory.maxWidth = widthSpecified; + } + if (heightSpecified) { + constraints.video.mandatory.maxHeight = heightSpecified; + } + return window.navigator.mediaDevices.getUserMedia(constraints); + }); + }; + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimMediaStream(window) { + window.MediaStream = window.MediaStream || window.webkitMediaStream; + } + + function shimOnTrack$1(window) { + if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in + window.RTCPeerConnection.prototype)) { + Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', { + get() { + return this._ontrack; + }, + set(f) { + if (this._ontrack) { + this.removeEventListener('track', this._ontrack); + } + this.addEventListener('track', this._ontrack = f); + }, + enumerable: true, + configurable: true + }); + const origSetRemoteDescription = + window.RTCPeerConnection.prototype.setRemoteDescription; + window.RTCPeerConnection.prototype.setRemoteDescription = + function setRemoteDescription() { + if (!this._ontrackpoly) { + this._ontrackpoly = (e) => { + // onaddstream does not fire when a track is added to an existing + // stream. But stream.onaddtrack is implemented so we use that. + e.stream.addEventListener('addtrack', te => { + let receiver; + if (window.RTCPeerConnection.prototype.getReceivers) { + receiver = this.getReceivers() + .find(r => r.track && r.track.id === te.track.id); + } else { + receiver = {track: te.track}; + } + + const event = new Event('track'); + event.track = te.track; + event.receiver = receiver; + event.transceiver = {receiver}; + event.streams = [e.stream]; + this.dispatchEvent(event); + }); + e.stream.getTracks().forEach(track => { + let receiver; + if (window.RTCPeerConnection.prototype.getReceivers) { + receiver = this.getReceivers() + .find(r => r.track && r.track.id === track.id); + } else { + receiver = {track}; + } + const event = new Event('track'); + event.track = track; + event.receiver = receiver; + event.transceiver = {receiver}; + event.streams = [e.stream]; + this.dispatchEvent(event); + }); + }; + this.addEventListener('addstream', this._ontrackpoly); + } + return origSetRemoteDescription.apply(this, arguments); + }; + } else { + // even if RTCRtpTransceiver is in window, it is only used and + // emitted in unified-plan. Unfortunately this means we need + // to unconditionally wrap the event. + wrapPeerConnectionEvent(window, 'track', e => { + if (!e.transceiver) { + Object.defineProperty(e, 'transceiver', + {value: {receiver: e.receiver}}); + } + return e; + }); + } + } + + function shimGetSendersWithDtmf(window) { + // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack. + if (typeof window === 'object' && window.RTCPeerConnection && + !('getSenders' in window.RTCPeerConnection.prototype) && + 'createDTMFSender' in window.RTCPeerConnection.prototype) { + const shimSenderWithDtmf = function(pc, track) { + return { + track, + get dtmf() { + if (this._dtmf === undefined) { + if (track.kind === 'audio') { + this._dtmf = pc.createDTMFSender(track); + } else { + this._dtmf = null; + } + } + return this._dtmf; + }, + _pc: pc + }; + }; + + // augment addTrack when getSenders is not available. + if (!window.RTCPeerConnection.prototype.getSenders) { + window.RTCPeerConnection.prototype.getSenders = function getSenders() { + this._senders = this._senders || []; + return this._senders.slice(); // return a copy of the internal state. + }; + const origAddTrack = window.RTCPeerConnection.prototype.addTrack; + window.RTCPeerConnection.prototype.addTrack = + function addTrack(track, stream) { + let sender = origAddTrack.apply(this, arguments); + if (!sender) { + sender = shimSenderWithDtmf(this, track); + this._senders.push(sender); + } + return sender; + }; + + const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack; + window.RTCPeerConnection.prototype.removeTrack = + function removeTrack(sender) { + origRemoveTrack.apply(this, arguments); + const idx = this._senders.indexOf(sender); + if (idx !== -1) { + this._senders.splice(idx, 1); + } + }; + } + const origAddStream = window.RTCPeerConnection.prototype.addStream; + window.RTCPeerConnection.prototype.addStream = function addStream(stream) { + this._senders = this._senders || []; + origAddStream.apply(this, [stream]); + stream.getTracks().forEach(track => { + this._senders.push(shimSenderWithDtmf(this, track)); + }); + }; + + const origRemoveStream = window.RTCPeerConnection.prototype.removeStream; + window.RTCPeerConnection.prototype.removeStream = + function removeStream(stream) { + this._senders = this._senders || []; + origRemoveStream.apply(this, [stream]); + + stream.getTracks().forEach(track => { + const sender = this._senders.find(s => s.track === track); + if (sender) { // remove sender + this._senders.splice(this._senders.indexOf(sender), 1); + } + }); + }; + } else if (typeof window === 'object' && window.RTCPeerConnection && + 'getSenders' in window.RTCPeerConnection.prototype && + 'createDTMFSender' in window.RTCPeerConnection.prototype && + window.RTCRtpSender && + !('dtmf' in window.RTCRtpSender.prototype)) { + const origGetSenders = window.RTCPeerConnection.prototype.getSenders; + window.RTCPeerConnection.prototype.getSenders = function getSenders() { + const senders = origGetSenders.apply(this, []); + senders.forEach(sender => sender._pc = this); + return senders; + }; + + Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', { + get() { + if (this._dtmf === undefined) { + if (this.track.kind === 'audio') { + this._dtmf = this._pc.createDTMFSender(this.track); + } else { + this._dtmf = null; + } + } + return this._dtmf; + } + }); + } + } + + function shimGetStats(window) { + if (!window.RTCPeerConnection) { + return; + } + + const origGetStats = window.RTCPeerConnection.prototype.getStats; + window.RTCPeerConnection.prototype.getStats = function getStats() { + const [selector, onSucc, onErr] = arguments; + + // If selector is a function then we are in the old style stats so just + // pass back the original getStats format to avoid breaking old users. + if (arguments.length > 0 && typeof selector === 'function') { + return origGetStats.apply(this, arguments); + } + + // When spec-style getStats is supported, return those when called with + // either no arguments or the selector argument is null. + if (origGetStats.length === 0 && (arguments.length === 0 || + typeof selector !== 'function')) { + return origGetStats.apply(this, []); + } + + const fixChromeStats_ = function(response) { + const standardReport = {}; + const reports = response.result(); + reports.forEach(report => { + const standardStats = { + id: report.id, + timestamp: report.timestamp, + type: { + localcandidate: 'local-candidate', + remotecandidate: 'remote-candidate' + }[report.type] || report.type + }; + report.names().forEach(name => { + standardStats[name] = report.stat(name); + }); + standardReport[standardStats.id] = standardStats; + }); + + return standardReport; + }; + + // shim getStats with maplike support + const makeMapStats = function(stats) { + return new Map(Object.keys(stats).map(key => [key, stats[key]])); + }; + + if (arguments.length >= 2) { + const successCallbackWrapper_ = function(response) { + onSucc(makeMapStats(fixChromeStats_(response))); + }; + + return origGetStats.apply(this, [successCallbackWrapper_, + selector]); + } + + // promise-support + return new Promise((resolve, reject) => { + origGetStats.apply(this, [ + function(response) { + resolve(makeMapStats(fixChromeStats_(response))); + }, reject]); + }).then(onSucc, onErr); + }; + } + + function shimSenderReceiverGetStats(window) { + if (!(typeof window === 'object' && window.RTCPeerConnection && + window.RTCRtpSender && window.RTCRtpReceiver)) { + return; + } + + // shim sender stats. + if (!('getStats' in window.RTCRtpSender.prototype)) { + const origGetSenders = window.RTCPeerConnection.prototype.getSenders; + if (origGetSenders) { + window.RTCPeerConnection.prototype.getSenders = function getSenders() { + const senders = origGetSenders.apply(this, []); + senders.forEach(sender => sender._pc = this); + return senders; + }; + } + + const origAddTrack = window.RTCPeerConnection.prototype.addTrack; + if (origAddTrack) { + window.RTCPeerConnection.prototype.addTrack = function addTrack() { + const sender = origAddTrack.apply(this, arguments); + sender._pc = this; + return sender; + }; + } + window.RTCRtpSender.prototype.getStats = function getStats() { + const sender = this; + return this._pc.getStats().then(result => + /* Note: this will include stats of all senders that + * send a track with the same id as sender.track as + * it is not possible to identify the RTCRtpSender. + */ + filterStats(result, sender.track, true)); + }; + } + + // shim receiver stats. + if (!('getStats' in window.RTCRtpReceiver.prototype)) { + const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers; + if (origGetReceivers) { + window.RTCPeerConnection.prototype.getReceivers = + function getReceivers() { + const receivers = origGetReceivers.apply(this, []); + receivers.forEach(receiver => receiver._pc = this); + return receivers; + }; + } + wrapPeerConnectionEvent(window, 'track', e => { + e.receiver._pc = e.srcElement; + return e; + }); + window.RTCRtpReceiver.prototype.getStats = function getStats() { + const receiver = this; + return this._pc.getStats().then(result => + filterStats(result, receiver.track, false)); + }; + } + + if (!('getStats' in window.RTCRtpSender.prototype && + 'getStats' in window.RTCRtpReceiver.prototype)) { + return; + } + + // shim RTCPeerConnection.getStats(track). + const origGetStats = window.RTCPeerConnection.prototype.getStats; + window.RTCPeerConnection.prototype.getStats = function getStats() { + if (arguments.length > 0 && + arguments[0] instanceof window.MediaStreamTrack) { + const track = arguments[0]; + let sender; + let receiver; + let err; + this.getSenders().forEach(s => { + if (s.track === track) { + if (sender) { + err = true; + } else { + sender = s; + } + } + }); + this.getReceivers().forEach(r => { + if (r.track === track) { + if (receiver) { + err = true; + } else { + receiver = r; + } + } + return r.track === track; + }); + if (err || (sender && receiver)) { + return Promise.reject(new DOMException( + 'There are more than one sender or receiver for the track.', + 'InvalidAccessError')); + } else if (sender) { + return sender.getStats(); + } else if (receiver) { + return receiver.getStats(); + } + return Promise.reject(new DOMException( + 'There is no sender or receiver for the track.', + 'InvalidAccessError')); + } + return origGetStats.apply(this, arguments); + }; + } + + function shimAddTrackRemoveTrackWithNative(window) { + // shim addTrack/removeTrack with native variants in order to make + // the interactions with legacy getLocalStreams behave as in other browsers. + // Keeps a mapping stream.id => [stream, rtpsenders...] + window.RTCPeerConnection.prototype.getLocalStreams = + function getLocalStreams() { + this._shimmedLocalStreams = this._shimmedLocalStreams || {}; + return Object.keys(this._shimmedLocalStreams) + .map(streamId => this._shimmedLocalStreams[streamId][0]); + }; + + const origAddTrack = window.RTCPeerConnection.prototype.addTrack; + window.RTCPeerConnection.prototype.addTrack = + function addTrack(track, stream) { + if (!stream) { + return origAddTrack.apply(this, arguments); + } + this._shimmedLocalStreams = this._shimmedLocalStreams || {}; + + const sender = origAddTrack.apply(this, arguments); + if (!this._shimmedLocalStreams[stream.id]) { + this._shimmedLocalStreams[stream.id] = [stream, sender]; + } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) { + this._shimmedLocalStreams[stream.id].push(sender); + } + return sender; + }; + + const origAddStream = window.RTCPeerConnection.prototype.addStream; + window.RTCPeerConnection.prototype.addStream = function addStream(stream) { + this._shimmedLocalStreams = this._shimmedLocalStreams || {}; + + stream.getTracks().forEach(track => { + const alreadyExists = this.getSenders().find(s => s.track === track); + if (alreadyExists) { + throw new DOMException('Track already exists.', + 'InvalidAccessError'); + } + }); + const existingSenders = this.getSenders(); + origAddStream.apply(this, arguments); + const newSenders = this.getSenders() + .filter(newSender => existingSenders.indexOf(newSender) === -1); + this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders); + }; + + const origRemoveStream = window.RTCPeerConnection.prototype.removeStream; + window.RTCPeerConnection.prototype.removeStream = + function removeStream(stream) { + this._shimmedLocalStreams = this._shimmedLocalStreams || {}; + delete this._shimmedLocalStreams[stream.id]; + return origRemoveStream.apply(this, arguments); + }; + + const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack; + window.RTCPeerConnection.prototype.removeTrack = + function removeTrack(sender) { + this._shimmedLocalStreams = this._shimmedLocalStreams || {}; + if (sender) { + Object.keys(this._shimmedLocalStreams).forEach(streamId => { + const idx = this._shimmedLocalStreams[streamId].indexOf(sender); + if (idx !== -1) { + this._shimmedLocalStreams[streamId].splice(idx, 1); + } + if (this._shimmedLocalStreams[streamId].length === 1) { + delete this._shimmedLocalStreams[streamId]; + } + }); + } + return origRemoveTrack.apply(this, arguments); + }; + } + + function shimAddTrackRemoveTrack(window, browserDetails) { + if (!window.RTCPeerConnection) { + return; + } + // shim addTrack and removeTrack. + if (window.RTCPeerConnection.prototype.addTrack && + browserDetails.version >= 65) { + return shimAddTrackRemoveTrackWithNative(window); + } + + // also shim pc.getLocalStreams when addTrack is shimmed + // to return the original streams. + const origGetLocalStreams = window.RTCPeerConnection.prototype + .getLocalStreams; + window.RTCPeerConnection.prototype.getLocalStreams = + function getLocalStreams() { + const nativeStreams = origGetLocalStreams.apply(this); + this._reverseStreams = this._reverseStreams || {}; + return nativeStreams.map(stream => this._reverseStreams[stream.id]); + }; + + const origAddStream = window.RTCPeerConnection.prototype.addStream; + window.RTCPeerConnection.prototype.addStream = function addStream(stream) { + this._streams = this._streams || {}; + this._reverseStreams = this._reverseStreams || {}; + + stream.getTracks().forEach(track => { + const alreadyExists = this.getSenders().find(s => s.track === track); + if (alreadyExists) { + throw new DOMException('Track already exists.', + 'InvalidAccessError'); + } + }); + // Add identity mapping for consistency with addTrack. + // Unless this is being used with a stream from addTrack. + if (!this._reverseStreams[stream.id]) { + const newStream = new window.MediaStream(stream.getTracks()); + this._streams[stream.id] = newStream; + this._reverseStreams[newStream.id] = stream; + stream = newStream; + } + origAddStream.apply(this, [stream]); + }; + + const origRemoveStream = window.RTCPeerConnection.prototype.removeStream; + window.RTCPeerConnection.prototype.removeStream = + function removeStream(stream) { + this._streams = this._streams || {}; + this._reverseStreams = this._reverseStreams || {}; + + origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]); + delete this._reverseStreams[(this._streams[stream.id] ? + this._streams[stream.id].id : stream.id)]; + delete this._streams[stream.id]; + }; + + window.RTCPeerConnection.prototype.addTrack = + function addTrack(track, stream) { + if (this.signalingState === 'closed') { + throw new DOMException( + 'The RTCPeerConnection\'s signalingState is \'closed\'.', + 'InvalidStateError'); + } + const streams = [].slice.call(arguments, 1); + if (streams.length !== 1 || + !streams[0].getTracks().find(t => t === track)) { + // this is not fully correct but all we can manage without + // [[associated MediaStreams]] internal slot. + throw new DOMException( + 'The adapter.js addTrack polyfill only supports a single ' + + ' stream which is associated with the specified track.', + 'NotSupportedError'); + } + + const alreadyExists = this.getSenders().find(s => s.track === track); + if (alreadyExists) { + throw new DOMException('Track already exists.', + 'InvalidAccessError'); + } + + this._streams = this._streams || {}; + this._reverseStreams = this._reverseStreams || {}; + const oldStream = this._streams[stream.id]; + if (oldStream) { + // this is using odd Chrome behaviour, use with caution: + // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815 + // Note: we rely on the high-level addTrack/dtmf shim to + // create the sender with a dtmf sender. + oldStream.addTrack(track); + + // Trigger ONN async. + Promise.resolve().then(() => { + this.dispatchEvent(new Event('negotiationneeded')); + }); + } else { + const newStream = new window.MediaStream([track]); + this._streams[stream.id] = newStream; + this._reverseStreams[newStream.id] = stream; + this.addStream(newStream); + } + return this.getSenders().find(s => s.track === track); + }; + + // replace the internal stream id with the external one and + // vice versa. + function replaceInternalStreamId(pc, description) { + let sdp = description.sdp; + Object.keys(pc._reverseStreams || []).forEach(internalId => { + const externalStream = pc._reverseStreams[internalId]; + const internalStream = pc._streams[externalStream.id]; + sdp = sdp.replace(new RegExp(internalStream.id, 'g'), + externalStream.id); + }); + return new RTCSessionDescription({ + type: description.type, + sdp + }); + } + function replaceExternalStreamId(pc, description) { + let sdp = description.sdp; + Object.keys(pc._reverseStreams || []).forEach(internalId => { + const externalStream = pc._reverseStreams[internalId]; + const internalStream = pc._streams[externalStream.id]; + sdp = sdp.replace(new RegExp(externalStream.id, 'g'), + internalStream.id); + }); + return new RTCSessionDescription({ + type: description.type, + sdp + }); + } + ['createOffer', 'createAnswer'].forEach(function(method) { + const nativeMethod = window.RTCPeerConnection.prototype[method]; + const methodObj = {[method]() { + const args = arguments; + const isLegacyCall = arguments.length && + typeof arguments[0] === 'function'; + if (isLegacyCall) { + return nativeMethod.apply(this, [ + (description) => { + const desc = replaceInternalStreamId(this, description); + args[0].apply(null, [desc]); + }, + (err) => { + if (args[1]) { + args[1].apply(null, err); + } + }, arguments[2] + ]); + } + return nativeMethod.apply(this, arguments) + .then(description => replaceInternalStreamId(this, description)); + }}; + window.RTCPeerConnection.prototype[method] = methodObj[method]; + }); + + const origSetLocalDescription = + window.RTCPeerConnection.prototype.setLocalDescription; + window.RTCPeerConnection.prototype.setLocalDescription = + function setLocalDescription() { + if (!arguments.length || !arguments[0].type) { + return origSetLocalDescription.apply(this, arguments); + } + arguments[0] = replaceExternalStreamId(this, arguments[0]); + return origSetLocalDescription.apply(this, arguments); + }; + + // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier + + const origLocalDescription = Object.getOwnPropertyDescriptor( + window.RTCPeerConnection.prototype, 'localDescription'); + Object.defineProperty(window.RTCPeerConnection.prototype, + 'localDescription', { + get() { + const description = origLocalDescription.get.apply(this); + if (description.type === '') { + return description; + } + return replaceInternalStreamId(this, description); + } + }); + + window.RTCPeerConnection.prototype.removeTrack = + function removeTrack(sender) { + if (this.signalingState === 'closed') { + throw new DOMException( + 'The RTCPeerConnection\'s signalingState is \'closed\'.', + 'InvalidStateError'); + } + // We can not yet check for sender instanceof RTCRtpSender + // since we shim RTPSender. So we check if sender._pc is set. + if (!sender._pc) { + throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' + + 'does not implement interface RTCRtpSender.', 'TypeError'); + } + const isLocal = sender._pc === this; + if (!isLocal) { + throw new DOMException('Sender was not created by this connection.', + 'InvalidAccessError'); + } + + // Search for the native stream the senders track belongs to. + this._streams = this._streams || {}; + let stream; + Object.keys(this._streams).forEach(streamid => { + const hasTrack = this._streams[streamid].getTracks() + .find(track => sender.track === track); + if (hasTrack) { + stream = this._streams[streamid]; + } + }); + + if (stream) { + if (stream.getTracks().length === 1) { + // if this is the last track of the stream, remove the stream. This + // takes care of any shimmed _senders. + this.removeStream(this._reverseStreams[stream.id]); + } else { + // relying on the same odd chrome behaviour as above. + stream.removeTrack(sender.track); + } + this.dispatchEvent(new Event('negotiationneeded')); + } + }; + } + + function shimPeerConnection$2(window, browserDetails) { + if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) { + // very basic support for old versions. + window.RTCPeerConnection = window.webkitRTCPeerConnection; + } + if (!window.RTCPeerConnection) { + return; + } + + // shim implicit creation of RTCSessionDescription/RTCIceCandidate + if (browserDetails.version < 53) { + ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] + .forEach(function(method) { + const nativeMethod = window.RTCPeerConnection.prototype[method]; + const methodObj = {[method]() { + arguments[0] = new ((method === 'addIceCandidate') ? + window.RTCIceCandidate : + window.RTCSessionDescription)(arguments[0]); + return nativeMethod.apply(this, arguments); + }}; + window.RTCPeerConnection.prototype[method] = methodObj[method]; + }); + } + } + + // Attempt to fix ONN in plan-b mode. + function fixNegotiationNeeded(window, browserDetails) { + wrapPeerConnectionEvent(window, 'negotiationneeded', e => { + const pc = e.target; + if (browserDetails.version < 72 || (pc.getConfiguration && + pc.getConfiguration().sdpSemantics === 'plan-b')) { + if (pc.signalingState !== 'stable') { + return; + } + } + return e; + }); + } + + var chromeShim = /*#__PURE__*/Object.freeze({ + __proto__: null, + shimMediaStream: shimMediaStream, + shimOnTrack: shimOnTrack$1, + shimGetSendersWithDtmf: shimGetSendersWithDtmf, + shimGetStats: shimGetStats, + shimSenderReceiverGetStats: shimSenderReceiverGetStats, + shimAddTrackRemoveTrackWithNative: shimAddTrackRemoveTrackWithNative, + shimAddTrackRemoveTrack: shimAddTrackRemoveTrack, + shimPeerConnection: shimPeerConnection$2, + fixNegotiationNeeded: fixNegotiationNeeded, + shimGetUserMedia: shimGetUserMedia$3, + shimGetDisplayMedia: shimGetDisplayMedia$2 + }); + + /* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + // Edge does not like + // 1) stun: filtered after 14393 unless ?transport=udp is present + // 2) turn: that does not have all of turn:host:port?transport=udp + // 3) turn: with ipv6 addresses + // 4) turn: occurring muliple times + function filterIceServers$1(iceServers, edgeVersion) { + let hasTurn = false; + iceServers = JSON.parse(JSON.stringify(iceServers)); + return iceServers.filter(server => { + if (server && (server.urls || server.url)) { + let urls = server.urls || server.url; + if (server.url && !server.urls) { + deprecated('RTCIceServer.url', 'RTCIceServer.urls'); + } + const isString = typeof urls === 'string'; + if (isString) { + urls = [urls]; + } + urls = urls.filter(url => { + // filter STUN unconditionally. + if (url.indexOf('stun:') === 0) { + return false; + } + + const validTurn = url.startsWith('turn') && + !url.startsWith('turn:[') && + url.includes('transport=udp'); + if (validTurn && !hasTurn) { + hasTurn = true; + return true; + } + return validTurn && !hasTurn; + }); + + delete server.url; + server.urls = isString ? urls[0] : urls; + return !!urls.length; + } + }); + } + + function createCommonjsModule(fn) { + var module = { exports: {} }; + return fn(module, module.exports), module.exports; + } + + /* eslint-env node */ + + var sdp = createCommonjsModule(function (module) { + + // SDP helpers. + var SDPUtils = {}; + + // Generate an alphanumeric identifier for cname or mids. + // TODO: use UUIDs instead? https://gist.github.com/jed/982883 + SDPUtils.generateIdentifier = function() { + return Math.random().toString(36).substr(2, 10); + }; + + // The RTCP CNAME used by all peerconnections from the same JS. + SDPUtils.localCName = SDPUtils.generateIdentifier(); + + // Splits SDP into lines, dealing with both CRLF and LF. + SDPUtils.splitLines = function(blob) { + return blob.trim().split('\n').map(function(line) { + return line.trim(); + }); + }; + // Splits SDP into sessionpart and mediasections. Ensures CRLF. + SDPUtils.splitSections = function(blob) { + var parts = blob.split('\nm='); + return parts.map(function(part, index) { + return (index > 0 ? 'm=' + part : part).trim() + '\r\n'; + }); + }; + + // returns the session description. + SDPUtils.getDescription = function(blob) { + var sections = SDPUtils.splitSections(blob); + return sections && sections[0]; + }; + + // returns the individual media sections. + SDPUtils.getMediaSections = function(blob) { + var sections = SDPUtils.splitSections(blob); + sections.shift(); + return sections; + }; + + // Returns lines that start with a certain prefix. + SDPUtils.matchPrefix = function(blob, prefix) { + return SDPUtils.splitLines(blob).filter(function(line) { + return line.indexOf(prefix) === 0; + }); + }; + + // Parses an ICE candidate line. Sample input: + // candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8 + // rport 55996" + SDPUtils.parseCandidate = function(line) { + var parts; + // Parse both variants. + if (line.indexOf('a=candidate:') === 0) { + parts = line.substring(12).split(' '); + } else { + parts = line.substring(10).split(' '); + } + + var candidate = { + foundation: parts[0], + component: parseInt(parts[1], 10), + protocol: parts[2].toLowerCase(), + priority: parseInt(parts[3], 10), + ip: parts[4], + address: parts[4], // address is an alias for ip. + port: parseInt(parts[5], 10), + // skip parts[6] == 'typ' + type: parts[7] + }; + + for (var i = 8; i < parts.length; i += 2) { + switch (parts[i]) { + case 'raddr': + candidate.relatedAddress = parts[i + 1]; + break; + case 'rport': + candidate.relatedPort = parseInt(parts[i + 1], 10); + break; + case 'tcptype': + candidate.tcpType = parts[i + 1]; + break; + case 'ufrag': + candidate.ufrag = parts[i + 1]; // for backward compability. + candidate.usernameFragment = parts[i + 1]; + break; + default: // extension handling, in particular ufrag + candidate[parts[i]] = parts[i + 1]; + break; + } + } + return candidate; + }; + + // Translates a candidate object into SDP candidate attribute. + SDPUtils.writeCandidate = function(candidate) { + var sdp = []; + sdp.push(candidate.foundation); + sdp.push(candidate.component); + sdp.push(candidate.protocol.toUpperCase()); + sdp.push(candidate.priority); + sdp.push(candidate.address || candidate.ip); + sdp.push(candidate.port); + + var type = candidate.type; + sdp.push('typ'); + sdp.push(type); + if (type !== 'host' && candidate.relatedAddress && + candidate.relatedPort) { + sdp.push('raddr'); + sdp.push(candidate.relatedAddress); + sdp.push('rport'); + sdp.push(candidate.relatedPort); + } + if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') { + sdp.push('tcptype'); + sdp.push(candidate.tcpType); + } + if (candidate.usernameFragment || candidate.ufrag) { + sdp.push('ufrag'); + sdp.push(candidate.usernameFragment || candidate.ufrag); + } + return 'candidate:' + sdp.join(' '); + }; + + // Parses an ice-options line, returns an array of option tags. + // a=ice-options:foo bar + SDPUtils.parseIceOptions = function(line) { + return line.substr(14).split(' '); + }; + + // Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input: + // a=rtpmap:111 opus/48000/2 + SDPUtils.parseRtpMap = function(line) { + var parts = line.substr(9).split(' '); + var parsed = { + payloadType: parseInt(parts.shift(), 10) // was: id + }; + + parts = parts[0].split('/'); + + parsed.name = parts[0]; + parsed.clockRate = parseInt(parts[1], 10); // was: clockrate + parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1; + // legacy alias, got renamed back to channels in ORTC. + parsed.numChannels = parsed.channels; + return parsed; + }; + + // Generate an a=rtpmap line from RTCRtpCodecCapability or + // RTCRtpCodecParameters. + SDPUtils.writeRtpMap = function(codec) { + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + var channels = codec.channels || codec.numChannels || 1; + return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate + + (channels !== 1 ? '/' + channels : '') + '\r\n'; + }; + + // Parses an a=extmap line (headerextension from RFC 5285). Sample input: + // a=extmap:2 urn:ietf:params:rtp-hdrext:toffset + // a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset + SDPUtils.parseExtmap = function(line) { + var parts = line.substr(9).split(' '); + return { + id: parseInt(parts[0], 10), + direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv', + uri: parts[1] + }; + }; + + // Generates a=extmap line from RTCRtpHeaderExtensionParameters or + // RTCRtpHeaderExtension. + SDPUtils.writeExtmap = function(headerExtension) { + return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) + + (headerExtension.direction && headerExtension.direction !== 'sendrecv' + ? '/' + headerExtension.direction + : '') + + ' ' + headerExtension.uri + '\r\n'; + }; + + // Parses an ftmp line, returns dictionary. Sample input: + // a=fmtp:96 vbr=on;cng=on + // Also deals with vbr=on; cng=on + SDPUtils.parseFmtp = function(line) { + var parsed = {}; + var kv; + var parts = line.substr(line.indexOf(' ') + 1).split(';'); + for (var j = 0; j < parts.length; j++) { + kv = parts[j].trim().split('='); + parsed[kv[0].trim()] = kv[1]; + } + return parsed; + }; + + // Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters. + SDPUtils.writeFmtp = function(codec) { + var line = ''; + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + if (codec.parameters && Object.keys(codec.parameters).length) { + var params = []; + Object.keys(codec.parameters).forEach(function(param) { + if (codec.parameters[param]) { + params.push(param + '=' + codec.parameters[param]); + } else { + params.push(param); + } + }); + line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\r\n'; + } + return line; + }; + + // Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input: + // a=rtcp-fb:98 nack rpsi + SDPUtils.parseRtcpFb = function(line) { + var parts = line.substr(line.indexOf(' ') + 1).split(' '); + return { + type: parts.shift(), + parameter: parts.join(' ') + }; + }; + // Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters. + SDPUtils.writeRtcpFb = function(codec) { + var lines = ''; + var pt = codec.payloadType; + if (codec.preferredPayloadType !== undefined) { + pt = codec.preferredPayloadType; + } + if (codec.rtcpFeedback && codec.rtcpFeedback.length) { + // FIXME: special handling for trr-int? + codec.rtcpFeedback.forEach(function(fb) { + lines += 'a=rtcp-fb:' + pt + ' ' + fb.type + + (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') + + '\r\n'; + }); + } + return lines; + }; + + // Parses an RFC 5576 ssrc media attribute. Sample input: + // a=ssrc:3735928559 cname:something + SDPUtils.parseSsrcMedia = function(line) { + var sp = line.indexOf(' '); + var parts = { + ssrc: parseInt(line.substr(7, sp - 7), 10) + }; + var colon = line.indexOf(':', sp); + if (colon > -1) { + parts.attribute = line.substr(sp + 1, colon - sp - 1); + parts.value = line.substr(colon + 1); + } else { + parts.attribute = line.substr(sp + 1); + } + return parts; + }; + + SDPUtils.parseSsrcGroup = function(line) { + var parts = line.substr(13).split(' '); + return { + semantics: parts.shift(), + ssrcs: parts.map(function(ssrc) { + return parseInt(ssrc, 10); + }) + }; + }; + + // Extracts the MID (RFC 5888) from a media section. + // returns the MID or undefined if no mid line was found. + SDPUtils.getMid = function(mediaSection) { + var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0]; + if (mid) { + return mid.substr(6); + } + }; + + SDPUtils.parseFingerprint = function(line) { + var parts = line.substr(14).split(' '); + return { + algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge. + value: parts[1] + }; + }; + + // Extracts DTLS parameters from SDP media section or sessionpart. + // FIXME: for consistency with other functions this should only + // get the fingerprint line as input. See also getIceParameters. + SDPUtils.getDtlsParameters = function(mediaSection, sessionpart) { + var lines = SDPUtils.matchPrefix(mediaSection + sessionpart, + 'a=fingerprint:'); + // Note: a=setup line is ignored since we use the 'auto' role. + // Note2: 'algorithm' is not case sensitive except in Edge. + return { + role: 'auto', + fingerprints: lines.map(SDPUtils.parseFingerprint) + }; + }; + + // Serializes DTLS parameters to SDP. + SDPUtils.writeDtlsParameters = function(params, setupType) { + var sdp = 'a=setup:' + setupType + '\r\n'; + params.fingerprints.forEach(function(fp) { + sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\r\n'; + }); + return sdp; + }; + + // Parses a=crypto lines into + // https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members + SDPUtils.parseCryptoLine = function(line) { + var parts = line.substr(9).split(' '); + return { + tag: parseInt(parts[0], 10), + cryptoSuite: parts[1], + keyParams: parts[2], + sessionParams: parts.slice(3), + }; + }; + + SDPUtils.writeCryptoLine = function(parameters) { + return 'a=crypto:' + parameters.tag + ' ' + + parameters.cryptoSuite + ' ' + + (typeof parameters.keyParams === 'object' + ? SDPUtils.writeCryptoKeyParams(parameters.keyParams) + : parameters.keyParams) + + (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') + + '\r\n'; + }; + + // Parses the crypto key parameters into + // https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam* + SDPUtils.parseCryptoKeyParams = function(keyParams) { + if (keyParams.indexOf('inline:') !== 0) { + return null; + } + var parts = keyParams.substr(7).split('|'); + return { + keyMethod: 'inline', + keySalt: parts[0], + lifeTime: parts[1], + mkiValue: parts[2] ? parts[2].split(':')[0] : undefined, + mkiLength: parts[2] ? parts[2].split(':')[1] : undefined, + }; + }; + + SDPUtils.writeCryptoKeyParams = function(keyParams) { + return keyParams.keyMethod + ':' + + keyParams.keySalt + + (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') + + (keyParams.mkiValue && keyParams.mkiLength + ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength + : ''); + }; + + // Extracts all SDES paramters. + SDPUtils.getCryptoParameters = function(mediaSection, sessionpart) { + var lines = SDPUtils.matchPrefix(mediaSection + sessionpart, + 'a=crypto:'); + return lines.map(SDPUtils.parseCryptoLine); + }; + + // Parses ICE information from SDP media section or sessionpart. + // FIXME: for consistency with other functions this should only + // get the ice-ufrag and ice-pwd lines as input. + SDPUtils.getIceParameters = function(mediaSection, sessionpart) { + var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart, + 'a=ice-ufrag:')[0]; + var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart, + 'a=ice-pwd:')[0]; + if (!(ufrag && pwd)) { + return null; + } + return { + usernameFragment: ufrag.substr(12), + password: pwd.substr(10), + }; + }; + + // Serializes ICE parameters to SDP. + SDPUtils.writeIceParameters = function(params) { + return 'a=ice-ufrag:' + params.usernameFragment + '\r\n' + + 'a=ice-pwd:' + params.password + '\r\n'; + }; + + // Parses the SDP media section and returns RTCRtpParameters. + SDPUtils.parseRtpParameters = function(mediaSection) { + var description = { + codecs: [], + headerExtensions: [], + fecMechanisms: [], + rtcp: [] + }; + var lines = SDPUtils.splitLines(mediaSection); + var mline = lines[0].split(' '); + for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..] + var pt = mline[i]; + var rtpmapline = SDPUtils.matchPrefix( + mediaSection, 'a=rtpmap:' + pt + ' ')[0]; + if (rtpmapline) { + var codec = SDPUtils.parseRtpMap(rtpmapline); + var fmtps = SDPUtils.matchPrefix( + mediaSection, 'a=fmtp:' + pt + ' '); + // Only the first a=fmtp: is considered. + codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {}; + codec.rtcpFeedback = SDPUtils.matchPrefix( + mediaSection, 'a=rtcp-fb:' + pt + ' ') + .map(SDPUtils.parseRtcpFb); + description.codecs.push(codec); + // parse FEC mechanisms from rtpmap lines. + switch (codec.name.toUpperCase()) { + case 'RED': + case 'ULPFEC': + description.fecMechanisms.push(codec.name.toUpperCase()); + break; + } + } + } + SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) { + description.headerExtensions.push(SDPUtils.parseExtmap(line)); + }); + // FIXME: parse rtcp. + return description; + }; + + // Generates parts of the SDP media section describing the capabilities / + // parameters. + SDPUtils.writeRtpDescription = function(kind, caps) { + var sdp = ''; + + // Build the mline. + sdp += 'm=' + kind + ' '; + sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs. + sdp += ' UDP/TLS/RTP/SAVPF '; + sdp += caps.codecs.map(function(codec) { + if (codec.preferredPayloadType !== undefined) { + return codec.preferredPayloadType; + } + return codec.payloadType; + }).join(' ') + '\r\n'; + + sdp += 'c=IN IP4 0.0.0.0\r\n'; + sdp += 'a=rtcp:9 IN IP4 0.0.0.0\r\n'; + + // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb. + caps.codecs.forEach(function(codec) { + sdp += SDPUtils.writeRtpMap(codec); + sdp += SDPUtils.writeFmtp(codec); + sdp += SDPUtils.writeRtcpFb(codec); + }); + var maxptime = 0; + caps.codecs.forEach(function(codec) { + if (codec.maxptime > maxptime) { + maxptime = codec.maxptime; + } + }); + if (maxptime > 0) { + sdp += 'a=maxptime:' + maxptime + '\r\n'; + } + sdp += 'a=rtcp-mux\r\n'; + + if (caps.headerExtensions) { + caps.headerExtensions.forEach(function(extension) { + sdp += SDPUtils.writeExtmap(extension); + }); + } + // FIXME: write fecMechanisms. + return sdp; + }; + + // Parses the SDP media section and returns an array of + // RTCRtpEncodingParameters. + SDPUtils.parseRtpEncodingParameters = function(mediaSection) { + var encodingParameters = []; + var description = SDPUtils.parseRtpParameters(mediaSection); + var hasRed = description.fecMechanisms.indexOf('RED') !== -1; + var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1; + + // filter a=ssrc:... cname:, ignore PlanB-msid + var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') + .map(function(line) { + return SDPUtils.parseSsrcMedia(line); + }) + .filter(function(parts) { + return parts.attribute === 'cname'; + }); + var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc; + var secondarySsrc; + + var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID') + .map(function(line) { + var parts = line.substr(17).split(' '); + return parts.map(function(part) { + return parseInt(part, 10); + }); + }); + if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) { + secondarySsrc = flows[0][1]; + } + + description.codecs.forEach(function(codec) { + if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) { + var encParam = { + ssrc: primarySsrc, + codecPayloadType: parseInt(codec.parameters.apt, 10) + }; + if (primarySsrc && secondarySsrc) { + encParam.rtx = {ssrc: secondarySsrc}; + } + encodingParameters.push(encParam); + if (hasRed) { + encParam = JSON.parse(JSON.stringify(encParam)); + encParam.fec = { + ssrc: primarySsrc, + mechanism: hasUlpfec ? 'red+ulpfec' : 'red' + }; + encodingParameters.push(encParam); + } + } + }); + if (encodingParameters.length === 0 && primarySsrc) { + encodingParameters.push({ + ssrc: primarySsrc + }); + } + + // we support both b=AS and b=TIAS but interpret AS as TIAS. + var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b='); + if (bandwidth.length) { + if (bandwidth[0].indexOf('b=TIAS:') === 0) { + bandwidth = parseInt(bandwidth[0].substr(7), 10); + } else if (bandwidth[0].indexOf('b=AS:') === 0) { + // use formula from JSEP to convert b=AS to TIAS value. + bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95 + - (50 * 40 * 8); + } else { + bandwidth = undefined; + } + encodingParameters.forEach(function(params) { + params.maxBitrate = bandwidth; + }); + } + return encodingParameters; + }; + + // parses http://draft.ortc.org/#rtcrtcpparameters* + SDPUtils.parseRtcpParameters = function(mediaSection) { + var rtcpParameters = {}; + + // Gets the first SSRC. Note tha with RTX there might be multiple + // SSRCs. + var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') + .map(function(line) { + return SDPUtils.parseSsrcMedia(line); + }) + .filter(function(obj) { + return obj.attribute === 'cname'; + })[0]; + if (remoteSsrc) { + rtcpParameters.cname = remoteSsrc.value; + rtcpParameters.ssrc = remoteSsrc.ssrc; + } + + // Edge uses the compound attribute instead of reducedSize + // compound is !reducedSize + var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize'); + rtcpParameters.reducedSize = rsize.length > 0; + rtcpParameters.compound = rsize.length === 0; + + // parses the rtcp-mux attrіbute. + // Note that Edge does not support unmuxed RTCP. + var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux'); + rtcpParameters.mux = mux.length > 0; + + return rtcpParameters; + }; + + // parses either a=msid: or a=ssrc:... msid lines and returns + // the id of the MediaStream and MediaStreamTrack. + SDPUtils.parseMsid = function(mediaSection) { + var parts; + var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:'); + if (spec.length === 1) { + parts = spec[0].substr(7).split(' '); + return {stream: parts[0], track: parts[1]}; + } + var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') + .map(function(line) { + return SDPUtils.parseSsrcMedia(line); + }) + .filter(function(msidParts) { + return msidParts.attribute === 'msid'; + }); + if (planB.length > 0) { + parts = planB[0].value.split(' '); + return {stream: parts[0], track: parts[1]}; + } + }; + + // SCTP + // parses draft-ietf-mmusic-sctp-sdp-26 first and falls back + // to draft-ietf-mmusic-sctp-sdp-05 + SDPUtils.parseSctpDescription = function(mediaSection) { + var mline = SDPUtils.parseMLine(mediaSection); + var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:'); + var maxMessageSize; + if (maxSizeLine.length > 0) { + maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10); + } + if (isNaN(maxMessageSize)) { + maxMessageSize = 65536; + } + var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:'); + if (sctpPort.length > 0) { + return { + port: parseInt(sctpPort[0].substr(12), 10), + protocol: mline.fmt, + maxMessageSize: maxMessageSize + }; + } + var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:'); + if (sctpMapLines.length > 0) { + var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0] + .substr(10) + .split(' '); + return { + port: parseInt(parts[0], 10), + protocol: parts[1], + maxMessageSize: maxMessageSize + }; + } + }; + + // SCTP + // outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers + // support by now receiving in this format, unless we originally parsed + // as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line + // protocol of DTLS/SCTP -- without UDP/ or TCP/) + SDPUtils.writeSctpDescription = function(media, sctp) { + var output = []; + if (media.protocol !== 'DTLS/SCTP') { + output = [ + 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\r\n', + 'c=IN IP4 0.0.0.0\r\n', + 'a=sctp-port:' + sctp.port + '\r\n' + ]; + } else { + output = [ + 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\r\n', + 'c=IN IP4 0.0.0.0\r\n', + 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\r\n' + ]; + } + if (sctp.maxMessageSize !== undefined) { + output.push('a=max-message-size:' + sctp.maxMessageSize + '\r\n'); + } + return output.join(''); + }; + + // Generate a session ID for SDP. + // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1 + // recommends using a cryptographically random +ve 64-bit value + // but right now this should be acceptable and within the right range + SDPUtils.generateSessionId = function() { + return Math.random().toString().substr(2, 21); + }; + + // Write boilder plate for start of SDP + // sessId argument is optional - if not supplied it will + // be generated randomly + // sessVersion is optional and defaults to 2 + // sessUser is optional and defaults to 'thisisadapterortc' + SDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) { + var sessionId; + var version = sessVer !== undefined ? sessVer : 2; + if (sessId) { + sessionId = sessId; + } else { + sessionId = SDPUtils.generateSessionId(); + } + var user = sessUser || 'thisisadapterortc'; + // FIXME: sess-id should be an NTP timestamp. + return 'v=0\r\n' + + 'o=' + user + ' ' + sessionId + ' ' + version + + ' IN IP4 127.0.0.1\r\n' + + 's=-\r\n' + + 't=0 0\r\n'; + }; + + SDPUtils.writeMediaSection = function(transceiver, caps, type, stream) { + var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps); + + // Map ICE parameters (ufrag, pwd) to SDP. + sdp += SDPUtils.writeIceParameters( + transceiver.iceGatherer.getLocalParameters()); + + // Map DTLS parameters to SDP. + sdp += SDPUtils.writeDtlsParameters( + transceiver.dtlsTransport.getLocalParameters(), + type === 'offer' ? 'actpass' : 'active'); + + sdp += 'a=mid:' + transceiver.mid + '\r\n'; + + if (transceiver.direction) { + sdp += 'a=' + transceiver.direction + '\r\n'; + } else if (transceiver.rtpSender && transceiver.rtpReceiver) { + sdp += 'a=sendrecv\r\n'; + } else if (transceiver.rtpSender) { + sdp += 'a=sendonly\r\n'; + } else if (transceiver.rtpReceiver) { + sdp += 'a=recvonly\r\n'; + } else { + sdp += 'a=inactive\r\n'; + } + + if (transceiver.rtpSender) { + // spec. + var msid = 'msid:' + stream.id + ' ' + + transceiver.rtpSender.track.id + '\r\n'; + sdp += 'a=' + msid; + + // for Chrome. + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' ' + msid; + if (transceiver.sendEncodingParameters[0].rtx) { + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc + + ' ' + msid; + sdp += 'a=ssrc-group:FID ' + + transceiver.sendEncodingParameters[0].ssrc + ' ' + + transceiver.sendEncodingParameters[0].rtx.ssrc + + '\r\n'; + } + } + // FIXME: this should be written by writeRtpDescription. + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' cname:' + SDPUtils.localCName + '\r\n'; + if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) { + sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc + + ' cname:' + SDPUtils.localCName + '\r\n'; + } + return sdp; + }; + + // Gets the direction from the mediaSection or the sessionpart. + SDPUtils.getDirection = function(mediaSection, sessionpart) { + // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv. + var lines = SDPUtils.splitLines(mediaSection); + for (var i = 0; i < lines.length; i++) { + switch (lines[i]) { + case 'a=sendrecv': + case 'a=sendonly': + case 'a=recvonly': + case 'a=inactive': + return lines[i].substr(2); + // FIXME: What should happen here? + } + } + if (sessionpart) { + return SDPUtils.getDirection(sessionpart); + } + return 'sendrecv'; + }; + + SDPUtils.getKind = function(mediaSection) { + var lines = SDPUtils.splitLines(mediaSection); + var mline = lines[0].split(' '); + return mline[0].substr(2); + }; + + SDPUtils.isRejected = function(mediaSection) { + return mediaSection.split(' ', 2)[1] === '0'; + }; + + SDPUtils.parseMLine = function(mediaSection) { + var lines = SDPUtils.splitLines(mediaSection); + var parts = lines[0].substr(2).split(' '); + return { + kind: parts[0], + port: parseInt(parts[1], 10), + protocol: parts[2], + fmt: parts.slice(3).join(' ') + }; + }; + + SDPUtils.parseOLine = function(mediaSection) { + var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0]; + var parts = line.substr(2).split(' '); + return { + username: parts[0], + sessionId: parts[1], + sessionVersion: parseInt(parts[2], 10), + netType: parts[3], + addressType: parts[4], + address: parts[5] + }; + }; + + // a very naive interpretation of a valid SDP. + SDPUtils.isValidSDP = function(blob) { + if (typeof blob !== 'string' || blob.length === 0) { + return false; + } + var lines = SDPUtils.splitLines(blob); + for (var i = 0; i < lines.length; i++) { + if (lines[i].length < 2 || lines[i].charAt(1) !== '=') { + return false; + } + // TODO: check the modifier a bit more. + } + return true; + }; + + // Expose public methods. + { + module.exports = SDPUtils; + } + }); + + /* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + + + function fixStatsType(stat) { + return { + inboundrtp: 'inbound-rtp', + outboundrtp: 'outbound-rtp', + candidatepair: 'candidate-pair', + localcandidate: 'local-candidate', + remotecandidate: 'remote-candidate' + }[stat.type] || stat.type; + } + + function writeMediaSection(transceiver, caps, type, stream, dtlsRole) { + var sdp$1 = sdp.writeRtpDescription(transceiver.kind, caps); + + // Map ICE parameters (ufrag, pwd) to SDP. + sdp$1 += sdp.writeIceParameters( + transceiver.iceGatherer.getLocalParameters()); + + // Map DTLS parameters to SDP. + sdp$1 += sdp.writeDtlsParameters( + transceiver.dtlsTransport.getLocalParameters(), + type === 'offer' ? 'actpass' : dtlsRole || 'active'); + + sdp$1 += 'a=mid:' + transceiver.mid + '\r\n'; + + if (transceiver.rtpSender && transceiver.rtpReceiver) { + sdp$1 += 'a=sendrecv\r\n'; + } else if (transceiver.rtpSender) { + sdp$1 += 'a=sendonly\r\n'; + } else if (transceiver.rtpReceiver) { + sdp$1 += 'a=recvonly\r\n'; + } else { + sdp$1 += 'a=inactive\r\n'; + } + + if (transceiver.rtpSender) { + var trackId = transceiver.rtpSender._initialTrackId || + transceiver.rtpSender.track.id; + transceiver.rtpSender._initialTrackId = trackId; + // spec. + var msid = 'msid:' + (stream ? stream.id : '-') + ' ' + + trackId + '\r\n'; + sdp$1 += 'a=' + msid; + // for Chrome. Legacy should no longer be required. + sdp$1 += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' ' + msid; + + // RTX + if (transceiver.sendEncodingParameters[0].rtx) { + sdp$1 += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc + + ' ' + msid; + sdp$1 += 'a=ssrc-group:FID ' + + transceiver.sendEncodingParameters[0].ssrc + ' ' + + transceiver.sendEncodingParameters[0].rtx.ssrc + + '\r\n'; + } + } + // FIXME: this should be written by writeRtpDescription. + sdp$1 += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + + ' cname:' + sdp.localCName + '\r\n'; + if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) { + sdp$1 += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc + + ' cname:' + sdp.localCName + '\r\n'; + } + return sdp$1; + } + + // Edge does not like + // 1) stun: filtered after 14393 unless ?transport=udp is present + // 2) turn: that does not have all of turn:host:port?transport=udp + // 3) turn: with ipv6 addresses + // 4) turn: occurring muliple times + function filterIceServers(iceServers, edgeVersion) { + var hasTurn = false; + iceServers = JSON.parse(JSON.stringify(iceServers)); + return iceServers.filter(function(server) { + if (server && (server.urls || server.url)) { + var urls = server.urls || server.url; + if (server.url && !server.urls) { + console.warn('RTCIceServer.url is deprecated! Use urls instead.'); + } + var isString = typeof urls === 'string'; + if (isString) { + urls = [urls]; + } + urls = urls.filter(function(url) { + var validTurn = url.indexOf('turn:') === 0 && + url.indexOf('transport=udp') !== -1 && + url.indexOf('turn:[') === -1 && + !hasTurn; + + if (validTurn) { + hasTurn = true; + return true; + } + return url.indexOf('stun:') === 0 && edgeVersion >= 14393 && + url.indexOf('?transport=udp') === -1; + }); + + delete server.url; + server.urls = isString ? urls[0] : urls; + return !!urls.length; + } + }); + } + + // Determines the intersection of local and remote capabilities. + function getCommonCapabilities(localCapabilities, remoteCapabilities) { + var commonCapabilities = { + codecs: [], + headerExtensions: [], + fecMechanisms: [] + }; + + var findCodecByPayloadType = function(pt, codecs) { + pt = parseInt(pt, 10); + for (var i = 0; i < codecs.length; i++) { + if (codecs[i].payloadType === pt || + codecs[i].preferredPayloadType === pt) { + return codecs[i]; + } + } + }; + + var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) { + var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs); + var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs); + return lCodec && rCodec && + lCodec.name.toLowerCase() === rCodec.name.toLowerCase(); + }; + + localCapabilities.codecs.forEach(function(lCodec) { + for (var i = 0; i < remoteCapabilities.codecs.length; i++) { + var rCodec = remoteCapabilities.codecs[i]; + if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() && + lCodec.clockRate === rCodec.clockRate) { + if (lCodec.name.toLowerCase() === 'rtx' && + lCodec.parameters && rCodec.parameters.apt) { + // for RTX we need to find the local rtx that has a apt + // which points to the same local codec as the remote one. + if (!rtxCapabilityMatches(lCodec, rCodec, + localCapabilities.codecs, remoteCapabilities.codecs)) { + continue; + } + } + rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy + // number of channels is the highest common number of channels + rCodec.numChannels = Math.min(lCodec.numChannels, + rCodec.numChannels); + // push rCodec so we reply with offerer payload type + commonCapabilities.codecs.push(rCodec); + + // determine common feedback mechanisms + rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) { + for (var j = 0; j < lCodec.rtcpFeedback.length; j++) { + if (lCodec.rtcpFeedback[j].type === fb.type && + lCodec.rtcpFeedback[j].parameter === fb.parameter) { + return true; + } + } + return false; + }); + // FIXME: also need to determine .parameters + // see https://github.com/openpeer/ortc/issues/569 + break; + } + } + }); + + localCapabilities.headerExtensions.forEach(function(lHeaderExtension) { + for (var i = 0; i < remoteCapabilities.headerExtensions.length; + i++) { + var rHeaderExtension = remoteCapabilities.headerExtensions[i]; + if (lHeaderExtension.uri === rHeaderExtension.uri) { + commonCapabilities.headerExtensions.push(rHeaderExtension); + break; + } + } + }); + + // FIXME: fecMechanisms + return commonCapabilities; + } + + // is action=setLocalDescription with type allowed in signalingState + function isActionAllowedInSignalingState(action, type, signalingState) { + return { + offer: { + setLocalDescription: ['stable', 'have-local-offer'], + setRemoteDescription: ['stable', 'have-remote-offer'] + }, + answer: { + setLocalDescription: ['have-remote-offer', 'have-local-pranswer'], + setRemoteDescription: ['have-local-offer', 'have-remote-pranswer'] + } + }[type][action].indexOf(signalingState) !== -1; + } + + function maybeAddCandidate(iceTransport, candidate) { + // Edge's internal representation adds some fields therefore + // not all fieldѕ are taken into account. + var alreadyAdded = iceTransport.getRemoteCandidates() + .find(function(remoteCandidate) { + return candidate.foundation === remoteCandidate.foundation && + candidate.ip === remoteCandidate.ip && + candidate.port === remoteCandidate.port && + candidate.priority === remoteCandidate.priority && + candidate.protocol === remoteCandidate.protocol && + candidate.type === remoteCandidate.type; + }); + if (!alreadyAdded) { + iceTransport.addRemoteCandidate(candidate); + } + return !alreadyAdded; + } + + + function makeError(name, description) { + var e = new Error(description); + e.name = name; + // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names + e.code = { + NotSupportedError: 9, + InvalidStateError: 11, + InvalidAccessError: 15, + TypeError: undefined, + OperationError: undefined + }[name]; + return e; + } + + var rtcpeerconnection = function(window, edgeVersion) { + // https://w3c.github.io/mediacapture-main/#mediastream + // Helper function to add the track to the stream and + // dispatch the event ourselves. + function addTrackToStreamAndFireEvent(track, stream) { + stream.addTrack(track); + stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack', + {track: track})); + } + + function removeTrackFromStreamAndFireEvent(track, stream) { + stream.removeTrack(track); + stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack', + {track: track})); + } + + function fireAddTrack(pc, track, receiver, streams) { + var trackEvent = new Event('track'); + trackEvent.track = track; + trackEvent.receiver = receiver; + trackEvent.transceiver = {receiver: receiver}; + trackEvent.streams = streams; + window.setTimeout(function() { + pc._dispatchEvent('track', trackEvent); + }); + } + + var RTCPeerConnection = function(config) { + var pc = this; + + var _eventTarget = document.createDocumentFragment(); + ['addEventListener', 'removeEventListener', 'dispatchEvent'] + .forEach(function(method) { + pc[method] = _eventTarget[method].bind(_eventTarget); + }); + + this.canTrickleIceCandidates = null; + + this.needNegotiation = false; + + this.localStreams = []; + this.remoteStreams = []; + + this._localDescription = null; + this._remoteDescription = null; + + this.signalingState = 'stable'; + this.iceConnectionState = 'new'; + this.connectionState = 'new'; + this.iceGatheringState = 'new'; + + config = JSON.parse(JSON.stringify(config || {})); + + this.usingBundle = config.bundlePolicy === 'max-bundle'; + if (config.rtcpMuxPolicy === 'negotiate') { + throw(makeError('NotSupportedError', + 'rtcpMuxPolicy \'negotiate\' is not supported')); + } else if (!config.rtcpMuxPolicy) { + config.rtcpMuxPolicy = 'require'; + } + + switch (config.iceTransportPolicy) { + case 'all': + case 'relay': + break; + default: + config.iceTransportPolicy = 'all'; + break; + } + + switch (config.bundlePolicy) { + case 'balanced': + case 'max-compat': + case 'max-bundle': + break; + default: + config.bundlePolicy = 'balanced'; + break; + } + + config.iceServers = filterIceServers(config.iceServers || [], edgeVersion); + + this._iceGatherers = []; + if (config.iceCandidatePoolSize) { + for (var i = config.iceCandidatePoolSize; i > 0; i--) { + this._iceGatherers.push(new window.RTCIceGatherer({ + iceServers: config.iceServers, + gatherPolicy: config.iceTransportPolicy + })); + } + } else { + config.iceCandidatePoolSize = 0; + } + + this._config = config; + + // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ... + // everything that is needed to describe a SDP m-line. + this.transceivers = []; + + this._sdpSessionId = sdp.generateSessionId(); + this._sdpSessionVersion = 0; + + this._dtlsRole = undefined; // role for a=setup to use in answers. + + this._isClosed = false; + }; + + Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', { + configurable: true, + get: function() { + return this._localDescription; + } + }); + Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', { + configurable: true, + get: function() { + return this._remoteDescription; + } + }); + + // set up event handlers on prototype + RTCPeerConnection.prototype.onicecandidate = null; + RTCPeerConnection.prototype.onaddstream = null; + RTCPeerConnection.prototype.ontrack = null; + RTCPeerConnection.prototype.onremovestream = null; + RTCPeerConnection.prototype.onsignalingstatechange = null; + RTCPeerConnection.prototype.oniceconnectionstatechange = null; + RTCPeerConnection.prototype.onconnectionstatechange = null; + RTCPeerConnection.prototype.onicegatheringstatechange = null; + RTCPeerConnection.prototype.onnegotiationneeded = null; + RTCPeerConnection.prototype.ondatachannel = null; + + RTCPeerConnection.prototype._dispatchEvent = function(name, event) { + if (this._isClosed) { + return; + } + this.dispatchEvent(event); + if (typeof this['on' + name] === 'function') { + this['on' + name](event); + } + }; + + RTCPeerConnection.prototype._emitGatheringStateChange = function() { + var event = new Event('icegatheringstatechange'); + this._dispatchEvent('icegatheringstatechange', event); + }; + + RTCPeerConnection.prototype.getConfiguration = function() { + return this._config; + }; + + RTCPeerConnection.prototype.getLocalStreams = function() { + return this.localStreams; + }; + + RTCPeerConnection.prototype.getRemoteStreams = function() { + return this.remoteStreams; + }; + + // internal helper to create a transceiver object. + // (which is not yet the same as the WebRTC 1.0 transceiver) + RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) { + var hasBundleTransport = this.transceivers.length > 0; + var transceiver = { + track: null, + iceGatherer: null, + iceTransport: null, + dtlsTransport: null, + localCapabilities: null, + remoteCapabilities: null, + rtpSender: null, + rtpReceiver: null, + kind: kind, + mid: null, + sendEncodingParameters: null, + recvEncodingParameters: null, + stream: null, + associatedRemoteMediaStreams: [], + wantReceive: true + }; + if (this.usingBundle && hasBundleTransport) { + transceiver.iceTransport = this.transceivers[0].iceTransport; + transceiver.dtlsTransport = this.transceivers[0].dtlsTransport; + } else { + var transports = this._createIceAndDtlsTransports(); + transceiver.iceTransport = transports.iceTransport; + transceiver.dtlsTransport = transports.dtlsTransport; + } + if (!doNotAdd) { + this.transceivers.push(transceiver); + } + return transceiver; + }; + + RTCPeerConnection.prototype.addTrack = function(track, stream) { + if (this._isClosed) { + throw makeError('InvalidStateError', + 'Attempted to call addTrack on a closed peerconnection.'); + } + + var alreadyExists = this.transceivers.find(function(s) { + return s.track === track; + }); + + if (alreadyExists) { + throw makeError('InvalidAccessError', 'Track already exists.'); + } + + var transceiver; + for (var i = 0; i < this.transceivers.length; i++) { + if (!this.transceivers[i].track && + this.transceivers[i].kind === track.kind) { + transceiver = this.transceivers[i]; + } + } + if (!transceiver) { + transceiver = this._createTransceiver(track.kind); + } + + this._maybeFireNegotiationNeeded(); + + if (this.localStreams.indexOf(stream) === -1) { + this.localStreams.push(stream); + } + + transceiver.track = track; + transceiver.stream = stream; + transceiver.rtpSender = new window.RTCRtpSender(track, + transceiver.dtlsTransport); + return transceiver.rtpSender; + }; + + RTCPeerConnection.prototype.addStream = function(stream) { + var pc = this; + if (edgeVersion >= 15025) { + stream.getTracks().forEach(function(track) { + pc.addTrack(track, stream); + }); + } else { + // Clone is necessary for local demos mostly, attaching directly + // to two different senders does not work (build 10547). + // Fixed in 15025 (or earlier) + var clonedStream = stream.clone(); + stream.getTracks().forEach(function(track, idx) { + var clonedTrack = clonedStream.getTracks()[idx]; + track.addEventListener('enabled', function(event) { + clonedTrack.enabled = event.enabled; + }); + }); + clonedStream.getTracks().forEach(function(track) { + pc.addTrack(track, clonedStream); + }); + } + }; + + RTCPeerConnection.prototype.removeTrack = function(sender) { + if (this._isClosed) { + throw makeError('InvalidStateError', + 'Attempted to call removeTrack on a closed peerconnection.'); + } + + if (!(sender instanceof window.RTCRtpSender)) { + throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' + + 'does not implement interface RTCRtpSender.'); + } + + var transceiver = this.transceivers.find(function(t) { + return t.rtpSender === sender; + }); + + if (!transceiver) { + throw makeError('InvalidAccessError', + 'Sender was not created by this connection.'); + } + var stream = transceiver.stream; + + transceiver.rtpSender.stop(); + transceiver.rtpSender = null; + transceiver.track = null; + transceiver.stream = null; + + // remove the stream from the set of local streams + var localStreams = this.transceivers.map(function(t) { + return t.stream; + }); + if (localStreams.indexOf(stream) === -1 && + this.localStreams.indexOf(stream) > -1) { + this.localStreams.splice(this.localStreams.indexOf(stream), 1); + } + + this._maybeFireNegotiationNeeded(); + }; + + RTCPeerConnection.prototype.removeStream = function(stream) { + var pc = this; + stream.getTracks().forEach(function(track) { + var sender = pc.getSenders().find(function(s) { + return s.track === track; + }); + if (sender) { + pc.removeTrack(sender); + } + }); + }; + + RTCPeerConnection.prototype.getSenders = function() { + return this.transceivers.filter(function(transceiver) { + return !!transceiver.rtpSender; + }) + .map(function(transceiver) { + return transceiver.rtpSender; + }); + }; + + RTCPeerConnection.prototype.getReceivers = function() { + return this.transceivers.filter(function(transceiver) { + return !!transceiver.rtpReceiver; + }) + .map(function(transceiver) { + return transceiver.rtpReceiver; + }); + }; + + + RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex, + usingBundle) { + var pc = this; + if (usingBundle && sdpMLineIndex > 0) { + return this.transceivers[0].iceGatherer; + } else if (this._iceGatherers.length) { + return this._iceGatherers.shift(); + } + var iceGatherer = new window.RTCIceGatherer({ + iceServers: this._config.iceServers, + gatherPolicy: this._config.iceTransportPolicy + }); + Object.defineProperty(iceGatherer, 'state', + {value: 'new', writable: true} + ); + + this.transceivers[sdpMLineIndex].bufferedCandidateEvents = []; + this.transceivers[sdpMLineIndex].bufferCandidates = function(event) { + var end = !event.candidate || Object.keys(event.candidate).length === 0; + // polyfill since RTCIceGatherer.state is not implemented in + // Edge 10547 yet. + iceGatherer.state = end ? 'completed' : 'gathering'; + if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) { + pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event); + } + }; + iceGatherer.addEventListener('localcandidate', + this.transceivers[sdpMLineIndex].bufferCandidates); + return iceGatherer; + }; + + // start gathering from an RTCIceGatherer. + RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) { + var pc = this; + var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer; + if (iceGatherer.onlocalcandidate) { + return; + } + var bufferedCandidateEvents = + this.transceivers[sdpMLineIndex].bufferedCandidateEvents; + this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null; + iceGatherer.removeEventListener('localcandidate', + this.transceivers[sdpMLineIndex].bufferCandidates); + iceGatherer.onlocalcandidate = function(evt) { + if (pc.usingBundle && sdpMLineIndex > 0) { + // if we know that we use bundle we can drop candidates with + // ѕdpMLineIndex > 0. If we don't do this then our state gets + // confused since we dispose the extra ice gatherer. + return; + } + var event = new Event('icecandidate'); + event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex}; + + var cand = evt.candidate; + // Edge emits an empty object for RTCIceCandidateComplete‥ + var end = !cand || Object.keys(cand).length === 0; + if (end) { + // polyfill since RTCIceGatherer.state is not implemented in + // Edge 10547 yet. + if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') { + iceGatherer.state = 'completed'; + } + } else { + if (iceGatherer.state === 'new') { + iceGatherer.state = 'gathering'; + } + // RTCIceCandidate doesn't have a component, needs to be added + cand.component = 1; + // also the usernameFragment. TODO: update SDP to take both variants. + cand.ufrag = iceGatherer.getLocalParameters().usernameFragment; + + var serializedCandidate = sdp.writeCandidate(cand); + event.candidate = Object.assign(event.candidate, + sdp.parseCandidate(serializedCandidate)); + + event.candidate.candidate = serializedCandidate; + event.candidate.toJSON = function() { + return { + candidate: event.candidate.candidate, + sdpMid: event.candidate.sdpMid, + sdpMLineIndex: event.candidate.sdpMLineIndex, + usernameFragment: event.candidate.usernameFragment + }; + }; + } + + // update local description. + var sections = sdp.getMediaSections(pc._localDescription.sdp); + if (!end) { + sections[event.candidate.sdpMLineIndex] += + 'a=' + event.candidate.candidate + '\r\n'; + } else { + sections[event.candidate.sdpMLineIndex] += + 'a=end-of-candidates\r\n'; + } + pc._localDescription.sdp = + sdp.getDescription(pc._localDescription.sdp) + + sections.join(''); + var complete = pc.transceivers.every(function(transceiver) { + return transceiver.iceGatherer && + transceiver.iceGatherer.state === 'completed'; + }); + + if (pc.iceGatheringState !== 'gathering') { + pc.iceGatheringState = 'gathering'; + pc._emitGatheringStateChange(); + } + + // Emit candidate. Also emit null candidate when all gatherers are + // complete. + if (!end) { + pc._dispatchEvent('icecandidate', event); + } + if (complete) { + pc._dispatchEvent('icecandidate', new Event('icecandidate')); + pc.iceGatheringState = 'complete'; + pc._emitGatheringStateChange(); + } + }; + + // emit already gathered candidates. + window.setTimeout(function() { + bufferedCandidateEvents.forEach(function(e) { + iceGatherer.onlocalcandidate(e); + }); + }, 0); + }; + + // Create ICE transport and DTLS transport. + RTCPeerConnection.prototype._createIceAndDtlsTransports = function() { + var pc = this; + var iceTransport = new window.RTCIceTransport(null); + iceTransport.onicestatechange = function() { + pc._updateIceConnectionState(); + pc._updateConnectionState(); + }; + + var dtlsTransport = new window.RTCDtlsTransport(iceTransport); + dtlsTransport.ondtlsstatechange = function() { + pc._updateConnectionState(); + }; + dtlsTransport.onerror = function() { + // onerror does not set state to failed by itself. + Object.defineProperty(dtlsTransport, 'state', + {value: 'failed', writable: true}); + pc._updateConnectionState(); + }; + + return { + iceTransport: iceTransport, + dtlsTransport: dtlsTransport + }; + }; + + // Destroy ICE gatherer, ICE transport and DTLS transport. + // Without triggering the callbacks. + RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function( + sdpMLineIndex) { + var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer; + if (iceGatherer) { + delete iceGatherer.onlocalcandidate; + delete this.transceivers[sdpMLineIndex].iceGatherer; + } + var iceTransport = this.transceivers[sdpMLineIndex].iceTransport; + if (iceTransport) { + delete iceTransport.onicestatechange; + delete this.transceivers[sdpMLineIndex].iceTransport; + } + var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport; + if (dtlsTransport) { + delete dtlsTransport.ondtlsstatechange; + delete dtlsTransport.onerror; + delete this.transceivers[sdpMLineIndex].dtlsTransport; + } + }; + + // Start the RTP Sender and Receiver for a transceiver. + RTCPeerConnection.prototype._transceive = function(transceiver, + send, recv) { + var params = getCommonCapabilities(transceiver.localCapabilities, + transceiver.remoteCapabilities); + if (send && transceiver.rtpSender) { + params.encodings = transceiver.sendEncodingParameters; + params.rtcp = { + cname: sdp.localCName, + compound: transceiver.rtcpParameters.compound + }; + if (transceiver.recvEncodingParameters.length) { + params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc; + } + transceiver.rtpSender.send(params); + } + if (recv && transceiver.rtpReceiver && params.codecs.length > 0) { + // remove RTX field in Edge 14942 + if (transceiver.kind === 'video' + && transceiver.recvEncodingParameters + && edgeVersion < 15019) { + transceiver.recvEncodingParameters.forEach(function(p) { + delete p.rtx; + }); + } + if (transceiver.recvEncodingParameters.length) { + params.encodings = transceiver.recvEncodingParameters; + } else { + params.encodings = [{}]; + } + params.rtcp = { + compound: transceiver.rtcpParameters.compound + }; + if (transceiver.rtcpParameters.cname) { + params.rtcp.cname = transceiver.rtcpParameters.cname; + } + if (transceiver.sendEncodingParameters.length) { + params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc; + } + transceiver.rtpReceiver.receive(params); + } + }; + + RTCPeerConnection.prototype.setLocalDescription = function(description) { + var pc = this; + + // Note: pranswer is not supported. + if (['offer', 'answer'].indexOf(description.type) === -1) { + return Promise.reject(makeError('TypeError', + 'Unsupported type "' + description.type + '"')); + } + + if (!isActionAllowedInSignalingState('setLocalDescription', + description.type, pc.signalingState) || pc._isClosed) { + return Promise.reject(makeError('InvalidStateError', + 'Can not set local ' + description.type + + ' in state ' + pc.signalingState)); + } + + var sections; + var sessionpart; + if (description.type === 'offer') { + // VERY limited support for SDP munging. Limited to: + // * changing the order of codecs + sections = sdp.splitSections(description.sdp); + sessionpart = sections.shift(); + sections.forEach(function(mediaSection, sdpMLineIndex) { + var caps = sdp.parseRtpParameters(mediaSection); + pc.transceivers[sdpMLineIndex].localCapabilities = caps; + }); + + pc.transceivers.forEach(function(transceiver, sdpMLineIndex) { + pc._gather(transceiver.mid, sdpMLineIndex); + }); + } else if (description.type === 'answer') { + sections = sdp.splitSections(pc._remoteDescription.sdp); + sessionpart = sections.shift(); + var isIceLite = sdp.matchPrefix(sessionpart, + 'a=ice-lite').length > 0; + sections.forEach(function(mediaSection, sdpMLineIndex) { + var transceiver = pc.transceivers[sdpMLineIndex]; + var iceGatherer = transceiver.iceGatherer; + var iceTransport = transceiver.iceTransport; + var dtlsTransport = transceiver.dtlsTransport; + var localCapabilities = transceiver.localCapabilities; + var remoteCapabilities = transceiver.remoteCapabilities; + + // treat bundle-only as not-rejected. + var rejected = sdp.isRejected(mediaSection) && + sdp.matchPrefix(mediaSection, 'a=bundle-only').length === 0; + + if (!rejected && !transceiver.rejected) { + var remoteIceParameters = sdp.getIceParameters( + mediaSection, sessionpart); + var remoteDtlsParameters = sdp.getDtlsParameters( + mediaSection, sessionpart); + if (isIceLite) { + remoteDtlsParameters.role = 'server'; + } + + if (!pc.usingBundle || sdpMLineIndex === 0) { + pc._gather(transceiver.mid, sdpMLineIndex); + if (iceTransport.state === 'new') { + iceTransport.start(iceGatherer, remoteIceParameters, + isIceLite ? 'controlling' : 'controlled'); + } + if (dtlsTransport.state === 'new') { + dtlsTransport.start(remoteDtlsParameters); + } + } + + // Calculate intersection of capabilities. + var params = getCommonCapabilities(localCapabilities, + remoteCapabilities); + + // Start the RTCRtpSender. The RTCRtpReceiver for this + // transceiver has already been started in setRemoteDescription. + pc._transceive(transceiver, + params.codecs.length > 0, + false); + } + }); + } + + pc._localDescription = { + type: description.type, + sdp: description.sdp + }; + if (description.type === 'offer') { + pc._updateSignalingState('have-local-offer'); + } else { + pc._updateSignalingState('stable'); + } + + return Promise.resolve(); + }; + + RTCPeerConnection.prototype.setRemoteDescription = function(description) { + var pc = this; + + // Note: pranswer is not supported. + if (['offer', 'answer'].indexOf(description.type) === -1) { + return Promise.reject(makeError('TypeError', + 'Unsupported type "' + description.type + '"')); + } + + if (!isActionAllowedInSignalingState('setRemoteDescription', + description.type, pc.signalingState) || pc._isClosed) { + return Promise.reject(makeError('InvalidStateError', + 'Can not set remote ' + description.type + + ' in state ' + pc.signalingState)); + } + + var streams = {}; + pc.remoteStreams.forEach(function(stream) { + streams[stream.id] = stream; + }); + var receiverList = []; + var sections = sdp.splitSections(description.sdp); + var sessionpart = sections.shift(); + var isIceLite = sdp.matchPrefix(sessionpart, + 'a=ice-lite').length > 0; + var usingBundle = sdp.matchPrefix(sessionpart, + 'a=group:BUNDLE ').length > 0; + pc.usingBundle = usingBundle; + var iceOptions = sdp.matchPrefix(sessionpart, + 'a=ice-options:')[0]; + if (iceOptions) { + pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ') + .indexOf('trickle') >= 0; + } else { + pc.canTrickleIceCandidates = false; + } + + sections.forEach(function(mediaSection, sdpMLineIndex) { + var lines = sdp.splitLines(mediaSection); + var kind = sdp.getKind(mediaSection); + // treat bundle-only as not-rejected. + var rejected = sdp.isRejected(mediaSection) && + sdp.matchPrefix(mediaSection, 'a=bundle-only').length === 0; + var protocol = lines[0].substr(2).split(' ')[2]; + + var direction = sdp.getDirection(mediaSection, sessionpart); + var remoteMsid = sdp.parseMsid(mediaSection); + + var mid = sdp.getMid(mediaSection) || sdp.generateIdentifier(); + + // Reject datachannels which are not implemented yet. + if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' || + protocol === 'UDP/DTLS/SCTP'))) { + // TODO: this is dangerous in the case where a non-rejected m-line + // becomes rejected. + pc.transceivers[sdpMLineIndex] = { + mid: mid, + kind: kind, + protocol: protocol, + rejected: true + }; + return; + } + + if (!rejected && pc.transceivers[sdpMLineIndex] && + pc.transceivers[sdpMLineIndex].rejected) { + // recycle a rejected transceiver. + pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true); + } + + var transceiver; + var iceGatherer; + var iceTransport; + var dtlsTransport; + var rtpReceiver; + var sendEncodingParameters; + var recvEncodingParameters; + var localCapabilities; + + var track; + // FIXME: ensure the mediaSection has rtcp-mux set. + var remoteCapabilities = sdp.parseRtpParameters(mediaSection); + var remoteIceParameters; + var remoteDtlsParameters; + if (!rejected) { + remoteIceParameters = sdp.getIceParameters(mediaSection, + sessionpart); + remoteDtlsParameters = sdp.getDtlsParameters(mediaSection, + sessionpart); + remoteDtlsParameters.role = 'client'; + } + recvEncodingParameters = + sdp.parseRtpEncodingParameters(mediaSection); + + var rtcpParameters = sdp.parseRtcpParameters(mediaSection); + + var isComplete = sdp.matchPrefix(mediaSection, + 'a=end-of-candidates', sessionpart).length > 0; + var cands = sdp.matchPrefix(mediaSection, 'a=candidate:') + .map(function(cand) { + return sdp.parseCandidate(cand); + }) + .filter(function(cand) { + return cand.component === 1; + }); + + // Check if we can use BUNDLE and dispose transports. + if ((description.type === 'offer' || description.type === 'answer') && + !rejected && usingBundle && sdpMLineIndex > 0 && + pc.transceivers[sdpMLineIndex]) { + pc._disposeIceAndDtlsTransports(sdpMLineIndex); + pc.transceivers[sdpMLineIndex].iceGatherer = + pc.transceivers[0].iceGatherer; + pc.transceivers[sdpMLineIndex].iceTransport = + pc.transceivers[0].iceTransport; + pc.transceivers[sdpMLineIndex].dtlsTransport = + pc.transceivers[0].dtlsTransport; + if (pc.transceivers[sdpMLineIndex].rtpSender) { + pc.transceivers[sdpMLineIndex].rtpSender.setTransport( + pc.transceivers[0].dtlsTransport); + } + if (pc.transceivers[sdpMLineIndex].rtpReceiver) { + pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport( + pc.transceivers[0].dtlsTransport); + } + } + if (description.type === 'offer' && !rejected) { + transceiver = pc.transceivers[sdpMLineIndex] || + pc._createTransceiver(kind); + transceiver.mid = mid; + + if (!transceiver.iceGatherer) { + transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex, + usingBundle); + } + + if (cands.length && transceiver.iceTransport.state === 'new') { + if (isComplete && (!usingBundle || sdpMLineIndex === 0)) { + transceiver.iceTransport.setRemoteCandidates(cands); + } else { + cands.forEach(function(candidate) { + maybeAddCandidate(transceiver.iceTransport, candidate); + }); + } + } + + localCapabilities = window.RTCRtpReceiver.getCapabilities(kind); + + // filter RTX until additional stuff needed for RTX is implemented + // in adapter.js + if (edgeVersion < 15019) { + localCapabilities.codecs = localCapabilities.codecs.filter( + function(codec) { + return codec.name !== 'rtx'; + }); + } + + sendEncodingParameters = transceiver.sendEncodingParameters || [{ + ssrc: (2 * sdpMLineIndex + 2) * 1001 + }]; + + // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams + var isNewTrack = false; + if (direction === 'sendrecv' || direction === 'sendonly') { + isNewTrack = !transceiver.rtpReceiver; + rtpReceiver = transceiver.rtpReceiver || + new window.RTCRtpReceiver(transceiver.dtlsTransport, kind); + + if (isNewTrack) { + var stream; + track = rtpReceiver.track; + // FIXME: does not work with Plan B. + if (remoteMsid && remoteMsid.stream === '-') ; else if (remoteMsid) { + if (!streams[remoteMsid.stream]) { + streams[remoteMsid.stream] = new window.MediaStream(); + Object.defineProperty(streams[remoteMsid.stream], 'id', { + get: function() { + return remoteMsid.stream; + } + }); + } + Object.defineProperty(track, 'id', { + get: function() { + return remoteMsid.track; + } + }); + stream = streams[remoteMsid.stream]; + } else { + if (!streams.default) { + streams.default = new window.MediaStream(); + } + stream = streams.default; + } + if (stream) { + addTrackToStreamAndFireEvent(track, stream); + transceiver.associatedRemoteMediaStreams.push(stream); + } + receiverList.push([track, rtpReceiver, stream]); + } + } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) { + transceiver.associatedRemoteMediaStreams.forEach(function(s) { + var nativeTrack = s.getTracks().find(function(t) { + return t.id === transceiver.rtpReceiver.track.id; + }); + if (nativeTrack) { + removeTrackFromStreamAndFireEvent(nativeTrack, s); + } + }); + transceiver.associatedRemoteMediaStreams = []; + } + + transceiver.localCapabilities = localCapabilities; + transceiver.remoteCapabilities = remoteCapabilities; + transceiver.rtpReceiver = rtpReceiver; + transceiver.rtcpParameters = rtcpParameters; + transceiver.sendEncodingParameters = sendEncodingParameters; + transceiver.recvEncodingParameters = recvEncodingParameters; + + // Start the RTCRtpReceiver now. The RTPSender is started in + // setLocalDescription. + pc._transceive(pc.transceivers[sdpMLineIndex], + false, + isNewTrack); + } else if (description.type === 'answer' && !rejected) { + transceiver = pc.transceivers[sdpMLineIndex]; + iceGatherer = transceiver.iceGatherer; + iceTransport = transceiver.iceTransport; + dtlsTransport = transceiver.dtlsTransport; + rtpReceiver = transceiver.rtpReceiver; + sendEncodingParameters = transceiver.sendEncodingParameters; + localCapabilities = transceiver.localCapabilities; + + pc.transceivers[sdpMLineIndex].recvEncodingParameters = + recvEncodingParameters; + pc.transceivers[sdpMLineIndex].remoteCapabilities = + remoteCapabilities; + pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters; + + if (cands.length && iceTransport.state === 'new') { + if ((isIceLite || isComplete) && + (!usingBundle || sdpMLineIndex === 0)) { + iceTransport.setRemoteCandidates(cands); + } else { + cands.forEach(function(candidate) { + maybeAddCandidate(transceiver.iceTransport, candidate); + }); + } + } + + if (!usingBundle || sdpMLineIndex === 0) { + if (iceTransport.state === 'new') { + iceTransport.start(iceGatherer, remoteIceParameters, + 'controlling'); + } + if (dtlsTransport.state === 'new') { + dtlsTransport.start(remoteDtlsParameters); + } + } + + // If the offer contained RTX but the answer did not, + // remove RTX from sendEncodingParameters. + var commonCapabilities = getCommonCapabilities( + transceiver.localCapabilities, + transceiver.remoteCapabilities); + + var hasRtx = commonCapabilities.codecs.filter(function(c) { + return c.name.toLowerCase() === 'rtx'; + }).length; + if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) { + delete transceiver.sendEncodingParameters[0].rtx; + } + + pc._transceive(transceiver, + direction === 'sendrecv' || direction === 'recvonly', + direction === 'sendrecv' || direction === 'sendonly'); + + // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams + if (rtpReceiver && + (direction === 'sendrecv' || direction === 'sendonly')) { + track = rtpReceiver.track; + if (remoteMsid) { + if (!streams[remoteMsid.stream]) { + streams[remoteMsid.stream] = new window.MediaStream(); + } + addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]); + receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]); + } else { + if (!streams.default) { + streams.default = new window.MediaStream(); + } + addTrackToStreamAndFireEvent(track, streams.default); + receiverList.push([track, rtpReceiver, streams.default]); + } + } else { + // FIXME: actually the receiver should be created later. + delete transceiver.rtpReceiver; + } + } + }); + + if (pc._dtlsRole === undefined) { + pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive'; + } + + pc._remoteDescription = { + type: description.type, + sdp: description.sdp + }; + if (description.type === 'offer') { + pc._updateSignalingState('have-remote-offer'); + } else { + pc._updateSignalingState('stable'); + } + Object.keys(streams).forEach(function(sid) { + var stream = streams[sid]; + if (stream.getTracks().length) { + if (pc.remoteStreams.indexOf(stream) === -1) { + pc.remoteStreams.push(stream); + var event = new Event('addstream'); + event.stream = stream; + window.setTimeout(function() { + pc._dispatchEvent('addstream', event); + }); + } + + receiverList.forEach(function(item) { + var track = item[0]; + var receiver = item[1]; + if (stream.id !== item[2].id) { + return; + } + fireAddTrack(pc, track, receiver, [stream]); + }); + } + }); + receiverList.forEach(function(item) { + if (item[2]) { + return; + } + fireAddTrack(pc, item[0], item[1], []); + }); + + // check whether addIceCandidate({}) was called within four seconds after + // setRemoteDescription. + window.setTimeout(function() { + if (!(pc && pc.transceivers)) { + return; + } + pc.transceivers.forEach(function(transceiver) { + if (transceiver.iceTransport && + transceiver.iceTransport.state === 'new' && + transceiver.iceTransport.getRemoteCandidates().length > 0) { + console.warn('Timeout for addRemoteCandidate. Consider sending ' + + 'an end-of-candidates notification'); + transceiver.iceTransport.addRemoteCandidate({}); + } + }); + }, 4000); + + return Promise.resolve(); + }; + + RTCPeerConnection.prototype.close = function() { + this.transceivers.forEach(function(transceiver) { + /* not yet + if (transceiver.iceGatherer) { + transceiver.iceGatherer.close(); + } + */ + if (transceiver.iceTransport) { + transceiver.iceTransport.stop(); + } + if (transceiver.dtlsTransport) { + transceiver.dtlsTransport.stop(); + } + if (transceiver.rtpSender) { + transceiver.rtpSender.stop(); + } + if (transceiver.rtpReceiver) { + transceiver.rtpReceiver.stop(); + } + }); + // FIXME: clean up tracks, local streams, remote streams, etc + this._isClosed = true; + this._updateSignalingState('closed'); + }; + + // Update the signaling state. + RTCPeerConnection.prototype._updateSignalingState = function(newState) { + this.signalingState = newState; + var event = new Event('signalingstatechange'); + this._dispatchEvent('signalingstatechange', event); + }; + + // Determine whether to fire the negotiationneeded event. + RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() { + var pc = this; + if (this.signalingState !== 'stable' || this.needNegotiation === true) { + return; + } + this.needNegotiation = true; + window.setTimeout(function() { + if (pc.needNegotiation) { + pc.needNegotiation = false; + var event = new Event('negotiationneeded'); + pc._dispatchEvent('negotiationneeded', event); + } + }, 0); + }; + + // Update the ice connection state. + RTCPeerConnection.prototype._updateIceConnectionState = function() { + var newState; + var states = { + 'new': 0, + closed: 0, + checking: 0, + connected: 0, + completed: 0, + disconnected: 0, + failed: 0 + }; + this.transceivers.forEach(function(transceiver) { + if (transceiver.iceTransport && !transceiver.rejected) { + states[transceiver.iceTransport.state]++; + } + }); + + newState = 'new'; + if (states.failed > 0) { + newState = 'failed'; + } else if (states.checking > 0) { + newState = 'checking'; + } else if (states.disconnected > 0) { + newState = 'disconnected'; + } else if (states.new > 0) { + newState = 'new'; + } else if (states.connected > 0) { + newState = 'connected'; + } else if (states.completed > 0) { + newState = 'completed'; + } + + if (newState !== this.iceConnectionState) { + this.iceConnectionState = newState; + var event = new Event('iceconnectionstatechange'); + this._dispatchEvent('iceconnectionstatechange', event); + } + }; + + // Update the connection state. + RTCPeerConnection.prototype._updateConnectionState = function() { + var newState; + var states = { + 'new': 0, + closed: 0, + connecting: 0, + connected: 0, + completed: 0, + disconnected: 0, + failed: 0 + }; + this.transceivers.forEach(function(transceiver) { + if (transceiver.iceTransport && transceiver.dtlsTransport && + !transceiver.rejected) { + states[transceiver.iceTransport.state]++; + states[transceiver.dtlsTransport.state]++; + } + }); + // ICETransport.completed and connected are the same for this purpose. + states.connected += states.completed; + + newState = 'new'; + if (states.failed > 0) { + newState = 'failed'; + } else if (states.connecting > 0) { + newState = 'connecting'; + } else if (states.disconnected > 0) { + newState = 'disconnected'; + } else if (states.new > 0) { + newState = 'new'; + } else if (states.connected > 0) { + newState = 'connected'; + } + + if (newState !== this.connectionState) { + this.connectionState = newState; + var event = new Event('connectionstatechange'); + this._dispatchEvent('connectionstatechange', event); + } + }; + + RTCPeerConnection.prototype.createOffer = function() { + var pc = this; + + if (pc._isClosed) { + return Promise.reject(makeError('InvalidStateError', + 'Can not call createOffer after close')); + } + + var numAudioTracks = pc.transceivers.filter(function(t) { + return t.kind === 'audio'; + }).length; + var numVideoTracks = pc.transceivers.filter(function(t) { + return t.kind === 'video'; + }).length; + + // Determine number of audio and video tracks we need to send/recv. + var offerOptions = arguments[0]; + if (offerOptions) { + // Reject Chrome legacy constraints. + if (offerOptions.mandatory || offerOptions.optional) { + throw new TypeError( + 'Legacy mandatory/optional constraints not supported.'); + } + if (offerOptions.offerToReceiveAudio !== undefined) { + if (offerOptions.offerToReceiveAudio === true) { + numAudioTracks = 1; + } else if (offerOptions.offerToReceiveAudio === false) { + numAudioTracks = 0; + } else { + numAudioTracks = offerOptions.offerToReceiveAudio; + } + } + if (offerOptions.offerToReceiveVideo !== undefined) { + if (offerOptions.offerToReceiveVideo === true) { + numVideoTracks = 1; + } else if (offerOptions.offerToReceiveVideo === false) { + numVideoTracks = 0; + } else { + numVideoTracks = offerOptions.offerToReceiveVideo; + } + } + } + + pc.transceivers.forEach(function(transceiver) { + if (transceiver.kind === 'audio') { + numAudioTracks--; + if (numAudioTracks < 0) { + transceiver.wantReceive = false; + } + } else if (transceiver.kind === 'video') { + numVideoTracks--; + if (numVideoTracks < 0) { + transceiver.wantReceive = false; + } + } + }); + + // Create M-lines for recvonly streams. + while (numAudioTracks > 0 || numVideoTracks > 0) { + if (numAudioTracks > 0) { + pc._createTransceiver('audio'); + numAudioTracks--; + } + if (numVideoTracks > 0) { + pc._createTransceiver('video'); + numVideoTracks--; + } + } + + var sdp$1 = sdp.writeSessionBoilerplate(pc._sdpSessionId, + pc._sdpSessionVersion++); + pc.transceivers.forEach(function(transceiver, sdpMLineIndex) { + // For each track, create an ice gatherer, ice transport, + // dtls transport, potentially rtpsender and rtpreceiver. + var track = transceiver.track; + var kind = transceiver.kind; + var mid = transceiver.mid || sdp.generateIdentifier(); + transceiver.mid = mid; + + if (!transceiver.iceGatherer) { + transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex, + pc.usingBundle); + } + + var localCapabilities = window.RTCRtpSender.getCapabilities(kind); + // filter RTX until additional stuff needed for RTX is implemented + // in adapter.js + if (edgeVersion < 15019) { + localCapabilities.codecs = localCapabilities.codecs.filter( + function(codec) { + return codec.name !== 'rtx'; + }); + } + localCapabilities.codecs.forEach(function(codec) { + // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552 + // by adding level-asymmetry-allowed=1 + if (codec.name === 'H264' && + codec.parameters['level-asymmetry-allowed'] === undefined) { + codec.parameters['level-asymmetry-allowed'] = '1'; + } + + // for subsequent offers, we might have to re-use the payload + // type of the last offer. + if (transceiver.remoteCapabilities && + transceiver.remoteCapabilities.codecs) { + transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) { + if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() && + codec.clockRate === remoteCodec.clockRate) { + codec.preferredPayloadType = remoteCodec.payloadType; + } + }); + } + }); + localCapabilities.headerExtensions.forEach(function(hdrExt) { + var remoteExtensions = transceiver.remoteCapabilities && + transceiver.remoteCapabilities.headerExtensions || []; + remoteExtensions.forEach(function(rHdrExt) { + if (hdrExt.uri === rHdrExt.uri) { + hdrExt.id = rHdrExt.id; + } + }); + }); + + // generate an ssrc now, to be used later in rtpSender.send + var sendEncodingParameters = transceiver.sendEncodingParameters || [{ + ssrc: (2 * sdpMLineIndex + 1) * 1001 + }]; + if (track) { + // add RTX + if (edgeVersion >= 15019 && kind === 'video' && + !sendEncodingParameters[0].rtx) { + sendEncodingParameters[0].rtx = { + ssrc: sendEncodingParameters[0].ssrc + 1 + }; + } + } + + if (transceiver.wantReceive) { + transceiver.rtpReceiver = new window.RTCRtpReceiver( + transceiver.dtlsTransport, kind); + } + + transceiver.localCapabilities = localCapabilities; + transceiver.sendEncodingParameters = sendEncodingParameters; + }); + + // always offer BUNDLE and dispose on return if not supported. + if (pc._config.bundlePolicy !== 'max-compat') { + sdp$1 += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) { + return t.mid; + }).join(' ') + '\r\n'; + } + sdp$1 += 'a=ice-options:trickle\r\n'; + + pc.transceivers.forEach(function(transceiver, sdpMLineIndex) { + sdp$1 += writeMediaSection(transceiver, transceiver.localCapabilities, + 'offer', transceiver.stream, pc._dtlsRole); + sdp$1 += 'a=rtcp-rsize\r\n'; + + if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' && + (sdpMLineIndex === 0 || !pc.usingBundle)) { + transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) { + cand.component = 1; + sdp$1 += 'a=' + sdp.writeCandidate(cand) + '\r\n'; + }); + + if (transceiver.iceGatherer.state === 'completed') { + sdp$1 += 'a=end-of-candidates\r\n'; + } + } + }); + + var desc = new window.RTCSessionDescription({ + type: 'offer', + sdp: sdp$1 + }); + return Promise.resolve(desc); + }; + + RTCPeerConnection.prototype.createAnswer = function() { + var pc = this; + + if (pc._isClosed) { + return Promise.reject(makeError('InvalidStateError', + 'Can not call createAnswer after close')); + } + + if (!(pc.signalingState === 'have-remote-offer' || + pc.signalingState === 'have-local-pranswer')) { + return Promise.reject(makeError('InvalidStateError', + 'Can not call createAnswer in signalingState ' + pc.signalingState)); + } + + var sdp$1 = sdp.writeSessionBoilerplate(pc._sdpSessionId, + pc._sdpSessionVersion++); + if (pc.usingBundle) { + sdp$1 += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) { + return t.mid; + }).join(' ') + '\r\n'; + } + sdp$1 += 'a=ice-options:trickle\r\n'; + + var mediaSectionsInOffer = sdp.getMediaSections( + pc._remoteDescription.sdp).length; + pc.transceivers.forEach(function(transceiver, sdpMLineIndex) { + if (sdpMLineIndex + 1 > mediaSectionsInOffer) { + return; + } + if (transceiver.rejected) { + if (transceiver.kind === 'application') { + if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt + sdp$1 += 'm=application 0 DTLS/SCTP 5000\r\n'; + } else { + sdp$1 += 'm=application 0 ' + transceiver.protocol + + ' webrtc-datachannel\r\n'; + } + } else if (transceiver.kind === 'audio') { + sdp$1 += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\r\n' + + 'a=rtpmap:0 PCMU/8000\r\n'; + } else if (transceiver.kind === 'video') { + sdp$1 += 'm=video 0 UDP/TLS/RTP/SAVPF 120\r\n' + + 'a=rtpmap:120 VP8/90000\r\n'; + } + sdp$1 += 'c=IN IP4 0.0.0.0\r\n' + + 'a=inactive\r\n' + + 'a=mid:' + transceiver.mid + '\r\n'; + return; + } + + // FIXME: look at direction. + if (transceiver.stream) { + var localTrack; + if (transceiver.kind === 'audio') { + localTrack = transceiver.stream.getAudioTracks()[0]; + } else if (transceiver.kind === 'video') { + localTrack = transceiver.stream.getVideoTracks()[0]; + } + if (localTrack) { + // add RTX + if (edgeVersion >= 15019 && transceiver.kind === 'video' && + !transceiver.sendEncodingParameters[0].rtx) { + transceiver.sendEncodingParameters[0].rtx = { + ssrc: transceiver.sendEncodingParameters[0].ssrc + 1 + }; + } + } + } + + // Calculate intersection of capabilities. + var commonCapabilities = getCommonCapabilities( + transceiver.localCapabilities, + transceiver.remoteCapabilities); + + var hasRtx = commonCapabilities.codecs.filter(function(c) { + return c.name.toLowerCase() === 'rtx'; + }).length; + if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) { + delete transceiver.sendEncodingParameters[0].rtx; + } + + sdp$1 += writeMediaSection(transceiver, commonCapabilities, + 'answer', transceiver.stream, pc._dtlsRole); + if (transceiver.rtcpParameters && + transceiver.rtcpParameters.reducedSize) { + sdp$1 += 'a=rtcp-rsize\r\n'; + } + }); + + var desc = new window.RTCSessionDescription({ + type: 'answer', + sdp: sdp$1 + }); + return Promise.resolve(desc); + }; + + RTCPeerConnection.prototype.addIceCandidate = function(candidate) { + var pc = this; + var sections; + if (candidate && !(candidate.sdpMLineIndex !== undefined || + candidate.sdpMid)) { + return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required')); + } + + // TODO: needs to go into ops queue. + return new Promise(function(resolve, reject) { + if (!pc._remoteDescription) { + return reject(makeError('InvalidStateError', + 'Can not add ICE candidate without a remote description')); + } else if (!candidate || candidate.candidate === '') { + for (var j = 0; j < pc.transceivers.length; j++) { + if (pc.transceivers[j].rejected) { + continue; + } + pc.transceivers[j].iceTransport.addRemoteCandidate({}); + sections = sdp.getMediaSections(pc._remoteDescription.sdp); + sections[j] += 'a=end-of-candidates\r\n'; + pc._remoteDescription.sdp = + sdp.getDescription(pc._remoteDescription.sdp) + + sections.join(''); + if (pc.usingBundle) { + break; + } + } + } else { + var sdpMLineIndex = candidate.sdpMLineIndex; + if (candidate.sdpMid) { + for (var i = 0; i < pc.transceivers.length; i++) { + if (pc.transceivers[i].mid === candidate.sdpMid) { + sdpMLineIndex = i; + break; + } + } + } + var transceiver = pc.transceivers[sdpMLineIndex]; + if (transceiver) { + if (transceiver.rejected) { + return resolve(); + } + var cand = Object.keys(candidate.candidate).length > 0 ? + sdp.parseCandidate(candidate.candidate) : {}; + // Ignore Chrome's invalid candidates since Edge does not like them. + if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) { + return resolve(); + } + // Ignore RTCP candidates, we assume RTCP-MUX. + if (cand.component && cand.component !== 1) { + return resolve(); + } + // when using bundle, avoid adding candidates to the wrong + // ice transport. And avoid adding candidates added in the SDP. + if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 && + transceiver.iceTransport !== pc.transceivers[0].iceTransport)) { + if (!maybeAddCandidate(transceiver.iceTransport, cand)) { + return reject(makeError('OperationError', + 'Can not add ICE candidate')); + } + } + + // update the remoteDescription. + var candidateString = candidate.candidate.trim(); + if (candidateString.indexOf('a=') === 0) { + candidateString = candidateString.substr(2); + } + sections = sdp.getMediaSections(pc._remoteDescription.sdp); + sections[sdpMLineIndex] += 'a=' + + (cand.type ? candidateString : 'end-of-candidates') + + '\r\n'; + pc._remoteDescription.sdp = + sdp.getDescription(pc._remoteDescription.sdp) + + sections.join(''); + } else { + return reject(makeError('OperationError', + 'Can not add ICE candidate')); + } + } + resolve(); + }); + }; + + RTCPeerConnection.prototype.getStats = function(selector) { + if (selector && selector instanceof window.MediaStreamTrack) { + var senderOrReceiver = null; + this.transceivers.forEach(function(transceiver) { + if (transceiver.rtpSender && + transceiver.rtpSender.track === selector) { + senderOrReceiver = transceiver.rtpSender; + } else if (transceiver.rtpReceiver && + transceiver.rtpReceiver.track === selector) { + senderOrReceiver = transceiver.rtpReceiver; + } + }); + if (!senderOrReceiver) { + throw makeError('InvalidAccessError', 'Invalid selector.'); + } + return senderOrReceiver.getStats(); + } + + var promises = []; + this.transceivers.forEach(function(transceiver) { + ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport', + 'dtlsTransport'].forEach(function(method) { + if (transceiver[method]) { + promises.push(transceiver[method].getStats()); + } + }); + }); + return Promise.all(promises).then(function(allStats) { + var results = new Map(); + allStats.forEach(function(stats) { + stats.forEach(function(stat) { + results.set(stat.id, stat); + }); + }); + return results; + }); + }; + + // fix low-level stat names and return Map instead of object. + var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer', + 'RTCIceTransport', 'RTCDtlsTransport']; + ortcObjects.forEach(function(ortcObjectName) { + var obj = window[ortcObjectName]; + if (obj && obj.prototype && obj.prototype.getStats) { + var nativeGetstats = obj.prototype.getStats; + obj.prototype.getStats = function() { + return nativeGetstats.apply(this) + .then(function(nativeStats) { + var mapStats = new Map(); + Object.keys(nativeStats).forEach(function(id) { + nativeStats[id].type = fixStatsType(nativeStats[id]); + mapStats.set(id, nativeStats[id]); + }); + return mapStats; + }); + }; + } + }); + + // legacy callback shims. Should be moved to adapter.js some days. + var methods = ['createOffer', 'createAnswer']; + methods.forEach(function(method) { + var nativeMethod = RTCPeerConnection.prototype[method]; + RTCPeerConnection.prototype[method] = function() { + var args = arguments; + if (typeof args[0] === 'function' || + typeof args[1] === 'function') { // legacy + return nativeMethod.apply(this, [arguments[2]]) + .then(function(description) { + if (typeof args[0] === 'function') { + args[0].apply(null, [description]); + } + }, function(error) { + if (typeof args[1] === 'function') { + args[1].apply(null, [error]); + } + }); + } + return nativeMethod.apply(this, arguments); + }; + }); + + methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']; + methods.forEach(function(method) { + var nativeMethod = RTCPeerConnection.prototype[method]; + RTCPeerConnection.prototype[method] = function() { + var args = arguments; + if (typeof args[1] === 'function' || + typeof args[2] === 'function') { // legacy + return nativeMethod.apply(this, arguments) + .then(function() { + if (typeof args[1] === 'function') { + args[1].apply(null); + } + }, function(error) { + if (typeof args[2] === 'function') { + args[2].apply(null, [error]); + } + }); + } + return nativeMethod.apply(this, arguments); + }; + }); + + // getStats is special. It doesn't have a spec legacy method yet we support + // getStats(something, cb) without error callbacks. + ['getStats'].forEach(function(method) { + var nativeMethod = RTCPeerConnection.prototype[method]; + RTCPeerConnection.prototype[method] = function() { + var args = arguments; + if (typeof args[1] === 'function') { + return nativeMethod.apply(this, arguments) + .then(function() { + if (typeof args[1] === 'function') { + args[1].apply(null); + } + }); + } + return nativeMethod.apply(this, arguments); + }; + }); + + return RTCPeerConnection; + }; + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimGetUserMedia$2(window) { + const navigator = window && window.navigator; + + const shimError_ = function(e) { + return { + name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name, + message: e.message, + constraint: e.constraint, + toString() { + return this.name; + } + }; + }; + + // getUserMedia error shim. + const origGetUserMedia = navigator.mediaDevices.getUserMedia. + bind(navigator.mediaDevices); + navigator.mediaDevices.getUserMedia = function(c) { + return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e))); + }; + } + + /* + * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimGetDisplayMedia$1(window) { + if (!('getDisplayMedia' in window.navigator)) { + return; + } + if (!(window.navigator.mediaDevices)) { + return; + } + if (window.navigator.mediaDevices && + 'getDisplayMedia' in window.navigator.mediaDevices) { + return; + } + window.navigator.mediaDevices.getDisplayMedia = + window.navigator.getDisplayMedia.bind(window.navigator); + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimPeerConnection$1(window, browserDetails) { + if (window.RTCIceGatherer) { + if (!window.RTCIceCandidate) { + window.RTCIceCandidate = function RTCIceCandidate(args) { + return args; + }; + } + if (!window.RTCSessionDescription) { + window.RTCSessionDescription = function RTCSessionDescription(args) { + return args; + }; + } + // this adds an additional event listener to MediaStrackTrack that signals + // when a tracks enabled property was changed. Workaround for a bug in + // addStream, see below. No longer required in 15025+ + if (browserDetails.version < 15025) { + const origMSTEnabled = Object.getOwnPropertyDescriptor( + window.MediaStreamTrack.prototype, 'enabled'); + Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', { + set(value) { + origMSTEnabled.set.call(this, value); + const ev = new Event('enabled'); + ev.enabled = value; + this.dispatchEvent(ev); + } + }); + } + } + + // ORTC defines the DTMF sender a bit different. + // https://github.com/w3c/ortc/issues/714 + if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) { + Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', { + get() { + if (this._dtmf === undefined) { + if (this.track.kind === 'audio') { + this._dtmf = new window.RTCDtmfSender(this); + } else if (this.track.kind === 'video') { + this._dtmf = null; + } + } + return this._dtmf; + } + }); + } + // Edge currently only implements the RTCDtmfSender, not the + // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2* + if (window.RTCDtmfSender && !window.RTCDTMFSender) { + window.RTCDTMFSender = window.RTCDtmfSender; + } + + const RTCPeerConnectionShim = rtcpeerconnection(window, + browserDetails.version); + window.RTCPeerConnection = function RTCPeerConnection(config) { + if (config && config.iceServers) { + config.iceServers = filterIceServers$1(config.iceServers, + browserDetails.version); + log$1('ICE servers after filtering:', config.iceServers); + } + return new RTCPeerConnectionShim(config); + }; + window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype; + } + + function shimReplaceTrack(window) { + // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614 + if (window.RTCRtpSender && + !('replaceTrack' in window.RTCRtpSender.prototype)) { + window.RTCRtpSender.prototype.replaceTrack = + window.RTCRtpSender.prototype.setTrack; + } + } + + var edgeShim = /*#__PURE__*/Object.freeze({ + __proto__: null, + shimPeerConnection: shimPeerConnection$1, + shimReplaceTrack: shimReplaceTrack, + shimGetUserMedia: shimGetUserMedia$2, + shimGetDisplayMedia: shimGetDisplayMedia$1 + }); + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimGetUserMedia$1(window, browserDetails) { + const navigator = window && window.navigator; + const MediaStreamTrack = window && window.MediaStreamTrack; + + navigator.getUserMedia = function(constraints, onSuccess, onError) { + // Replace Firefox 44+'s deprecation warning with unprefixed version. + deprecated('navigator.getUserMedia', + 'navigator.mediaDevices.getUserMedia'); + navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError); + }; + + if (!(browserDetails.version > 55 && + 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) { + const remap = function(obj, a, b) { + if (a in obj && !(b in obj)) { + obj[b] = obj[a]; + delete obj[a]; + } + }; + + const nativeGetUserMedia = navigator.mediaDevices.getUserMedia. + bind(navigator.mediaDevices); + navigator.mediaDevices.getUserMedia = function(c) { + if (typeof c === 'object' && typeof c.audio === 'object') { + c = JSON.parse(JSON.stringify(c)); + remap(c.audio, 'autoGainControl', 'mozAutoGainControl'); + remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression'); + } + return nativeGetUserMedia(c); + }; + + if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) { + const nativeGetSettings = MediaStreamTrack.prototype.getSettings; + MediaStreamTrack.prototype.getSettings = function() { + const obj = nativeGetSettings.apply(this, arguments); + remap(obj, 'mozAutoGainControl', 'autoGainControl'); + remap(obj, 'mozNoiseSuppression', 'noiseSuppression'); + return obj; + }; + } + + if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) { + const nativeApplyConstraints = + MediaStreamTrack.prototype.applyConstraints; + MediaStreamTrack.prototype.applyConstraints = function(c) { + if (this.kind === 'audio' && typeof c === 'object') { + c = JSON.parse(JSON.stringify(c)); + remap(c, 'autoGainControl', 'mozAutoGainControl'); + remap(c, 'noiseSuppression', 'mozNoiseSuppression'); + } + return nativeApplyConstraints.apply(this, [c]); + }; + } + } + } + + /* + * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimGetDisplayMedia(window, preferredMediaSource) { + if (window.navigator.mediaDevices && + 'getDisplayMedia' in window.navigator.mediaDevices) { + return; + } + if (!(window.navigator.mediaDevices)) { + return; + } + window.navigator.mediaDevices.getDisplayMedia = + function getDisplayMedia(constraints) { + if (!(constraints && constraints.video)) { + const err = new DOMException('getDisplayMedia without video ' + + 'constraints is undefined'); + err.name = 'NotFoundError'; + // from https://heycam.github.io/webidl/#idl-DOMException-error-names + err.code = 8; + return Promise.reject(err); + } + if (constraints.video === true) { + constraints.video = {mediaSource: preferredMediaSource}; + } else { + constraints.video.mediaSource = preferredMediaSource; + } + return window.navigator.mediaDevices.getUserMedia(constraints); + }; + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimOnTrack(window) { + if (typeof window === 'object' && window.RTCTrackEvent && + ('receiver' in window.RTCTrackEvent.prototype) && + !('transceiver' in window.RTCTrackEvent.prototype)) { + Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', { + get() { + return {receiver: this.receiver}; + } + }); + } + } + + function shimPeerConnection(window, browserDetails) { + if (typeof window !== 'object' || + !(window.RTCPeerConnection || window.mozRTCPeerConnection)) { + return; // probably media.peerconnection.enabled=false in about:config + } + if (!window.RTCPeerConnection && window.mozRTCPeerConnection) { + // very basic support for old versions. + window.RTCPeerConnection = window.mozRTCPeerConnection; + } + + if (browserDetails.version < 53) { + // shim away need for obsolete RTCIceCandidate/RTCSessionDescription. + ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] + .forEach(function(method) { + const nativeMethod = window.RTCPeerConnection.prototype[method]; + const methodObj = {[method]() { + arguments[0] = new ((method === 'addIceCandidate') ? + window.RTCIceCandidate : + window.RTCSessionDescription)(arguments[0]); + return nativeMethod.apply(this, arguments); + }}; + window.RTCPeerConnection.prototype[method] = methodObj[method]; + }); + } + + const modernStatsTypes = { + inboundrtp: 'inbound-rtp', + outboundrtp: 'outbound-rtp', + candidatepair: 'candidate-pair', + localcandidate: 'local-candidate', + remotecandidate: 'remote-candidate' + }; + + const nativeGetStats = window.RTCPeerConnection.prototype.getStats; + window.RTCPeerConnection.prototype.getStats = function getStats() { + const [selector, onSucc, onErr] = arguments; + return nativeGetStats.apply(this, [selector || null]) + .then(stats => { + if (browserDetails.version < 53 && !onSucc) { + // Shim only promise getStats with spec-hyphens in type names + // Leave callback version alone; misc old uses of forEach before Map + try { + stats.forEach(stat => { + stat.type = modernStatsTypes[stat.type] || stat.type; + }); + } catch (e) { + if (e.name !== 'TypeError') { + throw e; + } + // Avoid TypeError: "type" is read-only, in old versions. 34-43ish + stats.forEach((stat, i) => { + stats.set(i, Object.assign({}, stat, { + type: modernStatsTypes[stat.type] || stat.type + })); + }); + } + } + return stats; + }) + .then(onSucc, onErr); + }; + } + + function shimSenderGetStats(window) { + if (!(typeof window === 'object' && window.RTCPeerConnection && + window.RTCRtpSender)) { + return; + } + if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) { + return; + } + const origGetSenders = window.RTCPeerConnection.prototype.getSenders; + if (origGetSenders) { + window.RTCPeerConnection.prototype.getSenders = function getSenders() { + const senders = origGetSenders.apply(this, []); + senders.forEach(sender => sender._pc = this); + return senders; + }; + } + + const origAddTrack = window.RTCPeerConnection.prototype.addTrack; + if (origAddTrack) { + window.RTCPeerConnection.prototype.addTrack = function addTrack() { + const sender = origAddTrack.apply(this, arguments); + sender._pc = this; + return sender; + }; + } + window.RTCRtpSender.prototype.getStats = function getStats() { + return this.track ? this._pc.getStats(this.track) : + Promise.resolve(new Map()); + }; + } + + function shimReceiverGetStats(window) { + if (!(typeof window === 'object' && window.RTCPeerConnection && + window.RTCRtpSender)) { + return; + } + if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) { + return; + } + const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers; + if (origGetReceivers) { + window.RTCPeerConnection.prototype.getReceivers = function getReceivers() { + const receivers = origGetReceivers.apply(this, []); + receivers.forEach(receiver => receiver._pc = this); + return receivers; + }; + } + wrapPeerConnectionEvent(window, 'track', e => { + e.receiver._pc = e.srcElement; + return e; + }); + window.RTCRtpReceiver.prototype.getStats = function getStats() { + return this._pc.getStats(this.track); + }; + } + + function shimRemoveStream(window) { + if (!window.RTCPeerConnection || + 'removeStream' in window.RTCPeerConnection.prototype) { + return; + } + window.RTCPeerConnection.prototype.removeStream = + function removeStream(stream) { + deprecated('removeStream', 'removeTrack'); + this.getSenders().forEach(sender => { + if (sender.track && stream.getTracks().includes(sender.track)) { + this.removeTrack(sender); + } + }); + }; + } + + function shimRTCDataChannel(window) { + // rename DataChannel to RTCDataChannel (native fix in FF60): + // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851 + if (window.DataChannel && !window.RTCDataChannel) { + window.RTCDataChannel = window.DataChannel; + } + } + + function shimAddTransceiver(window) { + // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647 + // Firefox ignores the init sendEncodings options passed to addTransceiver + // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918 + if (!(typeof window === 'object' && window.RTCPeerConnection)) { + return; + } + const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver; + if (origAddTransceiver) { + window.RTCPeerConnection.prototype.addTransceiver = + function addTransceiver() { + this.setParametersPromises = []; + const initParameters = arguments[1]; + const shouldPerformCheck = initParameters && + 'sendEncodings' in initParameters; + if (shouldPerformCheck) { + // If sendEncodings params are provided, validate grammar + initParameters.sendEncodings.forEach((encodingParam) => { + if ('rid' in encodingParam) { + const ridRegex = /^[a-z0-9]{0,16}$/i; + if (!ridRegex.test(encodingParam.rid)) { + throw new TypeError('Invalid RID value provided.'); + } + } + if ('scaleResolutionDownBy' in encodingParam) { + if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) { + throw new RangeError('scale_resolution_down_by must be >= 1.0'); + } + } + if ('maxFramerate' in encodingParam) { + if (!(parseFloat(encodingParam.maxFramerate) >= 0)) { + throw new RangeError('max_framerate must be >= 0.0'); + } + } + }); + } + const transceiver = origAddTransceiver.apply(this, arguments); + if (shouldPerformCheck) { + // Check if the init options were applied. If not we do this in an + // asynchronous way and save the promise reference in a global object. + // This is an ugly hack, but at the same time is way more robust than + // checking the sender parameters before and after the createOffer + // Also note that after the createoffer we are not 100% sure that + // the params were asynchronously applied so we might miss the + // opportunity to recreate offer. + const {sender} = transceiver; + const params = sender.getParameters(); + if (!('encodings' in params) || + // Avoid being fooled by patched getParameters() below. + (params.encodings.length === 1 && + Object.keys(params.encodings[0]).length === 0)) { + params.encodings = initParameters.sendEncodings; + sender.sendEncodings = initParameters.sendEncodings; + this.setParametersPromises.push(sender.setParameters(params) + .then(() => { + delete sender.sendEncodings; + }).catch(() => { + delete sender.sendEncodings; + }) + ); + } + } + return transceiver; + }; + } + } + + function shimGetParameters(window) { + if (!(typeof window === 'object' && window.RTCRtpSender)) { + return; + } + const origGetParameters = window.RTCRtpSender.prototype.getParameters; + if (origGetParameters) { + window.RTCRtpSender.prototype.getParameters = + function getParameters() { + const params = origGetParameters.apply(this, arguments); + if (!('encodings' in params)) { + params.encodings = [].concat(this.sendEncodings || [{}]); + } + return params; + }; + } + } + + function shimCreateOffer(window) { + // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647 + // Firefox ignores the init sendEncodings options passed to addTransceiver + // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918 + if (!(typeof window === 'object' && window.RTCPeerConnection)) { + return; + } + const origCreateOffer = window.RTCPeerConnection.prototype.createOffer; + window.RTCPeerConnection.prototype.createOffer = function createOffer() { + if (this.setParametersPromises && this.setParametersPromises.length) { + return Promise.all(this.setParametersPromises) + .then(() => { + return origCreateOffer.apply(this, arguments); + }) + .finally(() => { + this.setParametersPromises = []; + }); + } + return origCreateOffer.apply(this, arguments); + }; + } + + function shimCreateAnswer(window) { + // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647 + // Firefox ignores the init sendEncodings options passed to addTransceiver + // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918 + if (!(typeof window === 'object' && window.RTCPeerConnection)) { + return; + } + const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer; + window.RTCPeerConnection.prototype.createAnswer = function createAnswer() { + if (this.setParametersPromises && this.setParametersPromises.length) { + return Promise.all(this.setParametersPromises) + .then(() => { + return origCreateAnswer.apply(this, arguments); + }) + .finally(() => { + this.setParametersPromises = []; + }); + } + return origCreateAnswer.apply(this, arguments); + }; + } + + var firefoxShim = /*#__PURE__*/Object.freeze({ + __proto__: null, + shimOnTrack: shimOnTrack, + shimPeerConnection: shimPeerConnection, + shimSenderGetStats: shimSenderGetStats, + shimReceiverGetStats: shimReceiverGetStats, + shimRemoveStream: shimRemoveStream, + shimRTCDataChannel: shimRTCDataChannel, + shimAddTransceiver: shimAddTransceiver, + shimGetParameters: shimGetParameters, + shimCreateOffer: shimCreateOffer, + shimCreateAnswer: shimCreateAnswer, + shimGetUserMedia: shimGetUserMedia$1, + shimGetDisplayMedia: shimGetDisplayMedia + }); + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimLocalStreamsAPI(window) { + if (typeof window !== 'object' || !window.RTCPeerConnection) { + return; + } + if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) { + window.RTCPeerConnection.prototype.getLocalStreams = + function getLocalStreams() { + if (!this._localStreams) { + this._localStreams = []; + } + return this._localStreams; + }; + } + if (!('addStream' in window.RTCPeerConnection.prototype)) { + const _addTrack = window.RTCPeerConnection.prototype.addTrack; + window.RTCPeerConnection.prototype.addStream = function addStream(stream) { + if (!this._localStreams) { + this._localStreams = []; + } + if (!this._localStreams.includes(stream)) { + this._localStreams.push(stream); + } + // Try to emulate Chrome's behaviour of adding in audio-video order. + // Safari orders by track id. + stream.getAudioTracks().forEach(track => _addTrack.call(this, track, + stream)); + stream.getVideoTracks().forEach(track => _addTrack.call(this, track, + stream)); + }; + + window.RTCPeerConnection.prototype.addTrack = + function addTrack(track, ...streams) { + if (streams) { + streams.forEach((stream) => { + if (!this._localStreams) { + this._localStreams = [stream]; + } else if (!this._localStreams.includes(stream)) { + this._localStreams.push(stream); + } + }); + } + return _addTrack.apply(this, arguments); + }; + } + if (!('removeStream' in window.RTCPeerConnection.prototype)) { + window.RTCPeerConnection.prototype.removeStream = + function removeStream(stream) { + if (!this._localStreams) { + this._localStreams = []; + } + const index = this._localStreams.indexOf(stream); + if (index === -1) { + return; + } + this._localStreams.splice(index, 1); + const tracks = stream.getTracks(); + this.getSenders().forEach(sender => { + if (tracks.includes(sender.track)) { + this.removeTrack(sender); + } + }); + }; + } + } + + function shimRemoteStreamsAPI(window) { + if (typeof window !== 'object' || !window.RTCPeerConnection) { + return; + } + if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) { + window.RTCPeerConnection.prototype.getRemoteStreams = + function getRemoteStreams() { + return this._remoteStreams ? this._remoteStreams : []; + }; + } + if (!('onaddstream' in window.RTCPeerConnection.prototype)) { + Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', { + get() { + return this._onaddstream; + }, + set(f) { + if (this._onaddstream) { + this.removeEventListener('addstream', this._onaddstream); + this.removeEventListener('track', this._onaddstreampoly); + } + this.addEventListener('addstream', this._onaddstream = f); + this.addEventListener('track', this._onaddstreampoly = (e) => { + e.streams.forEach(stream => { + if (!this._remoteStreams) { + this._remoteStreams = []; + } + if (this._remoteStreams.includes(stream)) { + return; + } + this._remoteStreams.push(stream); + const event = new Event('addstream'); + event.stream = stream; + this.dispatchEvent(event); + }); + }); + } + }); + const origSetRemoteDescription = + window.RTCPeerConnection.prototype.setRemoteDescription; + window.RTCPeerConnection.prototype.setRemoteDescription = + function setRemoteDescription() { + const pc = this; + if (!this._onaddstreampoly) { + this.addEventListener('track', this._onaddstreampoly = function(e) { + e.streams.forEach(stream => { + if (!pc._remoteStreams) { + pc._remoteStreams = []; + } + if (pc._remoteStreams.indexOf(stream) >= 0) { + return; + } + pc._remoteStreams.push(stream); + const event = new Event('addstream'); + event.stream = stream; + pc.dispatchEvent(event); + }); + }); + } + return origSetRemoteDescription.apply(pc, arguments); + }; + } + } + + function shimCallbacksAPI(window) { + if (typeof window !== 'object' || !window.RTCPeerConnection) { + return; + } + const prototype = window.RTCPeerConnection.prototype; + const origCreateOffer = prototype.createOffer; + const origCreateAnswer = prototype.createAnswer; + const setLocalDescription = prototype.setLocalDescription; + const setRemoteDescription = prototype.setRemoteDescription; + const addIceCandidate = prototype.addIceCandidate; + + prototype.createOffer = + function createOffer(successCallback, failureCallback) { + const options = (arguments.length >= 2) ? arguments[2] : arguments[0]; + const promise = origCreateOffer.apply(this, [options]); + if (!failureCallback) { + return promise; + } + promise.then(successCallback, failureCallback); + return Promise.resolve(); + }; + + prototype.createAnswer = + function createAnswer(successCallback, failureCallback) { + const options = (arguments.length >= 2) ? arguments[2] : arguments[0]; + const promise = origCreateAnswer.apply(this, [options]); + if (!failureCallback) { + return promise; + } + promise.then(successCallback, failureCallback); + return Promise.resolve(); + }; + + let withCallback = function(description, successCallback, failureCallback) { + const promise = setLocalDescription.apply(this, [description]); + if (!failureCallback) { + return promise; + } + promise.then(successCallback, failureCallback); + return Promise.resolve(); + }; + prototype.setLocalDescription = withCallback; + + withCallback = function(description, successCallback, failureCallback) { + const promise = setRemoteDescription.apply(this, [description]); + if (!failureCallback) { + return promise; + } + promise.then(successCallback, failureCallback); + return Promise.resolve(); + }; + prototype.setRemoteDescription = withCallback; + + withCallback = function(candidate, successCallback, failureCallback) { + const promise = addIceCandidate.apply(this, [candidate]); + if (!failureCallback) { + return promise; + } + promise.then(successCallback, failureCallback); + return Promise.resolve(); + }; + prototype.addIceCandidate = withCallback; + } + + function shimGetUserMedia(window) { + const navigator = window && window.navigator; + + if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) { + // shim not needed in Safari 12.1 + const mediaDevices = navigator.mediaDevices; + const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices); + navigator.mediaDevices.getUserMedia = (constraints) => { + return _getUserMedia(shimConstraints(constraints)); + }; + } + + if (!navigator.getUserMedia && navigator.mediaDevices && + navigator.mediaDevices.getUserMedia) { + navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) { + navigator.mediaDevices.getUserMedia(constraints) + .then(cb, errcb); + }.bind(navigator); + } + } + + function shimConstraints(constraints) { + if (constraints && constraints.video !== undefined) { + return Object.assign({}, + constraints, + {video: compactObject(constraints.video)} + ); + } + + return constraints; + } + + function shimRTCIceServerUrls(window) { + if (!window.RTCPeerConnection) { + return; + } + // migrate from non-spec RTCIceServer.url to RTCIceServer.urls + const OrigPeerConnection = window.RTCPeerConnection; + window.RTCPeerConnection = + function RTCPeerConnection(pcConfig, pcConstraints) { + if (pcConfig && pcConfig.iceServers) { + const newIceServers = []; + for (let i = 0; i < pcConfig.iceServers.length; i++) { + let server = pcConfig.iceServers[i]; + if (!server.hasOwnProperty('urls') && + server.hasOwnProperty('url')) { + deprecated('RTCIceServer.url', 'RTCIceServer.urls'); + server = JSON.parse(JSON.stringify(server)); + server.urls = server.url; + delete server.url; + newIceServers.push(server); + } else { + newIceServers.push(pcConfig.iceServers[i]); + } + } + pcConfig.iceServers = newIceServers; + } + return new OrigPeerConnection(pcConfig, pcConstraints); + }; + window.RTCPeerConnection.prototype = OrigPeerConnection.prototype; + // wrap static methods. Currently just generateCertificate. + if ('generateCertificate' in OrigPeerConnection) { + Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', { + get() { + return OrigPeerConnection.generateCertificate; + } + }); + } + } + + function shimTrackEventTransceiver(window) { + // Add event.transceiver member over deprecated event.receiver + if (typeof window === 'object' && window.RTCTrackEvent && + 'receiver' in window.RTCTrackEvent.prototype && + !('transceiver' in window.RTCTrackEvent.prototype)) { + Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', { + get() { + return {receiver: this.receiver}; + } + }); + } + } + + function shimCreateOfferLegacy(window) { + const origCreateOffer = window.RTCPeerConnection.prototype.createOffer; + window.RTCPeerConnection.prototype.createOffer = + function createOffer(offerOptions) { + if (offerOptions) { + if (typeof offerOptions.offerToReceiveAudio !== 'undefined') { + // support bit values + offerOptions.offerToReceiveAudio = + !!offerOptions.offerToReceiveAudio; + } + const audioTransceiver = this.getTransceivers().find(transceiver => + transceiver.receiver.track.kind === 'audio'); + if (offerOptions.offerToReceiveAudio === false && audioTransceiver) { + if (audioTransceiver.direction === 'sendrecv') { + if (audioTransceiver.setDirection) { + audioTransceiver.setDirection('sendonly'); + } else { + audioTransceiver.direction = 'sendonly'; + } + } else if (audioTransceiver.direction === 'recvonly') { + if (audioTransceiver.setDirection) { + audioTransceiver.setDirection('inactive'); + } else { + audioTransceiver.direction = 'inactive'; + } + } + } else if (offerOptions.offerToReceiveAudio === true && + !audioTransceiver) { + this.addTransceiver('audio'); + } + + if (typeof offerOptions.offerToReceiveVideo !== 'undefined') { + // support bit values + offerOptions.offerToReceiveVideo = + !!offerOptions.offerToReceiveVideo; + } + const videoTransceiver = this.getTransceivers().find(transceiver => + transceiver.receiver.track.kind === 'video'); + if (offerOptions.offerToReceiveVideo === false && videoTransceiver) { + if (videoTransceiver.direction === 'sendrecv') { + if (videoTransceiver.setDirection) { + videoTransceiver.setDirection('sendonly'); + } else { + videoTransceiver.direction = 'sendonly'; + } + } else if (videoTransceiver.direction === 'recvonly') { + if (videoTransceiver.setDirection) { + videoTransceiver.setDirection('inactive'); + } else { + videoTransceiver.direction = 'inactive'; + } + } + } else if (offerOptions.offerToReceiveVideo === true && + !videoTransceiver) { + this.addTransceiver('video'); + } + } + return origCreateOffer.apply(this, arguments); + }; + } + + function shimAudioContext(window) { + if (typeof window !== 'object' || window.AudioContext) { + return; + } + window.AudioContext = window.webkitAudioContext; + } + + var safariShim = /*#__PURE__*/Object.freeze({ + __proto__: null, + shimLocalStreamsAPI: shimLocalStreamsAPI, + shimRemoteStreamsAPI: shimRemoteStreamsAPI, + shimCallbacksAPI: shimCallbacksAPI, + shimGetUserMedia: shimGetUserMedia, + shimConstraints: shimConstraints, + shimRTCIceServerUrls: shimRTCIceServerUrls, + shimTrackEventTransceiver: shimTrackEventTransceiver, + shimCreateOfferLegacy: shimCreateOfferLegacy, + shimAudioContext: shimAudioContext + }); + + /* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + function shimRTCIceCandidate(window) { + // foundation is arbitrarily chosen as an indicator for full support for + // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface + if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in + window.RTCIceCandidate.prototype)) { + return; + } + + const NativeRTCIceCandidate = window.RTCIceCandidate; + window.RTCIceCandidate = function RTCIceCandidate(args) { + // Remove the a= which shouldn't be part of the candidate string. + if (typeof args === 'object' && args.candidate && + args.candidate.indexOf('a=') === 0) { + args = JSON.parse(JSON.stringify(args)); + args.candidate = args.candidate.substr(2); + } + + if (args.candidate && args.candidate.length) { + // Augment the native candidate with the parsed fields. + const nativeCandidate = new NativeRTCIceCandidate(args); + const parsedCandidate = sdp.parseCandidate(args.candidate); + const augmentedCandidate = Object.assign(nativeCandidate, + parsedCandidate); + + // Add a serializer that does not serialize the extra attributes. + augmentedCandidate.toJSON = function toJSON() { + return { + candidate: augmentedCandidate.candidate, + sdpMid: augmentedCandidate.sdpMid, + sdpMLineIndex: augmentedCandidate.sdpMLineIndex, + usernameFragment: augmentedCandidate.usernameFragment, + }; + }; + return augmentedCandidate; + } + return new NativeRTCIceCandidate(args); + }; + window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype; + + // Hook up the augmented candidate in onicecandidate and + // addEventListener('icecandidate', ...) + wrapPeerConnectionEvent(window, 'icecandidate', e => { + if (e.candidate) { + Object.defineProperty(e, 'candidate', { + value: new window.RTCIceCandidate(e.candidate), + writable: 'false' + }); + } + return e; + }); + } + + function shimMaxMessageSize(window, browserDetails) { + if (!window.RTCPeerConnection) { + return; + } + + if (!('sctp' in window.RTCPeerConnection.prototype)) { + Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', { + get() { + return typeof this._sctp === 'undefined' ? null : this._sctp; + } + }); + } + + const sctpInDescription = function(description) { + if (!description || !description.sdp) { + return false; + } + const sections = sdp.splitSections(description.sdp); + sections.shift(); + return sections.some(mediaSection => { + const mLine = sdp.parseMLine(mediaSection); + return mLine && mLine.kind === 'application' + && mLine.protocol.indexOf('SCTP') !== -1; + }); + }; + + const getRemoteFirefoxVersion = function(description) { + // TODO: Is there a better solution for detecting Firefox? + const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\d+)/); + if (match === null || match.length < 2) { + return -1; + } + const version = parseInt(match[1], 10); + // Test for NaN (yes, this is ugly) + return version !== version ? -1 : version; + }; + + const getCanSendMaxMessageSize = function(remoteIsFirefox) { + // Every implementation we know can send at least 64 KiB. + // Note: Although Chrome is technically able to send up to 256 KiB, the + // data does not reach the other peer reliably. + // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419 + let canSendMaxMessageSize = 65536; + if (browserDetails.browser === 'firefox') { + if (browserDetails.version < 57) { + if (remoteIsFirefox === -1) { + // FF < 57 will send in 16 KiB chunks using the deprecated PPID + // fragmentation. + canSendMaxMessageSize = 16384; + } else { + // However, other FF (and RAWRTC) can reassemble PPID-fragmented + // messages. Thus, supporting ~2 GiB when sending. + canSendMaxMessageSize = 2147483637; + } + } else if (browserDetails.version < 60) { + // Currently, all FF >= 57 will reset the remote maximum message size + // to the default value when a data channel is created at a later + // stage. :( + // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831 + canSendMaxMessageSize = + browserDetails.version === 57 ? 65535 : 65536; + } else { + // FF >= 60 supports sending ~2 GiB + canSendMaxMessageSize = 2147483637; + } + } + return canSendMaxMessageSize; + }; + + const getMaxMessageSize = function(description, remoteIsFirefox) { + // Note: 65536 bytes is the default value from the SDP spec. Also, + // every implementation we know supports receiving 65536 bytes. + let maxMessageSize = 65536; + + // FF 57 has a slightly incorrect default remote max message size, so + // we need to adjust it here to avoid a failure when sending. + // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697 + if (browserDetails.browser === 'firefox' + && browserDetails.version === 57) { + maxMessageSize = 65535; + } + + const match = sdp.matchPrefix(description.sdp, + 'a=max-message-size:'); + if (match.length > 0) { + maxMessageSize = parseInt(match[0].substr(19), 10); + } else if (browserDetails.browser === 'firefox' && + remoteIsFirefox !== -1) { + // If the maximum message size is not present in the remote SDP and + // both local and remote are Firefox, the remote peer can receive + // ~2 GiB. + maxMessageSize = 2147483637; + } + return maxMessageSize; + }; + + const origSetRemoteDescription = + window.RTCPeerConnection.prototype.setRemoteDescription; + window.RTCPeerConnection.prototype.setRemoteDescription = + function setRemoteDescription() { + this._sctp = null; + // Chrome decided to not expose .sctp in plan-b mode. + // As usual, adapter.js has to do an 'ugly worakaround' + // to cover up the mess. + if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) { + const {sdpSemantics} = this.getConfiguration(); + if (sdpSemantics === 'plan-b') { + Object.defineProperty(this, 'sctp', { + get() { + return typeof this._sctp === 'undefined' ? null : this._sctp; + }, + enumerable: true, + configurable: true, + }); + } + } + + if (sctpInDescription(arguments[0])) { + // Check if the remote is FF. + const isFirefox = getRemoteFirefoxVersion(arguments[0]); + + // Get the maximum message size the local peer is capable of sending + const canSendMMS = getCanSendMaxMessageSize(isFirefox); + + // Get the maximum message size of the remote peer. + const remoteMMS = getMaxMessageSize(arguments[0], isFirefox); + + // Determine final maximum message size + let maxMessageSize; + if (canSendMMS === 0 && remoteMMS === 0) { + maxMessageSize = Number.POSITIVE_INFINITY; + } else if (canSendMMS === 0 || remoteMMS === 0) { + maxMessageSize = Math.max(canSendMMS, remoteMMS); + } else { + maxMessageSize = Math.min(canSendMMS, remoteMMS); + } + + // Create a dummy RTCSctpTransport object and the 'maxMessageSize' + // attribute. + const sctp = {}; + Object.defineProperty(sctp, 'maxMessageSize', { + get() { + return maxMessageSize; + } + }); + this._sctp = sctp; + } + + return origSetRemoteDescription.apply(this, arguments); + }; + } + + function shimSendThrowTypeError(window) { + if (!(window.RTCPeerConnection && + 'createDataChannel' in window.RTCPeerConnection.prototype)) { + return; + } + + // Note: Although Firefox >= 57 has a native implementation, the maximum + // message size can be reset for all data channels at a later stage. + // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831 + + function wrapDcSend(dc, pc) { + const origDataChannelSend = dc.send; + dc.send = function send() { + const data = arguments[0]; + const length = data.length || data.size || data.byteLength; + if (dc.readyState === 'open' && + pc.sctp && length > pc.sctp.maxMessageSize) { + throw new TypeError('Message too large (can send a maximum of ' + + pc.sctp.maxMessageSize + ' bytes)'); + } + return origDataChannelSend.apply(dc, arguments); + }; + } + const origCreateDataChannel = + window.RTCPeerConnection.prototype.createDataChannel; + window.RTCPeerConnection.prototype.createDataChannel = + function createDataChannel() { + const dataChannel = origCreateDataChannel.apply(this, arguments); + wrapDcSend(dataChannel, this); + return dataChannel; + }; + wrapPeerConnectionEvent(window, 'datachannel', e => { + wrapDcSend(e.channel, e.target); + return e; + }); + } + + + /* shims RTCConnectionState by pretending it is the same as iceConnectionState. + * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12 + * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect + * since DTLS failures would be hidden. See + * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827 + * for the Firefox tracking bug. + */ + function shimConnectionState(window) { + if (!window.RTCPeerConnection || + 'connectionState' in window.RTCPeerConnection.prototype) { + return; + } + const proto = window.RTCPeerConnection.prototype; + Object.defineProperty(proto, 'connectionState', { + get() { + return { + completed: 'connected', + checking: 'connecting' + }[this.iceConnectionState] || this.iceConnectionState; + }, + enumerable: true, + configurable: true + }); + Object.defineProperty(proto, 'onconnectionstatechange', { + get() { + return this._onconnectionstatechange || null; + }, + set(cb) { + if (this._onconnectionstatechange) { + this.removeEventListener('connectionstatechange', + this._onconnectionstatechange); + delete this._onconnectionstatechange; + } + if (cb) { + this.addEventListener('connectionstatechange', + this._onconnectionstatechange = cb); + } + }, + enumerable: true, + configurable: true + }); + + ['setLocalDescription', 'setRemoteDescription'].forEach((method) => { + const origMethod = proto[method]; + proto[method] = function() { + if (!this._connectionstatechangepoly) { + this._connectionstatechangepoly = e => { + const pc = e.target; + if (pc._lastConnectionState !== pc.connectionState) { + pc._lastConnectionState = pc.connectionState; + const newEvent = new Event('connectionstatechange', e); + pc.dispatchEvent(newEvent); + } + return e; + }; + this.addEventListener('iceconnectionstatechange', + this._connectionstatechangepoly); + } + return origMethod.apply(this, arguments); + }; + }); + } + + function removeExtmapAllowMixed(window, browserDetails) { + /* remove a=extmap-allow-mixed for webrtc.org < M71 */ + if (!window.RTCPeerConnection) { + return; + } + if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) { + return; + } + if (browserDetails.browser === 'safari' && browserDetails.version >= 605) { + return; + } + const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription; + window.RTCPeerConnection.prototype.setRemoteDescription = + function setRemoteDescription(desc) { + if (desc && desc.sdp && desc.sdp.indexOf('\na=extmap-allow-mixed') !== -1) { + const sdp = desc.sdp.split('\n').filter((line) => { + return line.trim() !== 'a=extmap-allow-mixed'; + }).join('\n'); + // Safari enforces read-only-ness of RTCSessionDescription fields. + if (window.RTCSessionDescription && + desc instanceof window.RTCSessionDescription) { + arguments[0] = new window.RTCSessionDescription({ + type: desc.type, + sdp, + }); + } else { + desc.sdp = sdp; + } + } + return nativeSRD.apply(this, arguments); + }; + } + + function shimAddIceCandidateNullOrEmpty(window, browserDetails) { + // Support for addIceCandidate(null or undefined) + // as well as addIceCandidate({candidate: "", ...}) + // https://bugs.chromium.org/p/chromium/issues/detail?id=978582 + // Note: must be called before other polyfills which change the signature. + if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) { + return; + } + const nativeAddIceCandidate = + window.RTCPeerConnection.prototype.addIceCandidate; + if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) { + return; + } + window.RTCPeerConnection.prototype.addIceCandidate = + function addIceCandidate() { + if (!arguments[0]) { + if (arguments[1]) { + arguments[1].apply(null); + } + return Promise.resolve(); + } + // Firefox 68+ emits and processes {candidate: "", ...}, ignore + // in older versions. + // Native support for ignoring exists for Chrome M77+. + // Safari ignores as well, exact version unknown but works in the same + // version that also ignores addIceCandidate(null). + if (((browserDetails.browser === 'chrome' && browserDetails.version < 78) + || (browserDetails.browser === 'firefox' + && browserDetails.version < 68) + || (browserDetails.browser === 'safari')) + && arguments[0] && arguments[0].candidate === '') { + return Promise.resolve(); + } + return nativeAddIceCandidate.apply(this, arguments); + }; + } + + var commonShim = /*#__PURE__*/Object.freeze({ + __proto__: null, + shimRTCIceCandidate: shimRTCIceCandidate, + shimMaxMessageSize: shimMaxMessageSize, + shimSendThrowTypeError: shimSendThrowTypeError, + shimConnectionState: shimConnectionState, + removeExtmapAllowMixed: removeExtmapAllowMixed, + shimAddIceCandidateNullOrEmpty: shimAddIceCandidateNullOrEmpty + }); + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + // Shimming starts here. + function adapterFactory({window} = {}, options = { + shimChrome: true, + shimFirefox: true, + shimEdge: true, + shimSafari: true, + }) { + // Utils. + const logging = log$1; + const browserDetails = detectBrowser(window); + + const adapter = { + browserDetails, + commonShim, + extractVersion: extractVersion, + disableLog: disableLog, + disableWarnings: disableWarnings + }; + + // Shim browser if found. + switch (browserDetails.browser) { + case 'chrome': + if (!chromeShim || !shimPeerConnection$2 || + !options.shimChrome) { + logging('Chrome shim is not included in this adapter release.'); + return adapter; + } + if (browserDetails.version === null) { + logging('Chrome shim can not determine version, not shimming.'); + return adapter; + } + logging('adapter.js shimming chrome.'); + // Export to the adapter global object visible in the browser. + adapter.browserShim = chromeShim; + + // Must be called before shimPeerConnection. + shimAddIceCandidateNullOrEmpty(window, browserDetails); + + shimGetUserMedia$3(window, browserDetails); + shimMediaStream(window); + shimPeerConnection$2(window, browserDetails); + shimOnTrack$1(window); + shimAddTrackRemoveTrack(window, browserDetails); + shimGetSendersWithDtmf(window); + shimGetStats(window); + shimSenderReceiverGetStats(window); + fixNegotiationNeeded(window, browserDetails); + + shimRTCIceCandidate(window); + shimConnectionState(window); + shimMaxMessageSize(window, browserDetails); + shimSendThrowTypeError(window); + removeExtmapAllowMixed(window, browserDetails); + break; + case 'firefox': + if (!firefoxShim || !shimPeerConnection || + !options.shimFirefox) { + logging('Firefox shim is not included in this adapter release.'); + return adapter; + } + logging('adapter.js shimming firefox.'); + // Export to the adapter global object visible in the browser. + adapter.browserShim = firefoxShim; + + // Must be called before shimPeerConnection. + shimAddIceCandidateNullOrEmpty(window, browserDetails); + + shimGetUserMedia$1(window, browserDetails); + shimPeerConnection(window, browserDetails); + shimOnTrack(window); + shimRemoveStream(window); + shimSenderGetStats(window); + shimReceiverGetStats(window); + shimRTCDataChannel(window); + shimAddTransceiver(window); + shimGetParameters(window); + shimCreateOffer(window); + shimCreateAnswer(window); + + shimRTCIceCandidate(window); + shimConnectionState(window); + shimMaxMessageSize(window, browserDetails); + shimSendThrowTypeError(window); + break; + case 'edge': + if (!edgeShim || !shimPeerConnection$1 || !options.shimEdge) { + logging('MS edge shim is not included in this adapter release.'); + return adapter; + } + logging('adapter.js shimming edge.'); + // Export to the adapter global object visible in the browser. + adapter.browserShim = edgeShim; + + shimGetUserMedia$2(window); + shimGetDisplayMedia$1(window); + shimPeerConnection$1(window, browserDetails); + shimReplaceTrack(window); + + // the edge shim implements the full RTCIceCandidate object. + + shimMaxMessageSize(window, browserDetails); + shimSendThrowTypeError(window); + break; + case 'safari': + if (!safariShim || !options.shimSafari) { + logging('Safari shim is not included in this adapter release.'); + return adapter; + } + logging('adapter.js shimming safari.'); + // Export to the adapter global object visible in the browser. + adapter.browserShim = safariShim; + + // Must be called before shimCallbackAPI. + shimAddIceCandidateNullOrEmpty(window, browserDetails); + + shimRTCIceServerUrls(window); + shimCreateOfferLegacy(window); + shimCallbacksAPI(window); + shimLocalStreamsAPI(window); + shimRemoteStreamsAPI(window); + shimTrackEventTransceiver(window); + shimGetUserMedia(window); + shimAudioContext(window); + + shimRTCIceCandidate(window); + shimMaxMessageSize(window, browserDetails); + shimSendThrowTypeError(window); + removeExtmapAllowMixed(window, browserDetails); + break; + default: + logging('Unsupported browser!'); + break; + } + + return adapter; + } + + /* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. + */ + + adapterFactory({window: typeof window === 'undefined' ? undefined : window}); + + /** + * @class AudioTrackConstraints + * @classDesc Constraints for creating an audio MediaStreamTrack. + * @memberof Owt.Base + * @constructor + * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track. + */ + + class AudioTrackConstraints { + // eslint-disable-next-line require-jsdoc + constructor(source) { + if (!Object.values(AudioSourceInfo).some(v => v === source)) { + throw new TypeError('Invalid source.'); + } + /** + * @member {string} source + * @memberof Owt.Base.AudioTrackConstraints + * @desc Values could be "mic", "screen-cast", "file" or "mixed". + * @instance + */ + + + this.source = source; + /** + * @member {string} deviceId + * @memberof Owt.Base.AudioTrackConstraints + * @desc Do not provide deviceId if source is not "mic". + * @instance + * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId + */ + + this.deviceId = undefined; + } + + } + /** + * @class VideoTrackConstraints + * @classDesc Constraints for creating a video MediaStreamTrack. + * @memberof Owt.Base + * @constructor + * @param {Owt.Base.VideoSourceInfo} source Source info of this video track. + */ + + class VideoTrackConstraints { + // eslint-disable-next-line require-jsdoc + constructor(source) { + if (!Object.values(VideoSourceInfo).some(v => v === source)) { + throw new TypeError('Invalid source.'); + } + /** + * @member {string} source + * @memberof Owt.Base.VideoTrackConstraints + * @desc Values could be "camera", "screen-cast", "file" or "mixed". + * @instance + */ + + + this.source = source; + /** + * @member {string} deviceId + * @memberof Owt.Base.VideoTrackConstraints + * @desc Do not provide deviceId if source is not "camera". + * @instance + * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId + */ + + this.deviceId = undefined; + /** + * @member {Owt.Base.Resolution} resolution + * @memberof Owt.Base.VideoTrackConstraints + * @instance + */ + + this.resolution = undefined; + /** + * @member {number} frameRate + * @memberof Owt.Base.VideoTrackConstraints + * @instance + */ + + this.frameRate = undefined; + } + + } + /** + * @class StreamConstraints + * @classDesc Constraints for creating a MediaStream from screen mic and camera. + * @memberof Owt.Base + * @constructor + * @param {?Owt.Base.AudioTrackConstraints} audioConstraints + * @param {?Owt.Base.VideoTrackConstraints} videoConstraints + */ + + class StreamConstraints { + // eslint-disable-next-line require-jsdoc + constructor(audioConstraints = false, videoConstraints = false) { + /** + * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio + * @memberof Owt.Base.MediaStreamDeviceConstraints + * @instance + */ + this.audio = audioConstraints; + /** + * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video + * @memberof Owt.Base.MediaStreamDeviceConstraints + * @instance + */ + + this.video = videoConstraints; + } + + } // eslint-disable-next-line require-jsdoc + + function isVideoConstrainsForScreenCast(constraints) { + return typeof constraints.video === 'object' && constraints.video.source === VideoSourceInfo.SCREENCAST; + } + /** + * @class MediaStreamFactory + * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself. + * @memberof Owt.Base + */ + + + class MediaStreamFactory { + /** + * @function createMediaStream + * @static + * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are "screen-cast". + * @memberof Owt.Base.MediaStreamFactory + * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened: + * - One or more parameters cannot be satisfied. + * - Specified device is busy. + * - Cannot obtain necessary permission or operation is canceled by user. + * - Video source is screen cast, while audio source is not. + * - Audio source is screen cast, while video source is disabled. + * @param {Owt.Base.StreamConstraints} constraints + */ + static createMediaStream(constraints) { + if (typeof constraints !== 'object' || !constraints.audio && !constraints.video) { + return Promise.reject(new TypeError('Invalid constrains')); + } + + if (!isVideoConstrainsForScreenCast(constraints) && typeof constraints.audio === 'object' && constraints.audio.source === AudioSourceInfo.SCREENCAST) { + return Promise.reject(new TypeError('Cannot share screen without video.')); + } + + if (isVideoConstrainsForScreenCast(constraints) && !isChrome() && !isFirefox()) { + return Promise.reject(new TypeError('Screen sharing only supports Chrome and Firefox.')); + } + + if (isVideoConstrainsForScreenCast(constraints) && typeof constraints.audio === 'object' && constraints.audio.source !== AudioSourceInfo.SCREENCAST) { + return Promise.reject(new TypeError('Cannot capture video from screen cast while capture audio from' + ' other source.')); + } // Check and convert constraints. + + + if (!constraints.audio && !constraints.video) { + return Promise.reject(new TypeError('At least one of audio and video must be requested.')); + } + + const mediaConstraints = Object.create({}); + + if (typeof constraints.audio === 'object' && constraints.audio.source === AudioSourceInfo.MIC) { + mediaConstraints.audio = Object.create({}); + + if (isEdge()) { + mediaConstraints.audio.deviceId = constraints.audio.deviceId; + } else { + mediaConstraints.audio.deviceId = { + exact: constraints.audio.deviceId + }; + } + } else { + if (constraints.audio.source === AudioSourceInfo.SCREENCAST) { + mediaConstraints.audio = true; + } else { + mediaConstraints.audio = constraints.audio; + } + } + + if (typeof constraints.video === 'object') { + mediaConstraints.video = Object.create({}); + + if (typeof constraints.video.frameRate === 'number') { + mediaConstraints.video.frameRate = constraints.video.frameRate; + } + + if (constraints.video.resolution && constraints.video.resolution.width && constraints.video.resolution.height) { + if (constraints.video.source === VideoSourceInfo.SCREENCAST) { + mediaConstraints.video.width = constraints.video.resolution.width; + mediaConstraints.video.height = constraints.video.resolution.height; + } else { + mediaConstraints.video.width = Object.create({}); + mediaConstraints.video.width.exact = constraints.video.resolution.width; + mediaConstraints.video.height = Object.create({}); + mediaConstraints.video.height.exact = constraints.video.resolution.height; + } + } + + if (typeof constraints.video.deviceId === 'string') { + mediaConstraints.video.deviceId = { + exact: constraints.video.deviceId + }; + } + + if (isFirefox() && constraints.video.source === VideoSourceInfo.SCREENCAST) { + mediaConstraints.video.mediaSource = 'screen'; + } + } else { + mediaConstraints.video = constraints.video; + } + + if (isVideoConstrainsForScreenCast(constraints)) { + return navigator.mediaDevices.getDisplayMedia(mediaConstraints); + } else { + return navigator.mediaDevices.getUserMedia(mediaConstraints); + } + } + + } + + // Copyright (C) <2018> Intel Corporation + + var media = /*#__PURE__*/Object.freeze({ + __proto__: null, + AudioTrackConstraints: AudioTrackConstraints, + VideoTrackConstraints: VideoTrackConstraints, + StreamConstraints: StreamConstraints, + MediaStreamFactory: MediaStreamFactory, + AudioSourceInfo: AudioSourceInfo, + VideoSourceInfo: VideoSourceInfo, + TrackKind: TrackKind, + Resolution: Resolution + }); + + let logger; + let errorLogger; + function setLogger() { + /*eslint-disable */ + logger = console.log; + errorLogger = console.error; + /*eslint-enable */ + } + function log(message, ...optionalParams) { + if (logger) { + logger(message, ...optionalParams); + } + } + function error(message, ...optionalParams) { + if (errorLogger) { + errorLogger(message, ...optionalParams); + } + } + + class Event$1 { + constructor(type) { + this.listener = {}; + this.type = type | ''; + } + + on(event, fn) { + if (!this.listener[event]) { + this.listener[event] = []; + } + + this.listener[event].push(fn); + return true; + } + + off(event, fn) { + if (this.listener[event]) { + var index = this.listener[event].indexOf(fn); + + if (index > -1) { + this.listener[event].splice(index, 1); + } + + return true; + } + + return false; + } + + offAll() { + this.listener = {}; + } + + dispatch(event, data) { + if (this.listener[event]) { + this.listener[event].map(each => { + each.apply(null, [data]); + }); + return true; + } + + return false; + } + + } + + var bind = function bind(fn, thisArg) { + return function wrap() { + var args = new Array(arguments.length); + for (var i = 0; i < args.length; i++) { + args[i] = arguments[i]; + } + return fn.apply(thisArg, args); + }; + }; + + /*global toString:true*/ + + // utils is a library of generic helper functions non-specific to axios + + var toString = Object.prototype.toString; + + /** + * Determine if a value is an Array + * + * @param {Object} val The value to test + * @returns {boolean} True if value is an Array, otherwise false + */ + function isArray(val) { + return toString.call(val) === '[object Array]'; + } + + /** + * Determine if a value is undefined + * + * @param {Object} val The value to test + * @returns {boolean} True if the value is undefined, otherwise false + */ + function isUndefined(val) { + return typeof val === 'undefined'; + } + + /** + * Determine if a value is a Buffer + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Buffer, otherwise false + */ + function isBuffer(val) { + return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor) + && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val); + } + + /** + * Determine if a value is an ArrayBuffer + * + * @param {Object} val The value to test + * @returns {boolean} True if value is an ArrayBuffer, otherwise false + */ + function isArrayBuffer(val) { + return toString.call(val) === '[object ArrayBuffer]'; + } + + /** + * Determine if a value is a FormData + * + * @param {Object} val The value to test + * @returns {boolean} True if value is an FormData, otherwise false + */ + function isFormData(val) { + return (typeof FormData !== 'undefined') && (val instanceof FormData); + } + + /** + * Determine if a value is a view on an ArrayBuffer + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false + */ + function isArrayBufferView(val) { + var result; + if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) { + result = ArrayBuffer.isView(val); + } else { + result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer); + } + return result; + } + + /** + * Determine if a value is a String + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a String, otherwise false + */ + function isString(val) { + return typeof val === 'string'; + } + + /** + * Determine if a value is a Number + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Number, otherwise false + */ + function isNumber(val) { + return typeof val === 'number'; + } + + /** + * Determine if a value is an Object + * + * @param {Object} val The value to test + * @returns {boolean} True if value is an Object, otherwise false + */ + function isObject(val) { + return val !== null && typeof val === 'object'; + } + + /** + * Determine if a value is a plain Object + * + * @param {Object} val The value to test + * @return {boolean} True if value is a plain Object, otherwise false + */ + function isPlainObject(val) { + if (toString.call(val) !== '[object Object]') { + return false; + } + + var prototype = Object.getPrototypeOf(val); + return prototype === null || prototype === Object.prototype; + } + + /** + * Determine if a value is a Date + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Date, otherwise false + */ + function isDate(val) { + return toString.call(val) === '[object Date]'; + } + + /** + * Determine if a value is a File + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a File, otherwise false + */ + function isFile(val) { + return toString.call(val) === '[object File]'; + } + + /** + * Determine if a value is a Blob + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Blob, otherwise false + */ + function isBlob(val) { + return toString.call(val) === '[object Blob]'; + } + + /** + * Determine if a value is a Function + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Function, otherwise false + */ + function isFunction(val) { + return toString.call(val) === '[object Function]'; + } + + /** + * Determine if a value is a Stream + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a Stream, otherwise false + */ + function isStream(val) { + return isObject(val) && isFunction(val.pipe); + } + + /** + * Determine if a value is a URLSearchParams object + * + * @param {Object} val The value to test + * @returns {boolean} True if value is a URLSearchParams object, otherwise false + */ + function isURLSearchParams(val) { + return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams; + } + + /** + * Trim excess whitespace off the beginning and end of a string + * + * @param {String} str The String to trim + * @returns {String} The String freed of excess whitespace + */ + function trim(str) { + return str.replace(/^\s*/, '').replace(/\s*$/, ''); + } + + /** + * Determine if we're running in a standard browser environment + * + * This allows axios to run in a web worker, and react-native. + * Both environments support XMLHttpRequest, but not fully standard globals. + * + * web workers: + * typeof window -> undefined + * typeof document -> undefined + * + * react-native: + * navigator.product -> 'ReactNative' + * nativescript + * navigator.product -> 'NativeScript' or 'NS' + */ + function isStandardBrowserEnv() { + if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' || + navigator.product === 'NativeScript' || + navigator.product === 'NS')) { + return false; + } + return ( + typeof window !== 'undefined' && + typeof document !== 'undefined' + ); + } + + /** + * Iterate over an Array or an Object invoking a function for each item. + * + * If `obj` is an Array callback will be called passing + * the value, index, and complete array for each item. + * + * If 'obj' is an Object callback will be called passing + * the value, key, and complete object for each property. + * + * @param {Object|Array} obj The object to iterate + * @param {Function} fn The callback to invoke for each item + */ + function forEach(obj, fn) { + // Don't bother if no value provided + if (obj === null || typeof obj === 'undefined') { + return; + } + + // Force an array if not already something iterable + if (typeof obj !== 'object') { + /*eslint no-param-reassign:0*/ + obj = [obj]; + } + + if (isArray(obj)) { + // Iterate over array values + for (var i = 0, l = obj.length; i < l; i++) { + fn.call(null, obj[i], i, obj); + } + } else { + // Iterate over object keys + for (var key in obj) { + if (Object.prototype.hasOwnProperty.call(obj, key)) { + fn.call(null, obj[key], key, obj); + } + } + } + } + + /** + * Accepts varargs expecting each argument to be an object, then + * immutably merges the properties of each object and returns result. + * + * When multiple objects contain the same key the later object in + * the arguments list will take precedence. + * + * Example: + * + * ```js + * var result = merge({foo: 123}, {foo: 456}); + * console.log(result.foo); // outputs 456 + * ``` + * + * @param {Object} obj1 Object to merge + * @returns {Object} Result of all merge properties + */ + function merge(/* obj1, obj2, obj3, ... */) { + var result = {}; + function assignValue(val, key) { + if (isPlainObject(result[key]) && isPlainObject(val)) { + result[key] = merge(result[key], val); + } else if (isPlainObject(val)) { + result[key] = merge({}, val); + } else if (isArray(val)) { + result[key] = val.slice(); + } else { + result[key] = val; + } + } + + for (var i = 0, l = arguments.length; i < l; i++) { + forEach(arguments[i], assignValue); + } + return result; + } + + /** + * Extends object a by mutably adding to it the properties of object b. + * + * @param {Object} a The object to be extended + * @param {Object} b The object to copy properties from + * @param {Object} thisArg The object to bind function to + * @return {Object} The resulting value of object a + */ + function extend(a, b, thisArg) { + forEach(b, function assignValue(val, key) { + if (thisArg && typeof val === 'function') { + a[key] = bind(val, thisArg); + } else { + a[key] = val; + } + }); + return a; + } + + /** + * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM) + * + * @param {string} content with BOM + * @return {string} content value without BOM + */ + function stripBOM(content) { + if (content.charCodeAt(0) === 0xFEFF) { + content = content.slice(1); + } + return content; + } + + var utils = { + isArray: isArray, + isArrayBuffer: isArrayBuffer, + isBuffer: isBuffer, + isFormData: isFormData, + isArrayBufferView: isArrayBufferView, + isString: isString, + isNumber: isNumber, + isObject: isObject, + isPlainObject: isPlainObject, + isUndefined: isUndefined, + isDate: isDate, + isFile: isFile, + isBlob: isBlob, + isFunction: isFunction, + isStream: isStream, + isURLSearchParams: isURLSearchParams, + isStandardBrowserEnv: isStandardBrowserEnv, + forEach: forEach, + merge: merge, + extend: extend, + trim: trim, + stripBOM: stripBOM + }; + + function encode(val) { + return encodeURIComponent(val). + replace(/%3A/gi, ':'). + replace(/%24/g, '$'). + replace(/%2C/gi, ','). + replace(/%20/g, '+'). + replace(/%5B/gi, '['). + replace(/%5D/gi, ']'); + } + + /** + * Build a URL by appending params to the end + * + * @param {string} url The base of the url (e.g., http://www.google.com) + * @param {object} [params] The params to be appended + * @returns {string} The formatted url + */ + var buildURL = function buildURL(url, params, paramsSerializer) { + /*eslint no-param-reassign:0*/ + if (!params) { + return url; + } + + var serializedParams; + if (paramsSerializer) { + serializedParams = paramsSerializer(params); + } else if (utils.isURLSearchParams(params)) { + serializedParams = params.toString(); + } else { + var parts = []; + + utils.forEach(params, function serialize(val, key) { + if (val === null || typeof val === 'undefined') { + return; + } + + if (utils.isArray(val)) { + key = key + '[]'; + } else { + val = [val]; + } + + utils.forEach(val, function parseValue(v) { + if (utils.isDate(v)) { + v = v.toISOString(); + } else if (utils.isObject(v)) { + v = JSON.stringify(v); + } + parts.push(encode(key) + '=' + encode(v)); + }); + }); + + serializedParams = parts.join('&'); + } + + if (serializedParams) { + var hashmarkIndex = url.indexOf('#'); + if (hashmarkIndex !== -1) { + url = url.slice(0, hashmarkIndex); + } + + url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams; + } + + return url; + }; + + function InterceptorManager() { + this.handlers = []; + } + + /** + * Add a new interceptor to the stack + * + * @param {Function} fulfilled The function to handle `then` for a `Promise` + * @param {Function} rejected The function to handle `reject` for a `Promise` + * + * @return {Number} An ID used to remove interceptor later + */ + InterceptorManager.prototype.use = function use(fulfilled, rejected) { + this.handlers.push({ + fulfilled: fulfilled, + rejected: rejected + }); + return this.handlers.length - 1; + }; + + /** + * Remove an interceptor from the stack + * + * @param {Number} id The ID that was returned by `use` + */ + InterceptorManager.prototype.eject = function eject(id) { + if (this.handlers[id]) { + this.handlers[id] = null; + } + }; + + /** + * Iterate over all the registered interceptors + * + * This method is particularly useful for skipping over any + * interceptors that may have become `null` calling `eject`. + * + * @param {Function} fn The function to call for each interceptor + */ + InterceptorManager.prototype.forEach = function forEach(fn) { + utils.forEach(this.handlers, function forEachHandler(h) { + if (h !== null) { + fn(h); + } + }); + }; + + var InterceptorManager_1 = InterceptorManager; + + /** + * Transform the data for a request or a response + * + * @param {Object|String} data The data to be transformed + * @param {Array} headers The headers for the request or response + * @param {Array|Function} fns A single function or Array of functions + * @returns {*} The resulting transformed data + */ + var transformData = function transformData(data, headers, fns) { + /*eslint no-param-reassign:0*/ + utils.forEach(fns, function transform(fn) { + data = fn(data, headers); + }); + + return data; + }; + + var isCancel = function isCancel(value) { + return !!(value && value.__CANCEL__); + }; + + var normalizeHeaderName = function normalizeHeaderName(headers, normalizedName) { + utils.forEach(headers, function processHeader(value, name) { + if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) { + headers[normalizedName] = value; + delete headers[name]; + } + }); + }; + + /** + * Update an Error with the specified config, error code, and response. + * + * @param {Error} error The error to update. + * @param {Object} config The config. + * @param {string} [code] The error code (for example, 'ECONNABORTED'). + * @param {Object} [request] The request. + * @param {Object} [response] The response. + * @returns {Error} The error. + */ + var enhanceError = function enhanceError(error, config, code, request, response) { + error.config = config; + if (code) { + error.code = code; + } + + error.request = request; + error.response = response; + error.isAxiosError = true; + + error.toJSON = function toJSON() { + return { + // Standard + message: this.message, + name: this.name, + // Microsoft + description: this.description, + number: this.number, + // Mozilla + fileName: this.fileName, + lineNumber: this.lineNumber, + columnNumber: this.columnNumber, + stack: this.stack, + // Axios + config: this.config, + code: this.code + }; + }; + return error; + }; + + /** + * Create an Error with the specified message, config, error code, request and response. + * + * @param {string} message The error message. + * @param {Object} config The config. + * @param {string} [code] The error code (for example, 'ECONNABORTED'). + * @param {Object} [request] The request. + * @param {Object} [response] The response. + * @returns {Error} The created error. + */ + var createError = function createError(message, config, code, request, response) { + var error = new Error(message); + return enhanceError(error, config, code, request, response); + }; + + /** + * Resolve or reject a Promise based on response status. + * + * @param {Function} resolve A function that resolves the promise. + * @param {Function} reject A function that rejects the promise. + * @param {object} response The response. + */ + var settle = function settle(resolve, reject, response) { + var validateStatus = response.config.validateStatus; + if (!response.status || !validateStatus || validateStatus(response.status)) { + resolve(response); + } else { + reject(createError( + 'Request failed with status code ' + response.status, + response.config, + null, + response.request, + response + )); + } + }; + + var cookies = ( + utils.isStandardBrowserEnv() ? + + // Standard browser envs support document.cookie + (function standardBrowserEnv() { + return { + write: function write(name, value, expires, path, domain, secure) { + var cookie = []; + cookie.push(name + '=' + encodeURIComponent(value)); + + if (utils.isNumber(expires)) { + cookie.push('expires=' + new Date(expires).toGMTString()); + } + + if (utils.isString(path)) { + cookie.push('path=' + path); + } + + if (utils.isString(domain)) { + cookie.push('domain=' + domain); + } + + if (secure === true) { + cookie.push('secure'); + } + + document.cookie = cookie.join('; '); + }, + + read: function read(name) { + var match = document.cookie.match(new RegExp('(^|;\\s*)(' + name + ')=([^;]*)')); + return (match ? decodeURIComponent(match[3]) : null); + }, + + remove: function remove(name) { + this.write(name, '', Date.now() - 86400000); + } + }; + })() : + + // Non standard browser env (web workers, react-native) lack needed support. + (function nonStandardBrowserEnv() { + return { + write: function write() {}, + read: function read() { return null; }, + remove: function remove() {} + }; + })() + ); + + /** + * Determines whether the specified URL is absolute + * + * @param {string} url The URL to test + * @returns {boolean} True if the specified URL is absolute, otherwise false + */ + var isAbsoluteURL = function isAbsoluteURL(url) { + // A URL is considered absolute if it begins with "://" or "//" (protocol-relative URL). + // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed + // by any combination of letters, digits, plus, period, or hyphen. + return /^([a-z][a-z\d\+\-\.]*:)?\/\//i.test(url); + }; + + /** + * Creates a new URL by combining the specified URLs + * + * @param {string} baseURL The base URL + * @param {string} relativeURL The relative URL + * @returns {string} The combined URL + */ + var combineURLs = function combineURLs(baseURL, relativeURL) { + return relativeURL + ? baseURL.replace(/\/+$/, '') + '/' + relativeURL.replace(/^\/+/, '') + : baseURL; + }; + + /** + * Creates a new URL by combining the baseURL with the requestedURL, + * only when the requestedURL is not already an absolute URL. + * If the requestURL is absolute, this function returns the requestedURL untouched. + * + * @param {string} baseURL The base URL + * @param {string} requestedURL Absolute or relative URL to combine + * @returns {string} The combined full path + */ + var buildFullPath = function buildFullPath(baseURL, requestedURL) { + if (baseURL && !isAbsoluteURL(requestedURL)) { + return combineURLs(baseURL, requestedURL); + } + return requestedURL; + }; + + // Headers whose duplicates are ignored by node + // c.f. https://nodejs.org/api/http.html#http_message_headers + var ignoreDuplicateOf = [ + 'age', 'authorization', 'content-length', 'content-type', 'etag', + 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since', + 'last-modified', 'location', 'max-forwards', 'proxy-authorization', + 'referer', 'retry-after', 'user-agent' + ]; + + /** + * Parse headers into an object + * + * ``` + * Date: Wed, 27 Aug 2014 08:58:49 GMT + * Content-Type: application/json + * Connection: keep-alive + * Transfer-Encoding: chunked + * ``` + * + * @param {String} headers Headers needing to be parsed + * @returns {Object} Headers parsed into an object + */ + var parseHeaders = function parseHeaders(headers) { + var parsed = {}; + var key; + var val; + var i; + + if (!headers) { return parsed; } + + utils.forEach(headers.split('\n'), function parser(line) { + i = line.indexOf(':'); + key = utils.trim(line.substr(0, i)).toLowerCase(); + val = utils.trim(line.substr(i + 1)); + + if (key) { + if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) { + return; + } + if (key === 'set-cookie') { + parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]); + } else { + parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val; + } + } + }); + + return parsed; + }; + + var isURLSameOrigin = ( + utils.isStandardBrowserEnv() ? + + // Standard browser envs have full support of the APIs needed to test + // whether the request URL is of the same origin as current location. + (function standardBrowserEnv() { + var msie = /(msie|trident)/i.test(navigator.userAgent); + var urlParsingNode = document.createElement('a'); + var originURL; + + /** + * Parse a URL to discover it's components + * + * @param {String} url The URL to be parsed + * @returns {Object} + */ + function resolveURL(url) { + var href = url; + + if (msie) { + // IE needs attribute set twice to normalize properties + urlParsingNode.setAttribute('href', href); + href = urlParsingNode.href; + } + + urlParsingNode.setAttribute('href', href); + + // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils + return { + href: urlParsingNode.href, + protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '', + host: urlParsingNode.host, + search: urlParsingNode.search ? urlParsingNode.search.replace(/^\?/, '') : '', + hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '', + hostname: urlParsingNode.hostname, + port: urlParsingNode.port, + pathname: (urlParsingNode.pathname.charAt(0) === '/') ? + urlParsingNode.pathname : + '/' + urlParsingNode.pathname + }; + } + + originURL = resolveURL(window.location.href); + + /** + * Determine if a URL shares the same origin as the current location + * + * @param {String} requestURL The URL to test + * @returns {boolean} True if URL shares the same origin, otherwise false + */ + return function isURLSameOrigin(requestURL) { + var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL; + return (parsed.protocol === originURL.protocol && + parsed.host === originURL.host); + }; + })() : + + // Non standard browser envs (web workers, react-native) lack needed support. + (function nonStandardBrowserEnv() { + return function isURLSameOrigin() { + return true; + }; + })() + ); + + var xhr = function xhrAdapter(config) { + return new Promise(function dispatchXhrRequest(resolve, reject) { + var requestData = config.data; + var requestHeaders = config.headers; + + if (utils.isFormData(requestData)) { + delete requestHeaders['Content-Type']; // Let the browser set it + } + + var request = new XMLHttpRequest(); + + // HTTP basic authentication + if (config.auth) { + var username = config.auth.username || ''; + var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : ''; + requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password); + } + + var fullPath = buildFullPath(config.baseURL, config.url); + request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true); + + // Set the request timeout in MS + request.timeout = config.timeout; + + // Listen for ready state + request.onreadystatechange = function handleLoad() { + if (!request || request.readyState !== 4) { + return; + } + + // The request errored out and we didn't get a response, this will be + // handled by onerror instead + // With one exception: request that using file: protocol, most browsers + // will return status as 0 even though it's a successful request + if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) { + return; + } + + // Prepare the response + var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null; + var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response; + var response = { + data: responseData, + status: request.status, + statusText: request.statusText, + headers: responseHeaders, + config: config, + request: request + }; + + settle(resolve, reject, response); + + // Clean up request + request = null; + }; + + // Handle browser request cancellation (as opposed to a manual cancellation) + request.onabort = function handleAbort() { + if (!request) { + return; + } + + reject(createError('Request aborted', config, 'ECONNABORTED', request)); + + // Clean up request + request = null; + }; + + // Handle low level network errors + request.onerror = function handleError() { + // Real errors are hidden from us by the browser + // onerror should only fire if it's a network error + reject(createError('Network Error', config, null, request)); + + // Clean up request + request = null; + }; + + // Handle timeout + request.ontimeout = function handleTimeout() { + var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded'; + if (config.timeoutErrorMessage) { + timeoutErrorMessage = config.timeoutErrorMessage; + } + reject(createError(timeoutErrorMessage, config, 'ECONNABORTED', + request)); + + // Clean up request + request = null; + }; + + // Add xsrf header + // This is only done if running in a standard browser environment. + // Specifically not if we're in a web worker, or react-native. + if (utils.isStandardBrowserEnv()) { + // Add xsrf header + var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ? + cookies.read(config.xsrfCookieName) : + undefined; + + if (xsrfValue) { + requestHeaders[config.xsrfHeaderName] = xsrfValue; + } + } + + // Add headers to the request + if ('setRequestHeader' in request) { + utils.forEach(requestHeaders, function setRequestHeader(val, key) { + if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') { + // Remove Content-Type if data is undefined + delete requestHeaders[key]; + } else { + // Otherwise add header to the request + request.setRequestHeader(key, val); + } + }); + } + + // Add withCredentials to request if needed + if (!utils.isUndefined(config.withCredentials)) { + request.withCredentials = !!config.withCredentials; + } + + // Add responseType to request if needed + if (config.responseType) { + try { + request.responseType = config.responseType; + } catch (e) { + // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2. + // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function. + if (config.responseType !== 'json') { + throw e; + } + } + } + + // Handle progress if needed + if (typeof config.onDownloadProgress === 'function') { + request.addEventListener('progress', config.onDownloadProgress); + } + + // Not all browsers support upload events + if (typeof config.onUploadProgress === 'function' && request.upload) { + request.upload.addEventListener('progress', config.onUploadProgress); + } + + if (config.cancelToken) { + // Handle cancellation + config.cancelToken.promise.then(function onCanceled(cancel) { + if (!request) { + return; + } + + request.abort(); + reject(cancel); + // Clean up request + request = null; + }); + } + + if (!requestData) { + requestData = null; + } + + // Send the request + request.send(requestData); + }); + }; + + var DEFAULT_CONTENT_TYPE = { + 'Content-Type': 'application/x-www-form-urlencoded' + }; + + function setContentTypeIfUnset(headers, value) { + if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) { + headers['Content-Type'] = value; + } + } + + function getDefaultAdapter() { + var adapter; + if (typeof XMLHttpRequest !== 'undefined') { + // For browsers use XHR adapter + adapter = xhr; + } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') { + // For node use HTTP adapter + adapter = xhr; + } + return adapter; + } + + var defaults = { + adapter: getDefaultAdapter(), + + transformRequest: [function transformRequest(data, headers) { + normalizeHeaderName(headers, 'Accept'); + normalizeHeaderName(headers, 'Content-Type'); + if (utils.isFormData(data) || + utils.isArrayBuffer(data) || + utils.isBuffer(data) || + utils.isStream(data) || + utils.isFile(data) || + utils.isBlob(data) + ) { + return data; + } + if (utils.isArrayBufferView(data)) { + return data.buffer; + } + if (utils.isURLSearchParams(data)) { + setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8'); + return data.toString(); + } + if (utils.isObject(data)) { + setContentTypeIfUnset(headers, 'application/json;charset=utf-8'); + return JSON.stringify(data); + } + return data; + }], + + transformResponse: [function transformResponse(data) { + /*eslint no-param-reassign:0*/ + if (typeof data === 'string') { + try { + data = JSON.parse(data); + } catch (e) { /* Ignore */ } + } + return data; + }], + + /** + * A timeout in milliseconds to abort a request. If set to 0 (default) a + * timeout is not created. + */ + timeout: 0, + + xsrfCookieName: 'XSRF-TOKEN', + xsrfHeaderName: 'X-XSRF-TOKEN', + + maxContentLength: -1, + maxBodyLength: -1, + + validateStatus: function validateStatus(status) { + return status >= 200 && status < 300; + } + }; + + defaults.headers = { + common: { + 'Accept': 'application/json, text/plain, */*' + } + }; + + utils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) { + defaults.headers[method] = {}; + }); + + utils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) { + defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE); + }); + + var defaults_1 = defaults; + + /** + * Throws a `Cancel` if cancellation has been requested. + */ + function throwIfCancellationRequested(config) { + if (config.cancelToken) { + config.cancelToken.throwIfRequested(); + } + } + + /** + * Dispatch a request to the server using the configured adapter. + * + * @param {object} config The config that is to be used for the request + * @returns {Promise} The Promise to be fulfilled + */ + var dispatchRequest = function dispatchRequest(config) { + throwIfCancellationRequested(config); + + // Ensure headers exist + config.headers = config.headers || {}; + + // Transform request data + config.data = transformData( + config.data, + config.headers, + config.transformRequest + ); + + // Flatten headers + config.headers = utils.merge( + config.headers.common || {}, + config.headers[config.method] || {}, + config.headers + ); + + utils.forEach( + ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'], + function cleanHeaderConfig(method) { + delete config.headers[method]; + } + ); + + var adapter = config.adapter || defaults_1.adapter; + + return adapter(config).then(function onAdapterResolution(response) { + throwIfCancellationRequested(config); + + // Transform response data + response.data = transformData( + response.data, + response.headers, + config.transformResponse + ); + + return response; + }, function onAdapterRejection(reason) { + if (!isCancel(reason)) { + throwIfCancellationRequested(config); + + // Transform response data + if (reason && reason.response) { + reason.response.data = transformData( + reason.response.data, + reason.response.headers, + config.transformResponse + ); + } + } + + return Promise.reject(reason); + }); + }; + + /** + * Config-specific merge-function which creates a new config-object + * by merging two configuration objects together. + * + * @param {Object} config1 + * @param {Object} config2 + * @returns {Object} New object resulting from merging config2 to config1 + */ + var mergeConfig = function mergeConfig(config1, config2) { + // eslint-disable-next-line no-param-reassign + config2 = config2 || {}; + var config = {}; + + var valueFromConfig2Keys = ['url', 'method', 'data']; + var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params']; + var defaultToConfig2Keys = [ + 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer', + 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName', + 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress', + 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent', + 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding' + ]; + var directMergeKeys = ['validateStatus']; + + function getMergedValue(target, source) { + if (utils.isPlainObject(target) && utils.isPlainObject(source)) { + return utils.merge(target, source); + } else if (utils.isPlainObject(source)) { + return utils.merge({}, source); + } else if (utils.isArray(source)) { + return source.slice(); + } + return source; + } + + function mergeDeepProperties(prop) { + if (!utils.isUndefined(config2[prop])) { + config[prop] = getMergedValue(config1[prop], config2[prop]); + } else if (!utils.isUndefined(config1[prop])) { + config[prop] = getMergedValue(undefined, config1[prop]); + } + } + + utils.forEach(valueFromConfig2Keys, function valueFromConfig2(prop) { + if (!utils.isUndefined(config2[prop])) { + config[prop] = getMergedValue(undefined, config2[prop]); + } + }); + + utils.forEach(mergeDeepPropertiesKeys, mergeDeepProperties); + + utils.forEach(defaultToConfig2Keys, function defaultToConfig2(prop) { + if (!utils.isUndefined(config2[prop])) { + config[prop] = getMergedValue(undefined, config2[prop]); + } else if (!utils.isUndefined(config1[prop])) { + config[prop] = getMergedValue(undefined, config1[prop]); + } + }); + + utils.forEach(directMergeKeys, function merge(prop) { + if (prop in config2) { + config[prop] = getMergedValue(config1[prop], config2[prop]); + } else if (prop in config1) { + config[prop] = getMergedValue(undefined, config1[prop]); + } + }); + + var axiosKeys = valueFromConfig2Keys + .concat(mergeDeepPropertiesKeys) + .concat(defaultToConfig2Keys) + .concat(directMergeKeys); + + var otherKeys = Object + .keys(config1) + .concat(Object.keys(config2)) + .filter(function filterAxiosKeys(key) { + return axiosKeys.indexOf(key) === -1; + }); + + utils.forEach(otherKeys, mergeDeepProperties); + + return config; + }; + + /** + * Create a new instance of Axios + * + * @param {Object} instanceConfig The default config for the instance + */ + function Axios(instanceConfig) { + this.defaults = instanceConfig; + this.interceptors = { + request: new InterceptorManager_1(), + response: new InterceptorManager_1() + }; + } + + /** + * Dispatch a request + * + * @param {Object} config The config specific for this request (merged with this.defaults) + */ + Axios.prototype.request = function request(config) { + /*eslint no-param-reassign:0*/ + // Allow for axios('example/url'[, config]) a la fetch API + if (typeof config === 'string') { + config = arguments[1] || {}; + config.url = arguments[0]; + } else { + config = config || {}; + } + + config = mergeConfig(this.defaults, config); + + // Set config.method + if (config.method) { + config.method = config.method.toLowerCase(); + } else if (this.defaults.method) { + config.method = this.defaults.method.toLowerCase(); + } else { + config.method = 'get'; + } + + // Hook up interceptors middleware + var chain = [dispatchRequest, undefined]; + var promise = Promise.resolve(config); + + this.interceptors.request.forEach(function unshiftRequestInterceptors(interceptor) { + chain.unshift(interceptor.fulfilled, interceptor.rejected); + }); + + this.interceptors.response.forEach(function pushResponseInterceptors(interceptor) { + chain.push(interceptor.fulfilled, interceptor.rejected); + }); + + while (chain.length) { + promise = promise.then(chain.shift(), chain.shift()); + } + + return promise; + }; + + Axios.prototype.getUri = function getUri(config) { + config = mergeConfig(this.defaults, config); + return buildURL(config.url, config.params, config.paramsSerializer).replace(/^\?/, ''); + }; + + // Provide aliases for supported request methods + utils.forEach(['delete', 'get', 'head', 'options'], function forEachMethodNoData(method) { + /*eslint func-names:0*/ + Axios.prototype[method] = function(url, config) { + return this.request(mergeConfig(config || {}, { + method: method, + url: url, + data: (config || {}).data + })); + }; + }); + + utils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) { + /*eslint func-names:0*/ + Axios.prototype[method] = function(url, data, config) { + return this.request(mergeConfig(config || {}, { + method: method, + url: url, + data: data + })); + }; + }); + + var Axios_1 = Axios; + + /** + * A `Cancel` is an object that is thrown when an operation is canceled. + * + * @class + * @param {string=} message The message. + */ + function Cancel(message) { + this.message = message; + } + + Cancel.prototype.toString = function toString() { + return 'Cancel' + (this.message ? ': ' + this.message : ''); + }; + + Cancel.prototype.__CANCEL__ = true; + + var Cancel_1 = Cancel; + + /** + * A `CancelToken` is an object that can be used to request cancellation of an operation. + * + * @class + * @param {Function} executor The executor function. + */ + function CancelToken(executor) { + if (typeof executor !== 'function') { + throw new TypeError('executor must be a function.'); + } + + var resolvePromise; + this.promise = new Promise(function promiseExecutor(resolve) { + resolvePromise = resolve; + }); + + var token = this; + executor(function cancel(message) { + if (token.reason) { + // Cancellation has already been requested + return; + } + + token.reason = new Cancel_1(message); + resolvePromise(token.reason); + }); + } + + /** + * Throws a `Cancel` if cancellation has been requested. + */ + CancelToken.prototype.throwIfRequested = function throwIfRequested() { + if (this.reason) { + throw this.reason; + } + }; + + /** + * Returns an object that contains a new `CancelToken` and a function that, when called, + * cancels the `CancelToken`. + */ + CancelToken.source = function source() { + var cancel; + var token = new CancelToken(function executor(c) { + cancel = c; + }); + return { + token: token, + cancel: cancel + }; + }; + + var CancelToken_1 = CancelToken; + + /** + * Syntactic sugar for invoking a function and expanding an array for arguments. + * + * Common use case would be to use `Function.prototype.apply`. + * + * ```js + * function f(x, y, z) {} + * var args = [1, 2, 3]; + * f.apply(null, args); + * ``` + * + * With `spread` this example can be re-written. + * + * ```js + * spread(function(x, y, z) {})([1, 2, 3]); + * ``` + * + * @param {Function} callback + * @returns {Function} + */ + var spread = function spread(callback) { + return function wrap(arr) { + return callback.apply(null, arr); + }; + }; + + /** + * Determines whether the payload is an error thrown by Axios + * + * @param {*} payload The value to test + * @returns {boolean} True if the payload is an error thrown by Axios, otherwise false + */ + var isAxiosError = function isAxiosError(payload) { + return (typeof payload === 'object') && (payload.isAxiosError === true); + }; + + /** + * Create an instance of Axios + * + * @param {Object} defaultConfig The default config for the instance + * @return {Axios} A new instance of Axios + */ + function createInstance(defaultConfig) { + var context = new Axios_1(defaultConfig); + var instance = bind(Axios_1.prototype.request, context); + + // Copy axios.prototype to instance + utils.extend(instance, Axios_1.prototype, context); + + // Copy context to instance + utils.extend(instance, context); + + return instance; + } + + // Create the default instance to be exported + var axios$1 = createInstance(defaults_1); + + // Expose Axios class to allow class inheritance + axios$1.Axios = Axios_1; + + // Factory for creating new instances + axios$1.create = function create(instanceConfig) { + return createInstance(mergeConfig(axios$1.defaults, instanceConfig)); + }; + + // Expose Cancel & CancelToken + axios$1.Cancel = Cancel_1; + axios$1.CancelToken = CancelToken_1; + axios$1.isCancel = isCancel; + + // Expose all/spread + axios$1.all = function all(promises) { + return Promise.all(promises); + }; + axios$1.spread = spread; + + // Expose isAxiosError + axios$1.isAxiosError = isAxiosError; + + var axios_1 = axios$1; + + // Allow use of default import syntax in TypeScript + var _default = axios$1; + axios_1.default = _default; + + var axios = axios_1; + + class RTCEndpoint extends Event$1 { + constructor(options) { + super('RTCPusherPlayer'); + this.TAG = '[RTCPusherPlayer]'; + let defaults = { + element: '', + // html video element + debug: false, + // if output debug log + zlmsdpUrl: '', + simulecast: false + }; + this.options = Object.assign({}, defaults, options); + + if (this.options.debug) { + setLogger(); + } + + this.e = { + onicecandidate: this._onIceCandidate.bind(this), + ontrack: this._onTrack.bind(this), + onicecandidateerror: this._onIceCandidateError.bind(this) + }; + this._remoteStream = null; + this._localStream = null; + this.pc = new RTCPeerConnection(null); + this.pc.onicecandidate = this.e.onicecandidate; + this.pc.onicecandidateerror = this.e.onicecandidateerror; + this.pc.ontrack = this.e.ontrack; + this.start(); + } + + start() { + let audioConstraints = new AudioTrackConstraints(AudioSourceInfo.SCREENCAST); + let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.SCREENCAST); + MediaStreamFactory.createMediaStream(new StreamConstraints(audioConstraints, videoConstraints)).then(stream => { + this._localStream = stream; + this.dispatch(Events$1.WEBRTC_ON_LOCAL_STREAM, stream); + const AudioTransceiverInit = { + direction: 'sendrecv', + sendEncodings: [] + }; + const VideoTransceiverInit = { + direction: 'sendrecv', + sendEncodings: [] + }; + + if (this.options.simulecast) { + VideoTransceiverInit.sendEncodings = [{ + rid: 'q', + active: true, + scaleResolutionDownBy: 4.0 + }, { + rid: 'h', + active: true, + scaleResolutionDownBy: 2.0 + }, { + rid: 'f', + active: true + }]; + } + + this.pc.addTransceiver(stream.getAudioTracks()[0], AudioTransceiverInit); + this.pc.addTransceiver(stream.getVideoTracks()[0], VideoTransceiverInit); + /* + stream.getTracks().forEach((track,idx)=>{ + debug.log(this.TAG,track); + this.pc.addTrack(track); + }); + */ + + this.pc.createOffer().then(desc => { + log(this.TAG, 'offer:', desc.sdp); + this.pc.setLocalDescription(desc).then(() => { + axios({ + method: 'post', + url: this.options.zlmsdpUrl, + responseType: 'json', + data: desc.sdp, + headers: { + 'Content-Type': 'text/plain;charset=utf-8' + } + }).then(response => { + let ret = JSON.parse(response.data); + + if (ret.code != 0) { + // mean failed for offer/anwser exchange + this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); + return; + } + + let anwser = {}; + anwser.sdp = ret.sdp; + anwser.type = 'anwser'; + log(this.TAG, 'anwser:', ret.sdp); + this.pc.setRemoteDescription(anwser).then(() => { + log(this.TAG, 'set remote sucess'); + }).catch(e => { + error(this.TAG, e); + }); + }); + }); + }).catch(e => { + error(this.TAG, e); + }); + }).catch(e => { + error(this.TAG, e); + }); //const offerOptions = {}; + + /* + if (typeof this.pc.addTransceiver === 'function') { + // |direction| seems not working on Safari. + this.pc.addTransceiver('audio', { direction: 'recvonly' }); + this.pc.addTransceiver('video', { direction: 'recvonly' }); + } else { + offerOptions.offerToReceiveAudio = true; + offerOptions.offerToReceiveVideo = true; + } + */ + } + + _onIceCandidate(event) { + if (event.candidate) { + log('Remote ICE candidate: \n ' + event.candidate.candidate); // Send the candidate to the remote peer + } + } + + _onTrack(event) { + if (this.options.element && event.streams && event.streams.length > 0) { + this.options.element.srcObject = event.streams[0]; + this._remoteStream = event.streams[0]; + this.dispatch(Events$1.WEBRTC_ON_REMOTE_STREAMS, event); + } else { + error('element pararm is failed'); + } + } + + _onIceCandidateError(event) { + this.dispatch(Events$1.WEBRTC_ICE_CANDIDATE_ERROR, event); + } + + close() { + if (this.pc) { + this.pc.close(); + this.pc = null; + } + + if (this.options) { + this.options = null; + } + + if (this._localStream) { + this._localStream.getTracks().forEach((track, idx) => { + track.stop(); + }); + } + + if (this._remoteStream) { + this._remoteStream.getTracks().forEach((track, idx) => { + track.stop(); + }); + } + } + + get remoteStream() { + return this._remoteStream; + } + + get localStream() { + return this._localStream; + } + + } + + console.log('build date:', BUILD_DATE); + console.log('version:', VERSION); + const Events = Events$1; + const Media = media; + const Endpoint = RTCEndpoint; + + exports.Endpoint = Endpoint; + exports.Events = Events; + exports.Media = Media; + + Object.defineProperty(exports, '__esModule', { value: true }); + + return exports; + +}({})); +//# sourceMappingURL=ZLMRTCClient.js.map diff --git a/www/webrtc/ZLMRTCClient.js.map b/www/webrtc/ZLMRTCClient.js.map new file mode 100644 index 00000000..7f4bbb98 --- /dev/null +++ b/www/webrtc/ZLMRTCClient.js.map @@ -0,0 +1 @@ +{"version":3,"file":"ZLMRTCClient.js","sources":["../src/base/event.js","../src/ulity/version.js","../src/base/utils.js","../src/base/mediaformat.js","../node_modules/webrtc-adapter/src/js/utils.js","../node_modules/webrtc-adapter/src/js/chrome/getusermedia.js","../node_modules/webrtc-adapter/src/js/chrome/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/chrome/chrome_shim.js","../node_modules/webrtc-adapter/src/js/edge/filtericeservers.js","../node_modules/sdp/sdp.js","../node_modules/rtcpeerconnection-shim/rtcpeerconnection.js","../node_modules/webrtc-adapter/src/js/edge/getusermedia.js","../node_modules/webrtc-adapter/src/js/edge/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/edge/edge_shim.js","../node_modules/webrtc-adapter/src/js/firefox/getusermedia.js","../node_modules/webrtc-adapter/src/js/firefox/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/firefox/firefox_shim.js","../node_modules/webrtc-adapter/src/js/safari/safari_shim.js","../node_modules/webrtc-adapter/src/js/common_shim.js","../node_modules/webrtc-adapter/src/js/adapter_factory.js","../node_modules/webrtc-adapter/src/js/adapter_core.js","../src/base/mediastream-factory.js","../src/base/export.js","../src/ulity/debug.js","../src/ulity/event.js","../node_modules/axios/lib/helpers/bind.js","../node_modules/axios/lib/utils.js","../node_modules/axios/lib/helpers/buildURL.js","../node_modules/axios/lib/core/InterceptorManager.js","../node_modules/axios/lib/core/transformData.js","../node_modules/axios/lib/cancel/isCancel.js","../node_modules/axios/lib/helpers/normalizeHeaderName.js","../node_modules/axios/lib/core/enhanceError.js","../node_modules/axios/lib/core/createError.js","../node_modules/axios/lib/core/settle.js","../node_modules/axios/lib/helpers/cookies.js","../node_modules/axios/lib/helpers/isAbsoluteURL.js","../node_modules/axios/lib/helpers/combineURLs.js","../node_modules/axios/lib/core/buildFullPath.js","../node_modules/axios/lib/helpers/parseHeaders.js","../node_modules/axios/lib/helpers/isURLSameOrigin.js","../node_modules/axios/lib/adapters/xhr.js","../node_modules/axios/lib/defaults.js","../node_modules/axios/lib/core/dispatchRequest.js","../node_modules/axios/lib/core/mergeConfig.js","../node_modules/axios/lib/core/Axios.js","../node_modules/axios/lib/cancel/Cancel.js","../node_modules/axios/lib/cancel/CancelToken.js","../node_modules/axios/lib/helpers/spread.js","../node_modules/axios/lib/helpers/isAxiosError.js","../node_modules/axios/lib/axios.js","../node_modules/axios/index.js","../src/endpoint/endpoint.js","../src/export.js"],"sourcesContent":["const Events = {\n\tWEBRTC_NOT_SUPPORT : 'WEBRTC_NOT_SUPPORT',\n\tWEBRTC_ICE_CANDIDATE_ERROR : 'WEBRTC_ICE_CANDIDATE_ERROR',\n\tWEBRTC_OFFER_ANWSER_EXCHANGE_FAILED:'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED',\n\tWEBRTC_ON_REMOTE_STREAMS:'WEBRTC_ON_REMOTE_STREAMS',\n\tWEBRTC_ON_LOCAL_STREAM:'WEBRTC_ON_LOCAL_STREAM'\n};\n\nexport default Events;","export const VERSION = '__VERSION__';\nexport const BUILD_DATE = '__BUILD_DATE__';","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n\n// eslint-disable-next-line require-jsdoc\nexport function isFirefox() {\n return window.navigator.userAgent.match('Firefox') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isChrome() {\n return window.navigator.userAgent.match('Chrome') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isSafari() {\n return /^((?!chrome|android).)*safari/i.test(window.navigator.userAgent);\n}\n// eslint-disable-next-line require-jsdoc\nexport function isEdge() {\n return window.navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/) !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function createUuid() {\n return 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx'.replace(/[xy]/g, function(c) {\n const r = Math.random() * 16 | 0;\n const v = c === 'x' ? r : (r & 0x3 | 0x8);\n return v.toString(16);\n });\n}","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n/**\n * @class AudioSourceInfo\n * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const AudioSourceInfo = {\n MIC: 'mic',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class VideoSourceInfo\n * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const VideoSourceInfo = {\n CAMERA: 'camera',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class TrackKind\n * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const TrackKind = {\n /**\n * Audio tracks.\n * @type string\n */\n AUDIO: 'audio',\n /**\n * Video tracks.\n * @type string\n */\n VIDEO: 'video',\n /**\n * Both audio and video tracks.\n * @type string\n */\n AUDIO_AND_VIDEO: 'av',\n};\n/**\n * @class Resolution\n * @memberOf Owt.Base\n * @classDesc The Resolution defines the size of a rectangle.\n * @constructor\n * @param {number} width\n * @param {number} height\n */\nexport class Resolution {\n // eslint-disable-next-line require-jsdoc\n constructor(width, height) {\n /**\n * @member {number} width\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.width = width;\n /**\n * @member {number} height\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.height = height;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nlet logDisabled_ = true;\nlet deprecationWarnings_ = true;\n\n/**\n * Extract browser version out of the provided user agent string.\n *\n * @param {!string} uastring userAgent string.\n * @param {!string} expr Regular expression used as match criteria.\n * @param {!number} pos position in the version string to be returned.\n * @return {!number} browser version.\n */\nexport function extractVersion(uastring, expr, pos) {\n const match = uastring.match(expr);\n return match && match.length >= pos && parseInt(match[pos], 10);\n}\n\n// Wraps the peerconnection event eventNameToWrap in a function\n// which returns the modified event object (or false to prevent\n// the event).\nexport function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) {\n if (!window.RTCPeerConnection) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n const nativeAddEventListener = proto.addEventListener;\n proto.addEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap) {\n return nativeAddEventListener.apply(this, arguments);\n }\n const wrappedCallback = (e) => {\n const modifiedEvent = wrapper(e);\n if (modifiedEvent) {\n if (cb.handleEvent) {\n cb.handleEvent(modifiedEvent);\n } else {\n cb(modifiedEvent);\n }\n }\n };\n this._eventMap = this._eventMap || {};\n if (!this._eventMap[eventNameToWrap]) {\n this._eventMap[eventNameToWrap] = new Map();\n }\n this._eventMap[eventNameToWrap].set(cb, wrappedCallback);\n return nativeAddEventListener.apply(this, [nativeEventName,\n wrappedCallback]);\n };\n\n const nativeRemoveEventListener = proto.removeEventListener;\n proto.removeEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap || !this._eventMap\n || !this._eventMap[eventNameToWrap]) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n if (!this._eventMap[eventNameToWrap].has(cb)) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n const unwrappedCb = this._eventMap[eventNameToWrap].get(cb);\n this._eventMap[eventNameToWrap].delete(cb);\n if (this._eventMap[eventNameToWrap].size === 0) {\n delete this._eventMap[eventNameToWrap];\n }\n if (Object.keys(this._eventMap).length === 0) {\n delete this._eventMap;\n }\n return nativeRemoveEventListener.apply(this, [nativeEventName,\n unwrappedCb]);\n };\n\n Object.defineProperty(proto, 'on' + eventNameToWrap, {\n get() {\n return this['_on' + eventNameToWrap];\n },\n set(cb) {\n if (this['_on' + eventNameToWrap]) {\n this.removeEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap]);\n delete this['_on' + eventNameToWrap];\n }\n if (cb) {\n this.addEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap] = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n}\n\nexport function disableLog(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n logDisabled_ = bool;\n return (bool) ? 'adapter.js logging disabled' :\n 'adapter.js logging enabled';\n}\n\n/**\n * Disable or enable deprecation warnings\n * @param {!boolean} bool set to true to disable warnings.\n */\nexport function disableWarnings(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n deprecationWarnings_ = !bool;\n return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled');\n}\n\nexport function log() {\n if (typeof window === 'object') {\n if (logDisabled_) {\n return;\n }\n if (typeof console !== 'undefined' && typeof console.log === 'function') {\n console.log.apply(console, arguments);\n }\n }\n}\n\n/**\n * Shows a deprecation warning suggesting the modern and spec-compatible API.\n */\nexport function deprecated(oldMethod, newMethod) {\n if (!deprecationWarnings_) {\n return;\n }\n console.warn(oldMethod + ' is deprecated, please use ' + newMethod +\n ' instead.');\n}\n\n/**\n * Browser detector.\n *\n * @return {object} result containing browser and version\n * properties.\n */\nexport function detectBrowser(window) {\n // Returned result object.\n const result = {browser: null, version: null};\n\n // Fail early if it's not a browser\n if (typeof window === 'undefined' || !window.navigator) {\n result.browser = 'Not a browser.';\n return result;\n }\n\n const {navigator} = window;\n\n if (navigator.mozGetUserMedia) { // Firefox.\n result.browser = 'firefox';\n result.version = extractVersion(navigator.userAgent,\n /Firefox\\/(\\d+)\\./, 1);\n } else if (navigator.webkitGetUserMedia ||\n (window.isSecureContext === false && window.webkitRTCPeerConnection &&\n !window.RTCIceGatherer)) {\n // Chrome, Chromium, Webview, Opera.\n // Version matches Chrome/WebRTC version.\n // Chrome 74 removed webkitGetUserMedia on http as well so we need the\n // more complicated fallback to webkitRTCPeerConnection.\n result.browser = 'chrome';\n result.version = extractVersion(navigator.userAgent,\n /Chrom(e|ium)\\/(\\d+)\\./, 2);\n } else if (navigator.mediaDevices &&\n navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/)) { // Edge.\n result.browser = 'edge';\n result.version = extractVersion(navigator.userAgent,\n /Edge\\/(\\d+).(\\d+)$/, 2);\n } else if (window.RTCPeerConnection &&\n navigator.userAgent.match(/AppleWebKit\\/(\\d+)\\./)) { // Safari.\n result.browser = 'safari';\n result.version = extractVersion(navigator.userAgent,\n /AppleWebKit\\/(\\d+)\\./, 1);\n result.supportsUnifiedPlan = window.RTCRtpTransceiver &&\n 'currentDirection' in window.RTCRtpTransceiver.prototype;\n } else { // Default fallthrough: not supported.\n result.browser = 'Not a supported browser.';\n return result;\n }\n\n return result;\n}\n\n/**\n * Checks if something is an object.\n *\n * @param {*} val The something you want to check.\n * @return true if val is an object, false otherwise.\n */\nfunction isObject(val) {\n return Object.prototype.toString.call(val) === '[object Object]';\n}\n\n/**\n * Remove all empty objects and undefined values\n * from a nested object -- an enhanced and vanilla version\n * of Lodash's `compact`.\n */\nexport function compactObject(data) {\n if (!isObject(data)) {\n return data;\n }\n\n return Object.keys(data).reduce(function(accumulator, key) {\n const isObj = isObject(data[key]);\n const value = isObj ? compactObject(data[key]) : data[key];\n const isEmptyObject = isObj && !Object.keys(value).length;\n if (value === undefined || isEmptyObject) {\n return accumulator;\n }\n return Object.assign(accumulator, {[key]: value});\n }, {});\n}\n\n/* iterates the stats graph recursively. */\nexport function walkStats(stats, base, resultSet) {\n if (!base || resultSet.has(base.id)) {\n return;\n }\n resultSet.set(base.id, base);\n Object.keys(base).forEach(name => {\n if (name.endsWith('Id')) {\n walkStats(stats, stats.get(base[name]), resultSet);\n } else if (name.endsWith('Ids')) {\n base[name].forEach(id => {\n walkStats(stats, stats.get(id), resultSet);\n });\n }\n });\n}\n\n/* filter getStats for a sender/receiver track. */\nexport function filterStats(result, track, outbound) {\n const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp';\n const filteredResult = new Map();\n if (track === null) {\n return filteredResult;\n }\n const trackStats = [];\n result.forEach(value => {\n if (value.type === 'track' &&\n value.trackIdentifier === track.id) {\n trackStats.push(value);\n }\n });\n trackStats.forEach(trackStat => {\n result.forEach(stats => {\n if (stats.type === streamStatsType && stats.trackId === trackStat.id) {\n walkStats(result, stats, filteredResult);\n }\n });\n });\n return filteredResult;\n}\n\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\nconst logging = utils.log;\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n\n if (!navigator.mediaDevices) {\n return;\n }\n\n const constraintsToChrome_ = function(c) {\n if (typeof c !== 'object' || c.mandatory || c.optional) {\n return c;\n }\n const cc = {};\n Object.keys(c).forEach(key => {\n if (key === 'require' || key === 'advanced' || key === 'mediaSource') {\n return;\n }\n const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]};\n if (r.exact !== undefined && typeof r.exact === 'number') {\n r.min = r.max = r.exact;\n }\n const oldname_ = function(prefix, name) {\n if (prefix) {\n return prefix + name.charAt(0).toUpperCase() + name.slice(1);\n }\n return (name === 'deviceId') ? 'sourceId' : name;\n };\n if (r.ideal !== undefined) {\n cc.optional = cc.optional || [];\n let oc = {};\n if (typeof r.ideal === 'number') {\n oc[oldname_('min', key)] = r.ideal;\n cc.optional.push(oc);\n oc = {};\n oc[oldname_('max', key)] = r.ideal;\n cc.optional.push(oc);\n } else {\n oc[oldname_('', key)] = r.ideal;\n cc.optional.push(oc);\n }\n }\n if (r.exact !== undefined && typeof r.exact !== 'number') {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_('', key)] = r.exact;\n } else {\n ['min', 'max'].forEach(mix => {\n if (r[mix] !== undefined) {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_(mix, key)] = r[mix];\n }\n });\n }\n });\n if (c.advanced) {\n cc.optional = (cc.optional || []).concat(c.advanced);\n }\n return cc;\n };\n\n const shimConstraints_ = function(constraints, func) {\n if (browserDetails.version >= 61) {\n return func(constraints);\n }\n constraints = JSON.parse(JSON.stringify(constraints));\n if (constraints && typeof constraints.audio === 'object') {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n constraints = JSON.parse(JSON.stringify(constraints));\n remap(constraints.audio, 'autoGainControl', 'googAutoGainControl');\n remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression');\n constraints.audio = constraintsToChrome_(constraints.audio);\n }\n if (constraints && typeof constraints.video === 'object') {\n // Shim facingMode for mobile & surface pro.\n let face = constraints.video.facingMode;\n face = face && ((typeof face === 'object') ? face : {ideal: face});\n const getSupportedFacingModeLies = browserDetails.version < 66;\n\n if ((face && (face.exact === 'user' || face.exact === 'environment' ||\n face.ideal === 'user' || face.ideal === 'environment')) &&\n !(navigator.mediaDevices.getSupportedConstraints &&\n navigator.mediaDevices.getSupportedConstraints().facingMode &&\n !getSupportedFacingModeLies)) {\n delete constraints.video.facingMode;\n let matches;\n if (face.exact === 'environment' || face.ideal === 'environment') {\n matches = ['back', 'rear'];\n } else if (face.exact === 'user' || face.ideal === 'user') {\n matches = ['front'];\n }\n if (matches) {\n // Look for matches in label, or use last cam for back (typical).\n return navigator.mediaDevices.enumerateDevices()\n .then(devices => {\n devices = devices.filter(d => d.kind === 'videoinput');\n let dev = devices.find(d => matches.some(match =>\n d.label.toLowerCase().includes(match)));\n if (!dev && devices.length && matches.includes('back')) {\n dev = devices[devices.length - 1]; // more likely the back cam\n }\n if (dev) {\n constraints.video.deviceId = face.exact ? {exact: dev.deviceId} :\n {ideal: dev.deviceId};\n }\n constraints.video = constraintsToChrome_(constraints.video);\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n });\n }\n }\n constraints.video = constraintsToChrome_(constraints.video);\n }\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n };\n\n const shimError_ = function(e) {\n if (browserDetails.version >= 64) {\n return e;\n }\n return {\n name: {\n PermissionDeniedError: 'NotAllowedError',\n PermissionDismissedError: 'NotAllowedError',\n InvalidStateError: 'NotAllowedError',\n DevicesNotFoundError: 'NotFoundError',\n ConstraintNotSatisfiedError: 'OverconstrainedError',\n TrackStartError: 'NotReadableError',\n MediaDeviceFailedDueToShutdown: 'NotAllowedError',\n MediaDeviceKillSwitchOn: 'NotAllowedError',\n TabCaptureError: 'AbortError',\n ScreenCaptureError: 'AbortError',\n DeviceCaptureError: 'AbortError'\n }[e.name] || e.name,\n message: e.message,\n constraint: e.constraint || e.constraintName,\n toString() {\n return this.name + (this.message && ': ') + this.message;\n }\n };\n };\n\n const getUserMedia_ = function(constraints, onSuccess, onError) {\n shimConstraints_(constraints, c => {\n navigator.webkitGetUserMedia(c, onSuccess, e => {\n if (onError) {\n onError(shimError_(e));\n }\n });\n });\n };\n navigator.getUserMedia = getUserMedia_.bind(navigator);\n\n // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia\n // function which returns a Promise, it does not accept spec-style\n // constraints.\n if (navigator.mediaDevices.getUserMedia) {\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(cs) {\n return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => {\n if (c.audio && !stream.getAudioTracks().length ||\n c.video && !stream.getVideoTracks().length) {\n stream.getTracks().forEach(track => {\n track.stop();\n });\n throw new DOMException('', 'NotFoundError');\n }\n return stream;\n }, e => Promise.reject(shimError_(e))));\n };\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nexport function shimGetDisplayMedia(window, getSourceId) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n // getSourceId is a function that returns a promise resolving with\n // the sourceId of the screen/window/tab to be shared.\n if (typeof getSourceId !== 'function') {\n console.error('shimGetDisplayMedia: getSourceId argument is not ' +\n 'a function');\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n return getSourceId(constraints)\n .then(sourceId => {\n const widthSpecified = constraints.video && constraints.video.width;\n const heightSpecified = constraints.video &&\n constraints.video.height;\n const frameRateSpecified = constraints.video &&\n constraints.video.frameRate;\n constraints.video = {\n mandatory: {\n chromeMediaSource: 'desktop',\n chromeMediaSourceId: sourceId,\n maxFrameRate: frameRateSpecified || 3\n }\n };\n if (widthSpecified) {\n constraints.video.mandatory.maxWidth = widthSpecified;\n }\n if (heightSpecified) {\n constraints.video.mandatory.maxHeight = heightSpecified;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n });\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimMediaStream(window) {\n window.MediaStream = window.MediaStream || window.webkitMediaStream;\n}\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in\n window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', {\n get() {\n return this._ontrack;\n },\n set(f) {\n if (this._ontrack) {\n this.removeEventListener('track', this._ontrack);\n }\n this.addEventListener('track', this._ontrack = f);\n },\n enumerable: true,\n configurable: true\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n if (!this._ontrackpoly) {\n this._ontrackpoly = (e) => {\n // onaddstream does not fire when a track is added to an existing\n // stream. But stream.onaddtrack is implemented so we use that.\n e.stream.addEventListener('addtrack', te => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === te.track.id);\n } else {\n receiver = {track: te.track};\n }\n\n const event = new Event('track');\n event.track = te.track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n e.stream.getTracks().forEach(track => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === track.id);\n } else {\n receiver = {track};\n }\n const event = new Event('track');\n event.track = track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n };\n this.addEventListener('addstream', this._ontrackpoly);\n }\n return origSetRemoteDescription.apply(this, arguments);\n };\n } else {\n // even if RTCRtpTransceiver is in window, it is only used and\n // emitted in unified-plan. Unfortunately this means we need\n // to unconditionally wrap the event.\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n if (!e.transceiver) {\n Object.defineProperty(e, 'transceiver',\n {value: {receiver: e.receiver}});\n }\n return e;\n });\n }\n}\n\nexport function shimGetSendersWithDtmf(window) {\n // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack.\n if (typeof window === 'object' && window.RTCPeerConnection &&\n !('getSenders' in window.RTCPeerConnection.prototype) &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype) {\n const shimSenderWithDtmf = function(pc, track) {\n return {\n track,\n get dtmf() {\n if (this._dtmf === undefined) {\n if (track.kind === 'audio') {\n this._dtmf = pc.createDTMFSender(track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n },\n _pc: pc\n };\n };\n\n // augment addTrack when getSenders is not available.\n if (!window.RTCPeerConnection.prototype.getSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n this._senders = this._senders || [];\n return this._senders.slice(); // return a copy of the internal state.\n };\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n let sender = origAddTrack.apply(this, arguments);\n if (!sender) {\n sender = shimSenderWithDtmf(this, track);\n this._senders.push(sender);\n }\n return sender;\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n origRemoveTrack.apply(this, arguments);\n const idx = this._senders.indexOf(sender);\n if (idx !== -1) {\n this._senders.splice(idx, 1);\n }\n };\n }\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._senders = this._senders || [];\n origAddStream.apply(this, [stream]);\n stream.getTracks().forEach(track => {\n this._senders.push(shimSenderWithDtmf(this, track));\n });\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._senders = this._senders || [];\n origRemoveStream.apply(this, [stream]);\n\n stream.getTracks().forEach(track => {\n const sender = this._senders.find(s => s.track === track);\n if (sender) { // remove sender\n this._senders.splice(this._senders.indexOf(sender), 1);\n }\n });\n };\n } else if (typeof window === 'object' && window.RTCPeerConnection &&\n 'getSenders' in window.RTCPeerConnection.prototype &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype &&\n window.RTCRtpSender &&\n !('dtmf' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = this._pc.createDTMFSender(this.track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n}\n\nexport function shimGetStats(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n\n // If selector is a function then we are in the old style stats so just\n // pass back the original getStats format to avoid breaking old users.\n if (arguments.length > 0 && typeof selector === 'function') {\n return origGetStats.apply(this, arguments);\n }\n\n // When spec-style getStats is supported, return those when called with\n // either no arguments or the selector argument is null.\n if (origGetStats.length === 0 && (arguments.length === 0 ||\n typeof selector !== 'function')) {\n return origGetStats.apply(this, []);\n }\n\n const fixChromeStats_ = function(response) {\n const standardReport = {};\n const reports = response.result();\n reports.forEach(report => {\n const standardStats = {\n id: report.id,\n timestamp: report.timestamp,\n type: {\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[report.type] || report.type\n };\n report.names().forEach(name => {\n standardStats[name] = report.stat(name);\n });\n standardReport[standardStats.id] = standardStats;\n });\n\n return standardReport;\n };\n\n // shim getStats with maplike support\n const makeMapStats = function(stats) {\n return new Map(Object.keys(stats).map(key => [key, stats[key]]));\n };\n\n if (arguments.length >= 2) {\n const successCallbackWrapper_ = function(response) {\n onSucc(makeMapStats(fixChromeStats_(response)));\n };\n\n return origGetStats.apply(this, [successCallbackWrapper_,\n selector]);\n }\n\n // promise-support\n return new Promise((resolve, reject) => {\n origGetStats.apply(this, [\n function(response) {\n resolve(makeMapStats(fixChromeStats_(response)));\n }, reject]);\n }).then(onSucc, onErr);\n };\n}\n\nexport function shimSenderReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender && window.RTCRtpReceiver)) {\n return;\n }\n\n // shim sender stats.\n if (!('getStats' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n const sender = this;\n return this._pc.getStats().then(result =>\n /* Note: this will include stats of all senders that\n * send a track with the same id as sender.track as\n * it is not possible to identify the RTCRtpSender.\n */\n utils.filterStats(result, sender.track, true));\n };\n }\n\n // shim receiver stats.\n if (!('getStats' in window.RTCRtpReceiver.prototype)) {\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers =\n function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n const receiver = this;\n return this._pc.getStats().then(result =>\n utils.filterStats(result, receiver.track, false));\n };\n }\n\n if (!('getStats' in window.RTCRtpSender.prototype &&\n 'getStats' in window.RTCRtpReceiver.prototype)) {\n return;\n }\n\n // shim RTCPeerConnection.getStats(track).\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n if (arguments.length > 0 &&\n arguments[0] instanceof window.MediaStreamTrack) {\n const track = arguments[0];\n let sender;\n let receiver;\n let err;\n this.getSenders().forEach(s => {\n if (s.track === track) {\n if (sender) {\n err = true;\n } else {\n sender = s;\n }\n }\n });\n this.getReceivers().forEach(r => {\n if (r.track === track) {\n if (receiver) {\n err = true;\n } else {\n receiver = r;\n }\n }\n return r.track === track;\n });\n if (err || (sender && receiver)) {\n return Promise.reject(new DOMException(\n 'There are more than one sender or receiver for the track.',\n 'InvalidAccessError'));\n } else if (sender) {\n return sender.getStats();\n } else if (receiver) {\n return receiver.getStats();\n }\n return Promise.reject(new DOMException(\n 'There is no sender or receiver for the track.',\n 'InvalidAccessError'));\n }\n return origGetStats.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrackWithNative(window) {\n // shim addTrack/removeTrack with native variants in order to make\n // the interactions with legacy getLocalStreams behave as in other browsers.\n // Keeps a mapping stream.id => [stream, rtpsenders...]\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n return Object.keys(this._shimmedLocalStreams)\n .map(streamId => this._shimmedLocalStreams[streamId][0]);\n };\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (!stream) {\n return origAddTrack.apply(this, arguments);\n }\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n const sender = origAddTrack.apply(this, arguments);\n if (!this._shimmedLocalStreams[stream.id]) {\n this._shimmedLocalStreams[stream.id] = [stream, sender];\n } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) {\n this._shimmedLocalStreams[stream.id].push(sender);\n }\n return sender;\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n const existingSenders = this.getSenders();\n origAddStream.apply(this, arguments);\n const newSenders = this.getSenders()\n .filter(newSender => existingSenders.indexOf(newSender) === -1);\n this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n delete this._shimmedLocalStreams[stream.id];\n return origRemoveStream.apply(this, arguments);\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n if (sender) {\n Object.keys(this._shimmedLocalStreams).forEach(streamId => {\n const idx = this._shimmedLocalStreams[streamId].indexOf(sender);\n if (idx !== -1) {\n this._shimmedLocalStreams[streamId].splice(idx, 1);\n }\n if (this._shimmedLocalStreams[streamId].length === 1) {\n delete this._shimmedLocalStreams[streamId];\n }\n });\n }\n return origRemoveTrack.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrack(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // shim addTrack and removeTrack.\n if (window.RTCPeerConnection.prototype.addTrack &&\n browserDetails.version >= 65) {\n return shimAddTrackRemoveTrackWithNative(window);\n }\n\n // also shim pc.getLocalStreams when addTrack is shimmed\n // to return the original streams.\n const origGetLocalStreams = window.RTCPeerConnection.prototype\n .getLocalStreams;\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n const nativeStreams = origGetLocalStreams.apply(this);\n this._reverseStreams = this._reverseStreams || {};\n return nativeStreams.map(stream => this._reverseStreams[stream.id]);\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n // Add identity mapping for consistency with addTrack.\n // Unless this is being used with a stream from addTrack.\n if (!this._reverseStreams[stream.id]) {\n const newStream = new window.MediaStream(stream.getTracks());\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n stream = newStream;\n }\n origAddStream.apply(this, [stream]);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]);\n delete this._reverseStreams[(this._streams[stream.id] ?\n this._streams[stream.id].id : stream.id)];\n delete this._streams[stream.id];\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n const streams = [].slice.call(arguments, 1);\n if (streams.length !== 1 ||\n !streams[0].getTracks().find(t => t === track)) {\n // this is not fully correct but all we can manage without\n // [[associated MediaStreams]] internal slot.\n throw new DOMException(\n 'The adapter.js addTrack polyfill only supports a single ' +\n ' stream which is associated with the specified track.',\n 'NotSupportedError');\n }\n\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n const oldStream = this._streams[stream.id];\n if (oldStream) {\n // this is using odd Chrome behaviour, use with caution:\n // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815\n // Note: we rely on the high-level addTrack/dtmf shim to\n // create the sender with a dtmf sender.\n oldStream.addTrack(track);\n\n // Trigger ONN async.\n Promise.resolve().then(() => {\n this.dispatchEvent(new Event('negotiationneeded'));\n });\n } else {\n const newStream = new window.MediaStream([track]);\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n this.addStream(newStream);\n }\n return this.getSenders().find(s => s.track === track);\n };\n\n // replace the internal stream id with the external one and\n // vice versa.\n function replaceInternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(internalStream.id, 'g'),\n externalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n function replaceExternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(externalStream.id, 'g'),\n internalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n ['createOffer', 'createAnswer'].forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n const args = arguments;\n const isLegacyCall = arguments.length &&\n typeof arguments[0] === 'function';\n if (isLegacyCall) {\n return nativeMethod.apply(this, [\n (description) => {\n const desc = replaceInternalStreamId(this, description);\n args[0].apply(null, [desc]);\n },\n (err) => {\n if (args[1]) {\n args[1].apply(null, err);\n }\n }, arguments[2]\n ]);\n }\n return nativeMethod.apply(this, arguments)\n .then(description => replaceInternalStreamId(this, description));\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n\n const origSetLocalDescription =\n window.RTCPeerConnection.prototype.setLocalDescription;\n window.RTCPeerConnection.prototype.setLocalDescription =\n function setLocalDescription() {\n if (!arguments.length || !arguments[0].type) {\n return origSetLocalDescription.apply(this, arguments);\n }\n arguments[0] = replaceExternalStreamId(this, arguments[0]);\n return origSetLocalDescription.apply(this, arguments);\n };\n\n // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier\n\n const origLocalDescription = Object.getOwnPropertyDescriptor(\n window.RTCPeerConnection.prototype, 'localDescription');\n Object.defineProperty(window.RTCPeerConnection.prototype,\n 'localDescription', {\n get() {\n const description = origLocalDescription.get.apply(this);\n if (description.type === '') {\n return description;\n }\n return replaceInternalStreamId(this, description);\n }\n });\n\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n // We can not yet check for sender instanceof RTCRtpSender\n // since we shim RTPSender. So we check if sender._pc is set.\n if (!sender._pc) {\n throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.', 'TypeError');\n }\n const isLocal = sender._pc === this;\n if (!isLocal) {\n throw new DOMException('Sender was not created by this connection.',\n 'InvalidAccessError');\n }\n\n // Search for the native stream the senders track belongs to.\n this._streams = this._streams || {};\n let stream;\n Object.keys(this._streams).forEach(streamid => {\n const hasTrack = this._streams[streamid].getTracks()\n .find(track => sender.track === track);\n if (hasTrack) {\n stream = this._streams[streamid];\n }\n });\n\n if (stream) {\n if (stream.getTracks().length === 1) {\n // if this is the last track of the stream, remove the stream. This\n // takes care of any shimmed _senders.\n this.removeStream(this._reverseStreams[stream.id]);\n } else {\n // relying on the same odd chrome behaviour as above.\n stream.removeTrack(sender.track);\n }\n this.dispatchEvent(new Event('negotiationneeded'));\n }\n };\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.webkitRTCPeerConnection;\n }\n if (!window.RTCPeerConnection) {\n return;\n }\n\n // shim implicit creation of RTCSessionDescription/RTCIceCandidate\n if (browserDetails.version < 53) {\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n}\n\n// Attempt to fix ONN in plan-b mode.\nexport function fixNegotiationNeeded(window, browserDetails) {\n utils.wrapPeerConnectionEvent(window, 'negotiationneeded', e => {\n const pc = e.target;\n if (browserDetails.version < 72 || (pc.getConfiguration &&\n pc.getConfiguration().sdpSemantics === 'plan-b')) {\n if (pc.signalingState !== 'stable') {\n return;\n }\n }\n return e;\n });\n}\n","/*\n * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nexport function filterIceServers(iceServers, edgeVersion) {\n let hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(server => {\n if (server && (server.urls || server.url)) {\n let urls = server.urls || server.url;\n if (server.url && !server.urls) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n }\n const isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(url => {\n // filter STUN unconditionally.\n if (url.indexOf('stun:') === 0) {\n return false;\n }\n\n const validTurn = url.startsWith('turn') &&\n !url.startsWith('turn:[') &&\n url.includes('transport=udp');\n if (validTurn && !hasTurn) {\n hasTurn = true;\n return true;\n }\n return validTurn && !hasTurn;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n","/* eslint-env node */\n'use strict';\n\n// SDP helpers.\nvar SDPUtils = {};\n\n// Generate an alphanumeric identifier for cname or mids.\n// TODO: use UUIDs instead? https://gist.github.com/jed/982883\nSDPUtils.generateIdentifier = function() {\n return Math.random().toString(36).substr(2, 10);\n};\n\n// The RTCP CNAME used by all peerconnections from the same JS.\nSDPUtils.localCName = SDPUtils.generateIdentifier();\n\n// Splits SDP into lines, dealing with both CRLF and LF.\nSDPUtils.splitLines = function(blob) {\n return blob.trim().split('\\n').map(function(line) {\n return line.trim();\n });\n};\n// Splits SDP into sessionpart and mediasections. Ensures CRLF.\nSDPUtils.splitSections = function(blob) {\n var parts = blob.split('\\nm=');\n return parts.map(function(part, index) {\n return (index > 0 ? 'm=' + part : part).trim() + '\\r\\n';\n });\n};\n\n// returns the session description.\nSDPUtils.getDescription = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n return sections && sections[0];\n};\n\n// returns the individual media sections.\nSDPUtils.getMediaSections = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n sections.shift();\n return sections;\n};\n\n// Returns lines that start with a certain prefix.\nSDPUtils.matchPrefix = function(blob, prefix) {\n return SDPUtils.splitLines(blob).filter(function(line) {\n return line.indexOf(prefix) === 0;\n });\n};\n\n// Parses an ICE candidate line. Sample input:\n// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8\n// rport 55996\"\nSDPUtils.parseCandidate = function(line) {\n var parts;\n // Parse both variants.\n if (line.indexOf('a=candidate:') === 0) {\n parts = line.substring(12).split(' ');\n } else {\n parts = line.substring(10).split(' ');\n }\n\n var candidate = {\n foundation: parts[0],\n component: parseInt(parts[1], 10),\n protocol: parts[2].toLowerCase(),\n priority: parseInt(parts[3], 10),\n ip: parts[4],\n address: parts[4], // address is an alias for ip.\n port: parseInt(parts[5], 10),\n // skip parts[6] == 'typ'\n type: parts[7]\n };\n\n for (var i = 8; i < parts.length; i += 2) {\n switch (parts[i]) {\n case 'raddr':\n candidate.relatedAddress = parts[i + 1];\n break;\n case 'rport':\n candidate.relatedPort = parseInt(parts[i + 1], 10);\n break;\n case 'tcptype':\n candidate.tcpType = parts[i + 1];\n break;\n case 'ufrag':\n candidate.ufrag = parts[i + 1]; // for backward compability.\n candidate.usernameFragment = parts[i + 1];\n break;\n default: // extension handling, in particular ufrag\n candidate[parts[i]] = parts[i + 1];\n break;\n }\n }\n return candidate;\n};\n\n// Translates a candidate object into SDP candidate attribute.\nSDPUtils.writeCandidate = function(candidate) {\n var sdp = [];\n sdp.push(candidate.foundation);\n sdp.push(candidate.component);\n sdp.push(candidate.protocol.toUpperCase());\n sdp.push(candidate.priority);\n sdp.push(candidate.address || candidate.ip);\n sdp.push(candidate.port);\n\n var type = candidate.type;\n sdp.push('typ');\n sdp.push(type);\n if (type !== 'host' && candidate.relatedAddress &&\n candidate.relatedPort) {\n sdp.push('raddr');\n sdp.push(candidate.relatedAddress);\n sdp.push('rport');\n sdp.push(candidate.relatedPort);\n }\n if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') {\n sdp.push('tcptype');\n sdp.push(candidate.tcpType);\n }\n if (candidate.usernameFragment || candidate.ufrag) {\n sdp.push('ufrag');\n sdp.push(candidate.usernameFragment || candidate.ufrag);\n }\n return 'candidate:' + sdp.join(' ');\n};\n\n// Parses an ice-options line, returns an array of option tags.\n// a=ice-options:foo bar\nSDPUtils.parseIceOptions = function(line) {\n return line.substr(14).split(' ');\n};\n\n// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:\n// a=rtpmap:111 opus/48000/2\nSDPUtils.parseRtpMap = function(line) {\n var parts = line.substr(9).split(' ');\n var parsed = {\n payloadType: parseInt(parts.shift(), 10) // was: id\n };\n\n parts = parts[0].split('/');\n\n parsed.name = parts[0];\n parsed.clockRate = parseInt(parts[1], 10); // was: clockrate\n parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;\n // legacy alias, got renamed back to channels in ORTC.\n parsed.numChannels = parsed.channels;\n return parsed;\n};\n\n// Generate an a=rtpmap line from RTCRtpCodecCapability or\n// RTCRtpCodecParameters.\nSDPUtils.writeRtpMap = function(codec) {\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n var channels = codec.channels || codec.numChannels || 1;\n return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate +\n (channels !== 1 ? '/' + channels : '') + '\\r\\n';\n};\n\n// Parses an a=extmap line (headerextension from RFC 5285). Sample input:\n// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset\nSDPUtils.parseExtmap = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n id: parseInt(parts[0], 10),\n direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv',\n uri: parts[1]\n };\n};\n\n// Generates a=extmap line from RTCRtpHeaderExtensionParameters or\n// RTCRtpHeaderExtension.\nSDPUtils.writeExtmap = function(headerExtension) {\n return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) +\n (headerExtension.direction && headerExtension.direction !== 'sendrecv'\n ? '/' + headerExtension.direction\n : '') +\n ' ' + headerExtension.uri + '\\r\\n';\n};\n\n// Parses an ftmp line, returns dictionary. Sample input:\n// a=fmtp:96 vbr=on;cng=on\n// Also deals with vbr=on; cng=on\nSDPUtils.parseFmtp = function(line) {\n var parsed = {};\n var kv;\n var parts = line.substr(line.indexOf(' ') + 1).split(';');\n for (var j = 0; j < parts.length; j++) {\n kv = parts[j].trim().split('=');\n parsed[kv[0].trim()] = kv[1];\n }\n return parsed;\n};\n\n// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeFmtp = function(codec) {\n var line = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.parameters && Object.keys(codec.parameters).length) {\n var params = [];\n Object.keys(codec.parameters).forEach(function(param) {\n if (codec.parameters[param]) {\n params.push(param + '=' + codec.parameters[param]);\n } else {\n params.push(param);\n }\n });\n line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\\r\\n';\n }\n return line;\n};\n\n// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:\n// a=rtcp-fb:98 nack rpsi\nSDPUtils.parseRtcpFb = function(line) {\n var parts = line.substr(line.indexOf(' ') + 1).split(' ');\n return {\n type: parts.shift(),\n parameter: parts.join(' ')\n };\n};\n// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeRtcpFb = function(codec) {\n var lines = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.rtcpFeedback && codec.rtcpFeedback.length) {\n // FIXME: special handling for trr-int?\n codec.rtcpFeedback.forEach(function(fb) {\n lines += 'a=rtcp-fb:' + pt + ' ' + fb.type +\n (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') +\n '\\r\\n';\n });\n }\n return lines;\n};\n\n// Parses an RFC 5576 ssrc media attribute. Sample input:\n// a=ssrc:3735928559 cname:something\nSDPUtils.parseSsrcMedia = function(line) {\n var sp = line.indexOf(' ');\n var parts = {\n ssrc: parseInt(line.substr(7, sp - 7), 10)\n };\n var colon = line.indexOf(':', sp);\n if (colon > -1) {\n parts.attribute = line.substr(sp + 1, colon - sp - 1);\n parts.value = line.substr(colon + 1);\n } else {\n parts.attribute = line.substr(sp + 1);\n }\n return parts;\n};\n\nSDPUtils.parseSsrcGroup = function(line) {\n var parts = line.substr(13).split(' ');\n return {\n semantics: parts.shift(),\n ssrcs: parts.map(function(ssrc) {\n return parseInt(ssrc, 10);\n })\n };\n};\n\n// Extracts the MID (RFC 5888) from a media section.\n// returns the MID or undefined if no mid line was found.\nSDPUtils.getMid = function(mediaSection) {\n var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0];\n if (mid) {\n return mid.substr(6);\n }\n};\n\nSDPUtils.parseFingerprint = function(line) {\n var parts = line.substr(14).split(' ');\n return {\n algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.\n value: parts[1]\n };\n};\n\n// Extracts DTLS parameters from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the fingerprint line as input. See also getIceParameters.\nSDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=fingerprint:');\n // Note: a=setup line is ignored since we use the 'auto' role.\n // Note2: 'algorithm' is not case sensitive except in Edge.\n return {\n role: 'auto',\n fingerprints: lines.map(SDPUtils.parseFingerprint)\n };\n};\n\n// Serializes DTLS parameters to SDP.\nSDPUtils.writeDtlsParameters = function(params, setupType) {\n var sdp = 'a=setup:' + setupType + '\\r\\n';\n params.fingerprints.forEach(function(fp) {\n sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\\r\\n';\n });\n return sdp;\n};\n\n// Parses a=crypto lines into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members\nSDPUtils.parseCryptoLine = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n tag: parseInt(parts[0], 10),\n cryptoSuite: parts[1],\n keyParams: parts[2],\n sessionParams: parts.slice(3),\n };\n};\n\nSDPUtils.writeCryptoLine = function(parameters) {\n return 'a=crypto:' + parameters.tag + ' ' +\n parameters.cryptoSuite + ' ' +\n (typeof parameters.keyParams === 'object'\n ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)\n : parameters.keyParams) +\n (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') +\n '\\r\\n';\n};\n\n// Parses the crypto key parameters into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*\nSDPUtils.parseCryptoKeyParams = function(keyParams) {\n if (keyParams.indexOf('inline:') !== 0) {\n return null;\n }\n var parts = keyParams.substr(7).split('|');\n return {\n keyMethod: 'inline',\n keySalt: parts[0],\n lifeTime: parts[1],\n mkiValue: parts[2] ? parts[2].split(':')[0] : undefined,\n mkiLength: parts[2] ? parts[2].split(':')[1] : undefined,\n };\n};\n\nSDPUtils.writeCryptoKeyParams = function(keyParams) {\n return keyParams.keyMethod + ':'\n + keyParams.keySalt +\n (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') +\n (keyParams.mkiValue && keyParams.mkiLength\n ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength\n : '');\n};\n\n// Extracts all SDES paramters.\nSDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=crypto:');\n return lines.map(SDPUtils.parseCryptoLine);\n};\n\n// Parses ICE information from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the ice-ufrag and ice-pwd lines as input.\nSDPUtils.getIceParameters = function(mediaSection, sessionpart) {\n var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-ufrag:')[0];\n var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-pwd:')[0];\n if (!(ufrag && pwd)) {\n return null;\n }\n return {\n usernameFragment: ufrag.substr(12),\n password: pwd.substr(10),\n };\n};\n\n// Serializes ICE parameters to SDP.\nSDPUtils.writeIceParameters = function(params) {\n return 'a=ice-ufrag:' + params.usernameFragment + '\\r\\n' +\n 'a=ice-pwd:' + params.password + '\\r\\n';\n};\n\n// Parses the SDP media section and returns RTCRtpParameters.\nSDPUtils.parseRtpParameters = function(mediaSection) {\n var description = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: [],\n rtcp: []\n };\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..]\n var pt = mline[i];\n var rtpmapline = SDPUtils.matchPrefix(\n mediaSection, 'a=rtpmap:' + pt + ' ')[0];\n if (rtpmapline) {\n var codec = SDPUtils.parseRtpMap(rtpmapline);\n var fmtps = SDPUtils.matchPrefix(\n mediaSection, 'a=fmtp:' + pt + ' ');\n // Only the first a=fmtp: is considered.\n codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};\n codec.rtcpFeedback = SDPUtils.matchPrefix(\n mediaSection, 'a=rtcp-fb:' + pt + ' ')\n .map(SDPUtils.parseRtcpFb);\n description.codecs.push(codec);\n // parse FEC mechanisms from rtpmap lines.\n switch (codec.name.toUpperCase()) {\n case 'RED':\n case 'ULPFEC':\n description.fecMechanisms.push(codec.name.toUpperCase());\n break;\n default: // only RED and ULPFEC are recognized as FEC mechanisms.\n break;\n }\n }\n }\n SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) {\n description.headerExtensions.push(SDPUtils.parseExtmap(line));\n });\n // FIXME: parse rtcp.\n return description;\n};\n\n// Generates parts of the SDP media section describing the capabilities /\n// parameters.\nSDPUtils.writeRtpDescription = function(kind, caps) {\n var sdp = '';\n\n // Build the mline.\n sdp += 'm=' + kind + ' ';\n sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs.\n sdp += ' UDP/TLS/RTP/SAVPF ';\n sdp += caps.codecs.map(function(codec) {\n if (codec.preferredPayloadType !== undefined) {\n return codec.preferredPayloadType;\n }\n return codec.payloadType;\n }).join(' ') + '\\r\\n';\n\n sdp += 'c=IN IP4 0.0.0.0\\r\\n';\n sdp += 'a=rtcp:9 IN IP4 0.0.0.0\\r\\n';\n\n // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.\n caps.codecs.forEach(function(codec) {\n sdp += SDPUtils.writeRtpMap(codec);\n sdp += SDPUtils.writeFmtp(codec);\n sdp += SDPUtils.writeRtcpFb(codec);\n });\n var maxptime = 0;\n caps.codecs.forEach(function(codec) {\n if (codec.maxptime > maxptime) {\n maxptime = codec.maxptime;\n }\n });\n if (maxptime > 0) {\n sdp += 'a=maxptime:' + maxptime + '\\r\\n';\n }\n sdp += 'a=rtcp-mux\\r\\n';\n\n if (caps.headerExtensions) {\n caps.headerExtensions.forEach(function(extension) {\n sdp += SDPUtils.writeExtmap(extension);\n });\n }\n // FIXME: write fecMechanisms.\n return sdp;\n};\n\n// Parses the SDP media section and returns an array of\n// RTCRtpEncodingParameters.\nSDPUtils.parseRtpEncodingParameters = function(mediaSection) {\n var encodingParameters = [];\n var description = SDPUtils.parseRtpParameters(mediaSection);\n var hasRed = description.fecMechanisms.indexOf('RED') !== -1;\n var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1;\n\n // filter a=ssrc:... cname:, ignore PlanB-msid\n var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(parts) {\n return parts.attribute === 'cname';\n });\n var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;\n var secondarySsrc;\n\n var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID')\n .map(function(line) {\n var parts = line.substr(17).split(' ');\n return parts.map(function(part) {\n return parseInt(part, 10);\n });\n });\n if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {\n secondarySsrc = flows[0][1];\n }\n\n description.codecs.forEach(function(codec) {\n if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) {\n var encParam = {\n ssrc: primarySsrc,\n codecPayloadType: parseInt(codec.parameters.apt, 10)\n };\n if (primarySsrc && secondarySsrc) {\n encParam.rtx = {ssrc: secondarySsrc};\n }\n encodingParameters.push(encParam);\n if (hasRed) {\n encParam = JSON.parse(JSON.stringify(encParam));\n encParam.fec = {\n ssrc: primarySsrc,\n mechanism: hasUlpfec ? 'red+ulpfec' : 'red'\n };\n encodingParameters.push(encParam);\n }\n }\n });\n if (encodingParameters.length === 0 && primarySsrc) {\n encodingParameters.push({\n ssrc: primarySsrc\n });\n }\n\n // we support both b=AS and b=TIAS but interpret AS as TIAS.\n var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b=');\n if (bandwidth.length) {\n if (bandwidth[0].indexOf('b=TIAS:') === 0) {\n bandwidth = parseInt(bandwidth[0].substr(7), 10);\n } else if (bandwidth[0].indexOf('b=AS:') === 0) {\n // use formula from JSEP to convert b=AS to TIAS value.\n bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95\n - (50 * 40 * 8);\n } else {\n bandwidth = undefined;\n }\n encodingParameters.forEach(function(params) {\n params.maxBitrate = bandwidth;\n });\n }\n return encodingParameters;\n};\n\n// parses http://draft.ortc.org/#rtcrtcpparameters*\nSDPUtils.parseRtcpParameters = function(mediaSection) {\n var rtcpParameters = {};\n\n // Gets the first SSRC. Note tha with RTX there might be multiple\n // SSRCs.\n var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(obj) {\n return obj.attribute === 'cname';\n })[0];\n if (remoteSsrc) {\n rtcpParameters.cname = remoteSsrc.value;\n rtcpParameters.ssrc = remoteSsrc.ssrc;\n }\n\n // Edge uses the compound attribute instead of reducedSize\n // compound is !reducedSize\n var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize');\n rtcpParameters.reducedSize = rsize.length > 0;\n rtcpParameters.compound = rsize.length === 0;\n\n // parses the rtcp-mux attrіbute.\n // Note that Edge does not support unmuxed RTCP.\n var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux');\n rtcpParameters.mux = mux.length > 0;\n\n return rtcpParameters;\n};\n\n// parses either a=msid: or a=ssrc:... msid lines and returns\n// the id of the MediaStream and MediaStreamTrack.\nSDPUtils.parseMsid = function(mediaSection) {\n var parts;\n var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:');\n if (spec.length === 1) {\n parts = spec[0].substr(7).split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(msidParts) {\n return msidParts.attribute === 'msid';\n });\n if (planB.length > 0) {\n parts = planB[0].value.split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n};\n\n// SCTP\n// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back\n// to draft-ietf-mmusic-sctp-sdp-05\nSDPUtils.parseSctpDescription = function(mediaSection) {\n var mline = SDPUtils.parseMLine(mediaSection);\n var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:');\n var maxMessageSize;\n if (maxSizeLine.length > 0) {\n maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);\n }\n if (isNaN(maxMessageSize)) {\n maxMessageSize = 65536;\n }\n var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:');\n if (sctpPort.length > 0) {\n return {\n port: parseInt(sctpPort[0].substr(12), 10),\n protocol: mline.fmt,\n maxMessageSize: maxMessageSize\n };\n }\n var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:');\n if (sctpMapLines.length > 0) {\n var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0]\n .substr(10)\n .split(' ');\n return {\n port: parseInt(parts[0], 10),\n protocol: parts[1],\n maxMessageSize: maxMessageSize\n };\n }\n};\n\n// SCTP\n// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers\n// support by now receiving in this format, unless we originally parsed\n// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line\n// protocol of DTLS/SCTP -- without UDP/ or TCP/)\nSDPUtils.writeSctpDescription = function(media, sctp) {\n var output = [];\n if (media.protocol !== 'DTLS/SCTP') {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctp-port:' + sctp.port + '\\r\\n'\n ];\n } else {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\\r\\n'\n ];\n }\n if (sctp.maxMessageSize !== undefined) {\n output.push('a=max-message-size:' + sctp.maxMessageSize + '\\r\\n');\n }\n return output.join('');\n};\n\n// Generate a session ID for SDP.\n// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1\n// recommends using a cryptographically random +ve 64-bit value\n// but right now this should be acceptable and within the right range\nSDPUtils.generateSessionId = function() {\n return Math.random().toString().substr(2, 21);\n};\n\n// Write boilder plate for start of SDP\n// sessId argument is optional - if not supplied it will\n// be generated randomly\n// sessVersion is optional and defaults to 2\n// sessUser is optional and defaults to 'thisisadapterortc'\nSDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {\n var sessionId;\n var version = sessVer !== undefined ? sessVer : 2;\n if (sessId) {\n sessionId = sessId;\n } else {\n sessionId = SDPUtils.generateSessionId();\n }\n var user = sessUser || 'thisisadapterortc';\n // FIXME: sess-id should be an NTP timestamp.\n return 'v=0\\r\\n' +\n 'o=' + user + ' ' + sessionId + ' ' + version +\n ' IN IP4 127.0.0.1\\r\\n' +\n 's=-\\r\\n' +\n 't=0 0\\r\\n';\n};\n\nSDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.direction) {\n sdp += 'a=' + transceiver.direction + '\\r\\n';\n } else if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n // spec.\n var msid = 'msid:' + stream.id + ' ' +\n transceiver.rtpSender.track.id + '\\r\\n';\n sdp += 'a=' + msid;\n\n // for Chrome.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n};\n\n// Gets the direction from the mediaSection or the sessionpart.\nSDPUtils.getDirection = function(mediaSection, sessionpart) {\n // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.\n var lines = SDPUtils.splitLines(mediaSection);\n for (var i = 0; i < lines.length; i++) {\n switch (lines[i]) {\n case 'a=sendrecv':\n case 'a=sendonly':\n case 'a=recvonly':\n case 'a=inactive':\n return lines[i].substr(2);\n default:\n // FIXME: What should happen here?\n }\n }\n if (sessionpart) {\n return SDPUtils.getDirection(sessionpart);\n }\n return 'sendrecv';\n};\n\nSDPUtils.getKind = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n return mline[0].substr(2);\n};\n\nSDPUtils.isRejected = function(mediaSection) {\n return mediaSection.split(' ', 2)[1] === '0';\n};\n\nSDPUtils.parseMLine = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var parts = lines[0].substr(2).split(' ');\n return {\n kind: parts[0],\n port: parseInt(parts[1], 10),\n protocol: parts[2],\n fmt: parts.slice(3).join(' ')\n };\n};\n\nSDPUtils.parseOLine = function(mediaSection) {\n var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0];\n var parts = line.substr(2).split(' ');\n return {\n username: parts[0],\n sessionId: parts[1],\n sessionVersion: parseInt(parts[2], 10),\n netType: parts[3],\n addressType: parts[4],\n address: parts[5]\n };\n};\n\n// a very naive interpretation of a valid SDP.\nSDPUtils.isValidSDP = function(blob) {\n if (typeof blob !== 'string' || blob.length === 0) {\n return false;\n }\n var lines = SDPUtils.splitLines(blob);\n for (var i = 0; i < lines.length; i++) {\n if (lines[i].length < 2 || lines[i].charAt(1) !== '=') {\n return false;\n }\n // TODO: check the modifier a bit more.\n }\n return true;\n};\n\n// Expose public methods.\nif (typeof module === 'object') {\n module.exports = SDPUtils;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nvar SDPUtils = require('sdp');\n\nfunction fixStatsType(stat) {\n return {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[stat.type] || stat.type;\n}\n\nfunction writeMediaSection(transceiver, caps, type, stream, dtlsRole) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : dtlsRole || 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n var trackId = transceiver.rtpSender._initialTrackId ||\n transceiver.rtpSender.track.id;\n transceiver.rtpSender._initialTrackId = trackId;\n // spec.\n var msid = 'msid:' + (stream ? stream.id : '-') + ' ' +\n trackId + '\\r\\n';\n sdp += 'a=' + msid;\n // for Chrome. Legacy should no longer be required.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n\n // RTX\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n}\n\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nfunction filterIceServers(iceServers, edgeVersion) {\n var hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(function(server) {\n if (server && (server.urls || server.url)) {\n var urls = server.urls || server.url;\n if (server.url && !server.urls) {\n console.warn('RTCIceServer.url is deprecated! Use urls instead.');\n }\n var isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(function(url) {\n var validTurn = url.indexOf('turn:') === 0 &&\n url.indexOf('transport=udp') !== -1 &&\n url.indexOf('turn:[') === -1 &&\n !hasTurn;\n\n if (validTurn) {\n hasTurn = true;\n return true;\n }\n return url.indexOf('stun:') === 0 && edgeVersion >= 14393 &&\n url.indexOf('?transport=udp') === -1;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n\n// Determines the intersection of local and remote capabilities.\nfunction getCommonCapabilities(localCapabilities, remoteCapabilities) {\n var commonCapabilities = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: []\n };\n\n var findCodecByPayloadType = function(pt, codecs) {\n pt = parseInt(pt, 10);\n for (var i = 0; i < codecs.length; i++) {\n if (codecs[i].payloadType === pt ||\n codecs[i].preferredPayloadType === pt) {\n return codecs[i];\n }\n }\n };\n\n var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) {\n var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs);\n var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs);\n return lCodec && rCodec &&\n lCodec.name.toLowerCase() === rCodec.name.toLowerCase();\n };\n\n localCapabilities.codecs.forEach(function(lCodec) {\n for (var i = 0; i < remoteCapabilities.codecs.length; i++) {\n var rCodec = remoteCapabilities.codecs[i];\n if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() &&\n lCodec.clockRate === rCodec.clockRate) {\n if (lCodec.name.toLowerCase() === 'rtx' &&\n lCodec.parameters && rCodec.parameters.apt) {\n // for RTX we need to find the local rtx that has a apt\n // which points to the same local codec as the remote one.\n if (!rtxCapabilityMatches(lCodec, rCodec,\n localCapabilities.codecs, remoteCapabilities.codecs)) {\n continue;\n }\n }\n rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy\n // number of channels is the highest common number of channels\n rCodec.numChannels = Math.min(lCodec.numChannels,\n rCodec.numChannels);\n // push rCodec so we reply with offerer payload type\n commonCapabilities.codecs.push(rCodec);\n\n // determine common feedback mechanisms\n rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) {\n for (var j = 0; j < lCodec.rtcpFeedback.length; j++) {\n if (lCodec.rtcpFeedback[j].type === fb.type &&\n lCodec.rtcpFeedback[j].parameter === fb.parameter) {\n return true;\n }\n }\n return false;\n });\n // FIXME: also need to determine .parameters\n // see https://github.com/openpeer/ortc/issues/569\n break;\n }\n }\n });\n\n localCapabilities.headerExtensions.forEach(function(lHeaderExtension) {\n for (var i = 0; i < remoteCapabilities.headerExtensions.length;\n i++) {\n var rHeaderExtension = remoteCapabilities.headerExtensions[i];\n if (lHeaderExtension.uri === rHeaderExtension.uri) {\n commonCapabilities.headerExtensions.push(rHeaderExtension);\n break;\n }\n }\n });\n\n // FIXME: fecMechanisms\n return commonCapabilities;\n}\n\n// is action=setLocalDescription with type allowed in signalingState\nfunction isActionAllowedInSignalingState(action, type, signalingState) {\n return {\n offer: {\n setLocalDescription: ['stable', 'have-local-offer'],\n setRemoteDescription: ['stable', 'have-remote-offer']\n },\n answer: {\n setLocalDescription: ['have-remote-offer', 'have-local-pranswer'],\n setRemoteDescription: ['have-local-offer', 'have-remote-pranswer']\n }\n }[type][action].indexOf(signalingState) !== -1;\n}\n\nfunction maybeAddCandidate(iceTransport, candidate) {\n // Edge's internal representation adds some fields therefore\n // not all fieldѕ are taken into account.\n var alreadyAdded = iceTransport.getRemoteCandidates()\n .find(function(remoteCandidate) {\n return candidate.foundation === remoteCandidate.foundation &&\n candidate.ip === remoteCandidate.ip &&\n candidate.port === remoteCandidate.port &&\n candidate.priority === remoteCandidate.priority &&\n candidate.protocol === remoteCandidate.protocol &&\n candidate.type === remoteCandidate.type;\n });\n if (!alreadyAdded) {\n iceTransport.addRemoteCandidate(candidate);\n }\n return !alreadyAdded;\n}\n\n\nfunction makeError(name, description) {\n var e = new Error(description);\n e.name = name;\n // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names\n e.code = {\n NotSupportedError: 9,\n InvalidStateError: 11,\n InvalidAccessError: 15,\n TypeError: undefined,\n OperationError: undefined\n }[name];\n return e;\n}\n\nmodule.exports = function(window, edgeVersion) {\n // https://w3c.github.io/mediacapture-main/#mediastream\n // Helper function to add the track to the stream and\n // dispatch the event ourselves.\n function addTrackToStreamAndFireEvent(track, stream) {\n stream.addTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack',\n {track: track}));\n }\n\n function removeTrackFromStreamAndFireEvent(track, stream) {\n stream.removeTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack',\n {track: track}));\n }\n\n function fireAddTrack(pc, track, receiver, streams) {\n var trackEvent = new Event('track');\n trackEvent.track = track;\n trackEvent.receiver = receiver;\n trackEvent.transceiver = {receiver: receiver};\n trackEvent.streams = streams;\n window.setTimeout(function() {\n pc._dispatchEvent('track', trackEvent);\n });\n }\n\n var RTCPeerConnection = function(config) {\n var pc = this;\n\n var _eventTarget = document.createDocumentFragment();\n ['addEventListener', 'removeEventListener', 'dispatchEvent']\n .forEach(function(method) {\n pc[method] = _eventTarget[method].bind(_eventTarget);\n });\n\n this.canTrickleIceCandidates = null;\n\n this.needNegotiation = false;\n\n this.localStreams = [];\n this.remoteStreams = [];\n\n this._localDescription = null;\n this._remoteDescription = null;\n\n this.signalingState = 'stable';\n this.iceConnectionState = 'new';\n this.connectionState = 'new';\n this.iceGatheringState = 'new';\n\n config = JSON.parse(JSON.stringify(config || {}));\n\n this.usingBundle = config.bundlePolicy === 'max-bundle';\n if (config.rtcpMuxPolicy === 'negotiate') {\n throw(makeError('NotSupportedError',\n 'rtcpMuxPolicy \\'negotiate\\' is not supported'));\n } else if (!config.rtcpMuxPolicy) {\n config.rtcpMuxPolicy = 'require';\n }\n\n switch (config.iceTransportPolicy) {\n case 'all':\n case 'relay':\n break;\n default:\n config.iceTransportPolicy = 'all';\n break;\n }\n\n switch (config.bundlePolicy) {\n case 'balanced':\n case 'max-compat':\n case 'max-bundle':\n break;\n default:\n config.bundlePolicy = 'balanced';\n break;\n }\n\n config.iceServers = filterIceServers(config.iceServers || [], edgeVersion);\n\n this._iceGatherers = [];\n if (config.iceCandidatePoolSize) {\n for (var i = config.iceCandidatePoolSize; i > 0; i--) {\n this._iceGatherers.push(new window.RTCIceGatherer({\n iceServers: config.iceServers,\n gatherPolicy: config.iceTransportPolicy\n }));\n }\n } else {\n config.iceCandidatePoolSize = 0;\n }\n\n this._config = config;\n\n // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ...\n // everything that is needed to describe a SDP m-line.\n this.transceivers = [];\n\n this._sdpSessionId = SDPUtils.generateSessionId();\n this._sdpSessionVersion = 0;\n\n this._dtlsRole = undefined; // role for a=setup to use in answers.\n\n this._isClosed = false;\n };\n\n Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', {\n configurable: true,\n get: function() {\n return this._localDescription;\n }\n });\n Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', {\n configurable: true,\n get: function() {\n return this._remoteDescription;\n }\n });\n\n // set up event handlers on prototype\n RTCPeerConnection.prototype.onicecandidate = null;\n RTCPeerConnection.prototype.onaddstream = null;\n RTCPeerConnection.prototype.ontrack = null;\n RTCPeerConnection.prototype.onremovestream = null;\n RTCPeerConnection.prototype.onsignalingstatechange = null;\n RTCPeerConnection.prototype.oniceconnectionstatechange = null;\n RTCPeerConnection.prototype.onconnectionstatechange = null;\n RTCPeerConnection.prototype.onicegatheringstatechange = null;\n RTCPeerConnection.prototype.onnegotiationneeded = null;\n RTCPeerConnection.prototype.ondatachannel = null;\n\n RTCPeerConnection.prototype._dispatchEvent = function(name, event) {\n if (this._isClosed) {\n return;\n }\n this.dispatchEvent(event);\n if (typeof this['on' + name] === 'function') {\n this['on' + name](event);\n }\n };\n\n RTCPeerConnection.prototype._emitGatheringStateChange = function() {\n var event = new Event('icegatheringstatechange');\n this._dispatchEvent('icegatheringstatechange', event);\n };\n\n RTCPeerConnection.prototype.getConfiguration = function() {\n return this._config;\n };\n\n RTCPeerConnection.prototype.getLocalStreams = function() {\n return this.localStreams;\n };\n\n RTCPeerConnection.prototype.getRemoteStreams = function() {\n return this.remoteStreams;\n };\n\n // internal helper to create a transceiver object.\n // (which is not yet the same as the WebRTC 1.0 transceiver)\n RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) {\n var hasBundleTransport = this.transceivers.length > 0;\n var transceiver = {\n track: null,\n iceGatherer: null,\n iceTransport: null,\n dtlsTransport: null,\n localCapabilities: null,\n remoteCapabilities: null,\n rtpSender: null,\n rtpReceiver: null,\n kind: kind,\n mid: null,\n sendEncodingParameters: null,\n recvEncodingParameters: null,\n stream: null,\n associatedRemoteMediaStreams: [],\n wantReceive: true\n };\n if (this.usingBundle && hasBundleTransport) {\n transceiver.iceTransport = this.transceivers[0].iceTransport;\n transceiver.dtlsTransport = this.transceivers[0].dtlsTransport;\n } else {\n var transports = this._createIceAndDtlsTransports();\n transceiver.iceTransport = transports.iceTransport;\n transceiver.dtlsTransport = transports.dtlsTransport;\n }\n if (!doNotAdd) {\n this.transceivers.push(transceiver);\n }\n return transceiver;\n };\n\n RTCPeerConnection.prototype.addTrack = function(track, stream) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call addTrack on a closed peerconnection.');\n }\n\n var alreadyExists = this.transceivers.find(function(s) {\n return s.track === track;\n });\n\n if (alreadyExists) {\n throw makeError('InvalidAccessError', 'Track already exists.');\n }\n\n var transceiver;\n for (var i = 0; i < this.transceivers.length; i++) {\n if (!this.transceivers[i].track &&\n this.transceivers[i].kind === track.kind) {\n transceiver = this.transceivers[i];\n }\n }\n if (!transceiver) {\n transceiver = this._createTransceiver(track.kind);\n }\n\n this._maybeFireNegotiationNeeded();\n\n if (this.localStreams.indexOf(stream) === -1) {\n this.localStreams.push(stream);\n }\n\n transceiver.track = track;\n transceiver.stream = stream;\n transceiver.rtpSender = new window.RTCRtpSender(track,\n transceiver.dtlsTransport);\n return transceiver.rtpSender;\n };\n\n RTCPeerConnection.prototype.addStream = function(stream) {\n var pc = this;\n if (edgeVersion >= 15025) {\n stream.getTracks().forEach(function(track) {\n pc.addTrack(track, stream);\n });\n } else {\n // Clone is necessary for local demos mostly, attaching directly\n // to two different senders does not work (build 10547).\n // Fixed in 15025 (or earlier)\n var clonedStream = stream.clone();\n stream.getTracks().forEach(function(track, idx) {\n var clonedTrack = clonedStream.getTracks()[idx];\n track.addEventListener('enabled', function(event) {\n clonedTrack.enabled = event.enabled;\n });\n });\n clonedStream.getTracks().forEach(function(track) {\n pc.addTrack(track, clonedStream);\n });\n }\n };\n\n RTCPeerConnection.prototype.removeTrack = function(sender) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call removeTrack on a closed peerconnection.');\n }\n\n if (!(sender instanceof window.RTCRtpSender)) {\n throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.');\n }\n\n var transceiver = this.transceivers.find(function(t) {\n return t.rtpSender === sender;\n });\n\n if (!transceiver) {\n throw makeError('InvalidAccessError',\n 'Sender was not created by this connection.');\n }\n var stream = transceiver.stream;\n\n transceiver.rtpSender.stop();\n transceiver.rtpSender = null;\n transceiver.track = null;\n transceiver.stream = null;\n\n // remove the stream from the set of local streams\n var localStreams = this.transceivers.map(function(t) {\n return t.stream;\n });\n if (localStreams.indexOf(stream) === -1 &&\n this.localStreams.indexOf(stream) > -1) {\n this.localStreams.splice(this.localStreams.indexOf(stream), 1);\n }\n\n this._maybeFireNegotiationNeeded();\n };\n\n RTCPeerConnection.prototype.removeStream = function(stream) {\n var pc = this;\n stream.getTracks().forEach(function(track) {\n var sender = pc.getSenders().find(function(s) {\n return s.track === track;\n });\n if (sender) {\n pc.removeTrack(sender);\n }\n });\n };\n\n RTCPeerConnection.prototype.getSenders = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpSender;\n })\n .map(function(transceiver) {\n return transceiver.rtpSender;\n });\n };\n\n RTCPeerConnection.prototype.getReceivers = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpReceiver;\n })\n .map(function(transceiver) {\n return transceiver.rtpReceiver;\n });\n };\n\n\n RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex,\n usingBundle) {\n var pc = this;\n if (usingBundle && sdpMLineIndex > 0) {\n return this.transceivers[0].iceGatherer;\n } else if (this._iceGatherers.length) {\n return this._iceGatherers.shift();\n }\n var iceGatherer = new window.RTCIceGatherer({\n iceServers: this._config.iceServers,\n gatherPolicy: this._config.iceTransportPolicy\n });\n Object.defineProperty(iceGatherer, 'state',\n {value: 'new', writable: true}\n );\n\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = [];\n this.transceivers[sdpMLineIndex].bufferCandidates = function(event) {\n var end = !event.candidate || Object.keys(event.candidate).length === 0;\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n iceGatherer.state = end ? 'completed' : 'gathering';\n if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) {\n pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event);\n }\n };\n iceGatherer.addEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n return iceGatherer;\n };\n\n // start gathering from an RTCIceGatherer.\n RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) {\n var pc = this;\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer.onlocalcandidate) {\n return;\n }\n var bufferedCandidateEvents =\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents;\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null;\n iceGatherer.removeEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n iceGatherer.onlocalcandidate = function(evt) {\n if (pc.usingBundle && sdpMLineIndex > 0) {\n // if we know that we use bundle we can drop candidates with\n // ѕdpMLineIndex > 0. If we don't do this then our state gets\n // confused since we dispose the extra ice gatherer.\n return;\n }\n var event = new Event('icecandidate');\n event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex};\n\n var cand = evt.candidate;\n // Edge emits an empty object for RTCIceCandidateComplete‥\n var end = !cand || Object.keys(cand).length === 0;\n if (end) {\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') {\n iceGatherer.state = 'completed';\n }\n } else {\n if (iceGatherer.state === 'new') {\n iceGatherer.state = 'gathering';\n }\n // RTCIceCandidate doesn't have a component, needs to be added\n cand.component = 1;\n // also the usernameFragment. TODO: update SDP to take both variants.\n cand.ufrag = iceGatherer.getLocalParameters().usernameFragment;\n\n var serializedCandidate = SDPUtils.writeCandidate(cand);\n event.candidate = Object.assign(event.candidate,\n SDPUtils.parseCandidate(serializedCandidate));\n\n event.candidate.candidate = serializedCandidate;\n event.candidate.toJSON = function() {\n return {\n candidate: event.candidate.candidate,\n sdpMid: event.candidate.sdpMid,\n sdpMLineIndex: event.candidate.sdpMLineIndex,\n usernameFragment: event.candidate.usernameFragment\n };\n };\n }\n\n // update local description.\n var sections = SDPUtils.getMediaSections(pc._localDescription.sdp);\n if (!end) {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=' + event.candidate.candidate + '\\r\\n';\n } else {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=end-of-candidates\\r\\n';\n }\n pc._localDescription.sdp =\n SDPUtils.getDescription(pc._localDescription.sdp) +\n sections.join('');\n var complete = pc.transceivers.every(function(transceiver) {\n return transceiver.iceGatherer &&\n transceiver.iceGatherer.state === 'completed';\n });\n\n if (pc.iceGatheringState !== 'gathering') {\n pc.iceGatheringState = 'gathering';\n pc._emitGatheringStateChange();\n }\n\n // Emit candidate. Also emit null candidate when all gatherers are\n // complete.\n if (!end) {\n pc._dispatchEvent('icecandidate', event);\n }\n if (complete) {\n pc._dispatchEvent('icecandidate', new Event('icecandidate'));\n pc.iceGatheringState = 'complete';\n pc._emitGatheringStateChange();\n }\n };\n\n // emit already gathered candidates.\n window.setTimeout(function() {\n bufferedCandidateEvents.forEach(function(e) {\n iceGatherer.onlocalcandidate(e);\n });\n }, 0);\n };\n\n // Create ICE transport and DTLS transport.\n RTCPeerConnection.prototype._createIceAndDtlsTransports = function() {\n var pc = this;\n var iceTransport = new window.RTCIceTransport(null);\n iceTransport.onicestatechange = function() {\n pc._updateIceConnectionState();\n pc._updateConnectionState();\n };\n\n var dtlsTransport = new window.RTCDtlsTransport(iceTransport);\n dtlsTransport.ondtlsstatechange = function() {\n pc._updateConnectionState();\n };\n dtlsTransport.onerror = function() {\n // onerror does not set state to failed by itself.\n Object.defineProperty(dtlsTransport, 'state',\n {value: 'failed', writable: true});\n pc._updateConnectionState();\n };\n\n return {\n iceTransport: iceTransport,\n dtlsTransport: dtlsTransport\n };\n };\n\n // Destroy ICE gatherer, ICE transport and DTLS transport.\n // Without triggering the callbacks.\n RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function(\n sdpMLineIndex) {\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer) {\n delete iceGatherer.onlocalcandidate;\n delete this.transceivers[sdpMLineIndex].iceGatherer;\n }\n var iceTransport = this.transceivers[sdpMLineIndex].iceTransport;\n if (iceTransport) {\n delete iceTransport.onicestatechange;\n delete this.transceivers[sdpMLineIndex].iceTransport;\n }\n var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport;\n if (dtlsTransport) {\n delete dtlsTransport.ondtlsstatechange;\n delete dtlsTransport.onerror;\n delete this.transceivers[sdpMLineIndex].dtlsTransport;\n }\n };\n\n // Start the RTP Sender and Receiver for a transceiver.\n RTCPeerConnection.prototype._transceive = function(transceiver,\n send, recv) {\n var params = getCommonCapabilities(transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n if (send && transceiver.rtpSender) {\n params.encodings = transceiver.sendEncodingParameters;\n params.rtcp = {\n cname: SDPUtils.localCName,\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.recvEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc;\n }\n transceiver.rtpSender.send(params);\n }\n if (recv && transceiver.rtpReceiver && params.codecs.length > 0) {\n // remove RTX field in Edge 14942\n if (transceiver.kind === 'video'\n && transceiver.recvEncodingParameters\n && edgeVersion < 15019) {\n transceiver.recvEncodingParameters.forEach(function(p) {\n delete p.rtx;\n });\n }\n if (transceiver.recvEncodingParameters.length) {\n params.encodings = transceiver.recvEncodingParameters;\n } else {\n params.encodings = [{}];\n }\n params.rtcp = {\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.rtcpParameters.cname) {\n params.rtcp.cname = transceiver.rtcpParameters.cname;\n }\n if (transceiver.sendEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc;\n }\n transceiver.rtpReceiver.receive(params);\n }\n };\n\n RTCPeerConnection.prototype.setLocalDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setLocalDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set local ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var sections;\n var sessionpart;\n if (description.type === 'offer') {\n // VERY limited support for SDP munging. Limited to:\n // * changing the order of codecs\n sections = SDPUtils.splitSections(description.sdp);\n sessionpart = sections.shift();\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var caps = SDPUtils.parseRtpParameters(mediaSection);\n pc.transceivers[sdpMLineIndex].localCapabilities = caps;\n });\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n });\n } else if (description.type === 'answer') {\n sections = SDPUtils.splitSections(pc._remoteDescription.sdp);\n sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var transceiver = pc.transceivers[sdpMLineIndex];\n var iceGatherer = transceiver.iceGatherer;\n var iceTransport = transceiver.iceTransport;\n var dtlsTransport = transceiver.dtlsTransport;\n var localCapabilities = transceiver.localCapabilities;\n var remoteCapabilities = transceiver.remoteCapabilities;\n\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n\n if (!rejected && !transceiver.rejected) {\n var remoteIceParameters = SDPUtils.getIceParameters(\n mediaSection, sessionpart);\n var remoteDtlsParameters = SDPUtils.getDtlsParameters(\n mediaSection, sessionpart);\n if (isIceLite) {\n remoteDtlsParameters.role = 'server';\n }\n\n if (!pc.usingBundle || sdpMLineIndex === 0) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n isIceLite ? 'controlling' : 'controlled');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // Calculate intersection of capabilities.\n var params = getCommonCapabilities(localCapabilities,\n remoteCapabilities);\n\n // Start the RTCRtpSender. The RTCRtpReceiver for this\n // transceiver has already been started in setRemoteDescription.\n pc._transceive(transceiver,\n params.codecs.length > 0,\n false);\n }\n });\n }\n\n pc._localDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-local-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.setRemoteDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setRemoteDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set remote ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var streams = {};\n pc.remoteStreams.forEach(function(stream) {\n streams[stream.id] = stream;\n });\n var receiverList = [];\n var sections = SDPUtils.splitSections(description.sdp);\n var sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n var usingBundle = SDPUtils.matchPrefix(sessionpart,\n 'a=group:BUNDLE ').length > 0;\n pc.usingBundle = usingBundle;\n var iceOptions = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-options:')[0];\n if (iceOptions) {\n pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ')\n .indexOf('trickle') >= 0;\n } else {\n pc.canTrickleIceCandidates = false;\n }\n\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var lines = SDPUtils.splitLines(mediaSection);\n var kind = SDPUtils.getKind(mediaSection);\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n var protocol = lines[0].substr(2).split(' ')[2];\n\n var direction = SDPUtils.getDirection(mediaSection, sessionpart);\n var remoteMsid = SDPUtils.parseMsid(mediaSection);\n\n var mid = SDPUtils.getMid(mediaSection) || SDPUtils.generateIdentifier();\n\n // Reject datachannels which are not implemented yet.\n if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' ||\n protocol === 'UDP/DTLS/SCTP'))) {\n // TODO: this is dangerous in the case where a non-rejected m-line\n // becomes rejected.\n pc.transceivers[sdpMLineIndex] = {\n mid: mid,\n kind: kind,\n protocol: protocol,\n rejected: true\n };\n return;\n }\n\n if (!rejected && pc.transceivers[sdpMLineIndex] &&\n pc.transceivers[sdpMLineIndex].rejected) {\n // recycle a rejected transceiver.\n pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true);\n }\n\n var transceiver;\n var iceGatherer;\n var iceTransport;\n var dtlsTransport;\n var rtpReceiver;\n var sendEncodingParameters;\n var recvEncodingParameters;\n var localCapabilities;\n\n var track;\n // FIXME: ensure the mediaSection has rtcp-mux set.\n var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection);\n var remoteIceParameters;\n var remoteDtlsParameters;\n if (!rejected) {\n remoteIceParameters = SDPUtils.getIceParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters.role = 'client';\n }\n recvEncodingParameters =\n SDPUtils.parseRtpEncodingParameters(mediaSection);\n\n var rtcpParameters = SDPUtils.parseRtcpParameters(mediaSection);\n\n var isComplete = SDPUtils.matchPrefix(mediaSection,\n 'a=end-of-candidates', sessionpart).length > 0;\n var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:')\n .map(function(cand) {\n return SDPUtils.parseCandidate(cand);\n })\n .filter(function(cand) {\n return cand.component === 1;\n });\n\n // Check if we can use BUNDLE and dispose transports.\n if ((description.type === 'offer' || description.type === 'answer') &&\n !rejected && usingBundle && sdpMLineIndex > 0 &&\n pc.transceivers[sdpMLineIndex]) {\n pc._disposeIceAndDtlsTransports(sdpMLineIndex);\n pc.transceivers[sdpMLineIndex].iceGatherer =\n pc.transceivers[0].iceGatherer;\n pc.transceivers[sdpMLineIndex].iceTransport =\n pc.transceivers[0].iceTransport;\n pc.transceivers[sdpMLineIndex].dtlsTransport =\n pc.transceivers[0].dtlsTransport;\n if (pc.transceivers[sdpMLineIndex].rtpSender) {\n pc.transceivers[sdpMLineIndex].rtpSender.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n if (pc.transceivers[sdpMLineIndex].rtpReceiver) {\n pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n }\n if (description.type === 'offer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex] ||\n pc._createTransceiver(kind);\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n usingBundle);\n }\n\n if (cands.length && transceiver.iceTransport.state === 'new') {\n if (isComplete && (!usingBundle || sdpMLineIndex === 0)) {\n transceiver.iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n localCapabilities = window.RTCRtpReceiver.getCapabilities(kind);\n\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n\n sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 2) * 1001\n }];\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n var isNewTrack = false;\n if (direction === 'sendrecv' || direction === 'sendonly') {\n isNewTrack = !transceiver.rtpReceiver;\n rtpReceiver = transceiver.rtpReceiver ||\n new window.RTCRtpReceiver(transceiver.dtlsTransport, kind);\n\n if (isNewTrack) {\n var stream;\n track = rtpReceiver.track;\n // FIXME: does not work with Plan B.\n if (remoteMsid && remoteMsid.stream === '-') {\n // no-op. a stream id of '-' means: no associated stream.\n } else if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n Object.defineProperty(streams[remoteMsid.stream], 'id', {\n get: function() {\n return remoteMsid.stream;\n }\n });\n }\n Object.defineProperty(track, 'id', {\n get: function() {\n return remoteMsid.track;\n }\n });\n stream = streams[remoteMsid.stream];\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n stream = streams.default;\n }\n if (stream) {\n addTrackToStreamAndFireEvent(track, stream);\n transceiver.associatedRemoteMediaStreams.push(stream);\n }\n receiverList.push([track, rtpReceiver, stream]);\n }\n } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) {\n transceiver.associatedRemoteMediaStreams.forEach(function(s) {\n var nativeTrack = s.getTracks().find(function(t) {\n return t.id === transceiver.rtpReceiver.track.id;\n });\n if (nativeTrack) {\n removeTrackFromStreamAndFireEvent(nativeTrack, s);\n }\n });\n transceiver.associatedRemoteMediaStreams = [];\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.remoteCapabilities = remoteCapabilities;\n transceiver.rtpReceiver = rtpReceiver;\n transceiver.rtcpParameters = rtcpParameters;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n transceiver.recvEncodingParameters = recvEncodingParameters;\n\n // Start the RTCRtpReceiver now. The RTPSender is started in\n // setLocalDescription.\n pc._transceive(pc.transceivers[sdpMLineIndex],\n false,\n isNewTrack);\n } else if (description.type === 'answer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex];\n iceGatherer = transceiver.iceGatherer;\n iceTransport = transceiver.iceTransport;\n dtlsTransport = transceiver.dtlsTransport;\n rtpReceiver = transceiver.rtpReceiver;\n sendEncodingParameters = transceiver.sendEncodingParameters;\n localCapabilities = transceiver.localCapabilities;\n\n pc.transceivers[sdpMLineIndex].recvEncodingParameters =\n recvEncodingParameters;\n pc.transceivers[sdpMLineIndex].remoteCapabilities =\n remoteCapabilities;\n pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters;\n\n if (cands.length && iceTransport.state === 'new') {\n if ((isIceLite || isComplete) &&\n (!usingBundle || sdpMLineIndex === 0)) {\n iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n if (!usingBundle || sdpMLineIndex === 0) {\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n 'controlling');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // If the offer contained RTX but the answer did not,\n // remove RTX from sendEncodingParameters.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n pc._transceive(transceiver,\n direction === 'sendrecv' || direction === 'recvonly',\n direction === 'sendrecv' || direction === 'sendonly');\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n if (rtpReceiver &&\n (direction === 'sendrecv' || direction === 'sendonly')) {\n track = rtpReceiver.track;\n if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]);\n receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]);\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams.default);\n receiverList.push([track, rtpReceiver, streams.default]);\n }\n } else {\n // FIXME: actually the receiver should be created later.\n delete transceiver.rtpReceiver;\n }\n }\n });\n\n if (pc._dtlsRole === undefined) {\n pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive';\n }\n\n pc._remoteDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-remote-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n Object.keys(streams).forEach(function(sid) {\n var stream = streams[sid];\n if (stream.getTracks().length) {\n if (pc.remoteStreams.indexOf(stream) === -1) {\n pc.remoteStreams.push(stream);\n var event = new Event('addstream');\n event.stream = stream;\n window.setTimeout(function() {\n pc._dispatchEvent('addstream', event);\n });\n }\n\n receiverList.forEach(function(item) {\n var track = item[0];\n var receiver = item[1];\n if (stream.id !== item[2].id) {\n return;\n }\n fireAddTrack(pc, track, receiver, [stream]);\n });\n }\n });\n receiverList.forEach(function(item) {\n if (item[2]) {\n return;\n }\n fireAddTrack(pc, item[0], item[1], []);\n });\n\n // check whether addIceCandidate({}) was called within four seconds after\n // setRemoteDescription.\n window.setTimeout(function() {\n if (!(pc && pc.transceivers)) {\n return;\n }\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport &&\n transceiver.iceTransport.state === 'new' &&\n transceiver.iceTransport.getRemoteCandidates().length > 0) {\n console.warn('Timeout for addRemoteCandidate. Consider sending ' +\n 'an end-of-candidates notification');\n transceiver.iceTransport.addRemoteCandidate({});\n }\n });\n }, 4000);\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.close = function() {\n this.transceivers.forEach(function(transceiver) {\n /* not yet\n if (transceiver.iceGatherer) {\n transceiver.iceGatherer.close();\n }\n */\n if (transceiver.iceTransport) {\n transceiver.iceTransport.stop();\n }\n if (transceiver.dtlsTransport) {\n transceiver.dtlsTransport.stop();\n }\n if (transceiver.rtpSender) {\n transceiver.rtpSender.stop();\n }\n if (transceiver.rtpReceiver) {\n transceiver.rtpReceiver.stop();\n }\n });\n // FIXME: clean up tracks, local streams, remote streams, etc\n this._isClosed = true;\n this._updateSignalingState('closed');\n };\n\n // Update the signaling state.\n RTCPeerConnection.prototype._updateSignalingState = function(newState) {\n this.signalingState = newState;\n var event = new Event('signalingstatechange');\n this._dispatchEvent('signalingstatechange', event);\n };\n\n // Determine whether to fire the negotiationneeded event.\n RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() {\n var pc = this;\n if (this.signalingState !== 'stable' || this.needNegotiation === true) {\n return;\n }\n this.needNegotiation = true;\n window.setTimeout(function() {\n if (pc.needNegotiation) {\n pc.needNegotiation = false;\n var event = new Event('negotiationneeded');\n pc._dispatchEvent('negotiationneeded', event);\n }\n }, 0);\n };\n\n // Update the ice connection state.\n RTCPeerConnection.prototype._updateIceConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n checking: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n }\n });\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.checking > 0) {\n newState = 'checking';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n } else if (states.completed > 0) {\n newState = 'completed';\n }\n\n if (newState !== this.iceConnectionState) {\n this.iceConnectionState = newState;\n var event = new Event('iceconnectionstatechange');\n this._dispatchEvent('iceconnectionstatechange', event);\n }\n };\n\n // Update the connection state.\n RTCPeerConnection.prototype._updateConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n connecting: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && transceiver.dtlsTransport &&\n !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n states[transceiver.dtlsTransport.state]++;\n }\n });\n // ICETransport.completed and connected are the same for this purpose.\n states.connected += states.completed;\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.connecting > 0) {\n newState = 'connecting';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n }\n\n if (newState !== this.connectionState) {\n this.connectionState = newState;\n var event = new Event('connectionstatechange');\n this._dispatchEvent('connectionstatechange', event);\n }\n };\n\n RTCPeerConnection.prototype.createOffer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createOffer after close'));\n }\n\n var numAudioTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'audio';\n }).length;\n var numVideoTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'video';\n }).length;\n\n // Determine number of audio and video tracks we need to send/recv.\n var offerOptions = arguments[0];\n if (offerOptions) {\n // Reject Chrome legacy constraints.\n if (offerOptions.mandatory || offerOptions.optional) {\n throw new TypeError(\n 'Legacy mandatory/optional constraints not supported.');\n }\n if (offerOptions.offerToReceiveAudio !== undefined) {\n if (offerOptions.offerToReceiveAudio === true) {\n numAudioTracks = 1;\n } else if (offerOptions.offerToReceiveAudio === false) {\n numAudioTracks = 0;\n } else {\n numAudioTracks = offerOptions.offerToReceiveAudio;\n }\n }\n if (offerOptions.offerToReceiveVideo !== undefined) {\n if (offerOptions.offerToReceiveVideo === true) {\n numVideoTracks = 1;\n } else if (offerOptions.offerToReceiveVideo === false) {\n numVideoTracks = 0;\n } else {\n numVideoTracks = offerOptions.offerToReceiveVideo;\n }\n }\n }\n\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.kind === 'audio') {\n numAudioTracks--;\n if (numAudioTracks < 0) {\n transceiver.wantReceive = false;\n }\n } else if (transceiver.kind === 'video') {\n numVideoTracks--;\n if (numVideoTracks < 0) {\n transceiver.wantReceive = false;\n }\n }\n });\n\n // Create M-lines for recvonly streams.\n while (numAudioTracks > 0 || numVideoTracks > 0) {\n if (numAudioTracks > 0) {\n pc._createTransceiver('audio');\n numAudioTracks--;\n }\n if (numVideoTracks > 0) {\n pc._createTransceiver('video');\n numVideoTracks--;\n }\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n // For each track, create an ice gatherer, ice transport,\n // dtls transport, potentially rtpsender and rtpreceiver.\n var track = transceiver.track;\n var kind = transceiver.kind;\n var mid = transceiver.mid || SDPUtils.generateIdentifier();\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n pc.usingBundle);\n }\n\n var localCapabilities = window.RTCRtpSender.getCapabilities(kind);\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n localCapabilities.codecs.forEach(function(codec) {\n // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552\n // by adding level-asymmetry-allowed=1\n if (codec.name === 'H264' &&\n codec.parameters['level-asymmetry-allowed'] === undefined) {\n codec.parameters['level-asymmetry-allowed'] = '1';\n }\n\n // for subsequent offers, we might have to re-use the payload\n // type of the last offer.\n if (transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.codecs) {\n transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) {\n if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() &&\n codec.clockRate === remoteCodec.clockRate) {\n codec.preferredPayloadType = remoteCodec.payloadType;\n }\n });\n }\n });\n localCapabilities.headerExtensions.forEach(function(hdrExt) {\n var remoteExtensions = transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.headerExtensions || [];\n remoteExtensions.forEach(function(rHdrExt) {\n if (hdrExt.uri === rHdrExt.uri) {\n hdrExt.id = rHdrExt.id;\n }\n });\n });\n\n // generate an ssrc now, to be used later in rtpSender.send\n var sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 1) * 1001\n }];\n if (track) {\n // add RTX\n if (edgeVersion >= 15019 && kind === 'video' &&\n !sendEncodingParameters[0].rtx) {\n sendEncodingParameters[0].rtx = {\n ssrc: sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n\n if (transceiver.wantReceive) {\n transceiver.rtpReceiver = new window.RTCRtpReceiver(\n transceiver.dtlsTransport, kind);\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n });\n\n // always offer BUNDLE and dispose on return if not supported.\n if (pc._config.bundlePolicy !== 'max-compat') {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n sdp += writeMediaSection(transceiver, transceiver.localCapabilities,\n 'offer', transceiver.stream, pc._dtlsRole);\n sdp += 'a=rtcp-rsize\\r\\n';\n\n if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' &&\n (sdpMLineIndex === 0 || !pc.usingBundle)) {\n transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) {\n cand.component = 1;\n sdp += 'a=' + SDPUtils.writeCandidate(cand) + '\\r\\n';\n });\n\n if (transceiver.iceGatherer.state === 'completed') {\n sdp += 'a=end-of-candidates\\r\\n';\n }\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'offer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.createAnswer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer after close'));\n }\n\n if (!(pc.signalingState === 'have-remote-offer' ||\n pc.signalingState === 'have-local-pranswer')) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer in signalingState ' + pc.signalingState));\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n if (pc.usingBundle) {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n var mediaSectionsInOffer = SDPUtils.getMediaSections(\n pc._remoteDescription.sdp).length;\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n if (sdpMLineIndex + 1 > mediaSectionsInOffer) {\n return;\n }\n if (transceiver.rejected) {\n if (transceiver.kind === 'application') {\n if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt\n sdp += 'm=application 0 DTLS/SCTP 5000\\r\\n';\n } else {\n sdp += 'm=application 0 ' + transceiver.protocol +\n ' webrtc-datachannel\\r\\n';\n }\n } else if (transceiver.kind === 'audio') {\n sdp += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\\r\\n' +\n 'a=rtpmap:0 PCMU/8000\\r\\n';\n } else if (transceiver.kind === 'video') {\n sdp += 'm=video 0 UDP/TLS/RTP/SAVPF 120\\r\\n' +\n 'a=rtpmap:120 VP8/90000\\r\\n';\n }\n sdp += 'c=IN IP4 0.0.0.0\\r\\n' +\n 'a=inactive\\r\\n' +\n 'a=mid:' + transceiver.mid + '\\r\\n';\n return;\n }\n\n // FIXME: look at direction.\n if (transceiver.stream) {\n var localTrack;\n if (transceiver.kind === 'audio') {\n localTrack = transceiver.stream.getAudioTracks()[0];\n } else if (transceiver.kind === 'video') {\n localTrack = transceiver.stream.getVideoTracks()[0];\n }\n if (localTrack) {\n // add RTX\n if (edgeVersion >= 15019 && transceiver.kind === 'video' &&\n !transceiver.sendEncodingParameters[0].rtx) {\n transceiver.sendEncodingParameters[0].rtx = {\n ssrc: transceiver.sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n }\n\n // Calculate intersection of capabilities.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n sdp += writeMediaSection(transceiver, commonCapabilities,\n 'answer', transceiver.stream, pc._dtlsRole);\n if (transceiver.rtcpParameters &&\n transceiver.rtcpParameters.reducedSize) {\n sdp += 'a=rtcp-rsize\\r\\n';\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'answer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.addIceCandidate = function(candidate) {\n var pc = this;\n var sections;\n if (candidate && !(candidate.sdpMLineIndex !== undefined ||\n candidate.sdpMid)) {\n return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required'));\n }\n\n // TODO: needs to go into ops queue.\n return new Promise(function(resolve, reject) {\n if (!pc._remoteDescription) {\n return reject(makeError('InvalidStateError',\n 'Can not add ICE candidate without a remote description'));\n } else if (!candidate || candidate.candidate === '') {\n for (var j = 0; j < pc.transceivers.length; j++) {\n if (pc.transceivers[j].rejected) {\n continue;\n }\n pc.transceivers[j].iceTransport.addRemoteCandidate({});\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[j] += 'a=end-of-candidates\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n if (pc.usingBundle) {\n break;\n }\n }\n } else {\n var sdpMLineIndex = candidate.sdpMLineIndex;\n if (candidate.sdpMid) {\n for (var i = 0; i < pc.transceivers.length; i++) {\n if (pc.transceivers[i].mid === candidate.sdpMid) {\n sdpMLineIndex = i;\n break;\n }\n }\n }\n var transceiver = pc.transceivers[sdpMLineIndex];\n if (transceiver) {\n if (transceiver.rejected) {\n return resolve();\n }\n var cand = Object.keys(candidate.candidate).length > 0 ?\n SDPUtils.parseCandidate(candidate.candidate) : {};\n // Ignore Chrome's invalid candidates since Edge does not like them.\n if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) {\n return resolve();\n }\n // Ignore RTCP candidates, we assume RTCP-MUX.\n if (cand.component && cand.component !== 1) {\n return resolve();\n }\n // when using bundle, avoid adding candidates to the wrong\n // ice transport. And avoid adding candidates added in the SDP.\n if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 &&\n transceiver.iceTransport !== pc.transceivers[0].iceTransport)) {\n if (!maybeAddCandidate(transceiver.iceTransport, cand)) {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n\n // update the remoteDescription.\n var candidateString = candidate.candidate.trim();\n if (candidateString.indexOf('a=') === 0) {\n candidateString = candidateString.substr(2);\n }\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[sdpMLineIndex] += 'a=' +\n (cand.type ? candidateString : 'end-of-candidates')\n + '\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n } else {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n resolve();\n });\n };\n\n RTCPeerConnection.prototype.getStats = function(selector) {\n if (selector && selector instanceof window.MediaStreamTrack) {\n var senderOrReceiver = null;\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.rtpSender &&\n transceiver.rtpSender.track === selector) {\n senderOrReceiver = transceiver.rtpSender;\n } else if (transceiver.rtpReceiver &&\n transceiver.rtpReceiver.track === selector) {\n senderOrReceiver = transceiver.rtpReceiver;\n }\n });\n if (!senderOrReceiver) {\n throw makeError('InvalidAccessError', 'Invalid selector.');\n }\n return senderOrReceiver.getStats();\n }\n\n var promises = [];\n this.transceivers.forEach(function(transceiver) {\n ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport',\n 'dtlsTransport'].forEach(function(method) {\n if (transceiver[method]) {\n promises.push(transceiver[method].getStats());\n }\n });\n });\n return Promise.all(promises).then(function(allStats) {\n var results = new Map();\n allStats.forEach(function(stats) {\n stats.forEach(function(stat) {\n results.set(stat.id, stat);\n });\n });\n return results;\n });\n };\n\n // fix low-level stat names and return Map instead of object.\n var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer',\n 'RTCIceTransport', 'RTCDtlsTransport'];\n ortcObjects.forEach(function(ortcObjectName) {\n var obj = window[ortcObjectName];\n if (obj && obj.prototype && obj.prototype.getStats) {\n var nativeGetstats = obj.prototype.getStats;\n obj.prototype.getStats = function() {\n return nativeGetstats.apply(this)\n .then(function(nativeStats) {\n var mapStats = new Map();\n Object.keys(nativeStats).forEach(function(id) {\n nativeStats[id].type = fixStatsType(nativeStats[id]);\n mapStats.set(id, nativeStats[id]);\n });\n return mapStats;\n });\n };\n }\n });\n\n // legacy callback shims. Should be moved to adapter.js some days.\n var methods = ['createOffer', 'createAnswer'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[0] === 'function' ||\n typeof args[1] === 'function') { // legacy\n return nativeMethod.apply(this, [arguments[2]])\n .then(function(description) {\n if (typeof args[0] === 'function') {\n args[0].apply(null, [description]);\n }\n }, function(error) {\n if (typeof args[1] === 'function') {\n args[1].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function' ||\n typeof args[2] === 'function') { // legacy\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n }, function(error) {\n if (typeof args[2] === 'function') {\n args[2].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n // getStats is special. It doesn't have a spec legacy method yet we support\n // getStats(something, cb) without error callbacks.\n ['getStats'].forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function') {\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n return RTCPeerConnection;\n};\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n const shimError_ = function(e) {\n return {\n name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name,\n message: e.message,\n constraint: e.constraint,\n toString() {\n return this.name;\n }\n };\n };\n\n // getUserMedia error shim.\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e)));\n };\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window) {\n if (!('getDisplayMedia' in window.navigator)) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n window.navigator.getDisplayMedia.bind(window.navigator);\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nimport {filterIceServers} from './filtericeservers';\nimport shimRTCPeerConnection from 'rtcpeerconnection-shim';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimPeerConnection(window, browserDetails) {\n if (window.RTCIceGatherer) {\n if (!window.RTCIceCandidate) {\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n return args;\n };\n }\n if (!window.RTCSessionDescription) {\n window.RTCSessionDescription = function RTCSessionDescription(args) {\n return args;\n };\n }\n // this adds an additional event listener to MediaStrackTrack that signals\n // when a tracks enabled property was changed. Workaround for a bug in\n // addStream, see below. No longer required in 15025+\n if (browserDetails.version < 15025) {\n const origMSTEnabled = Object.getOwnPropertyDescriptor(\n window.MediaStreamTrack.prototype, 'enabled');\n Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', {\n set(value) {\n origMSTEnabled.set.call(this, value);\n const ev = new Event('enabled');\n ev.enabled = value;\n this.dispatchEvent(ev);\n }\n });\n }\n }\n\n // ORTC defines the DTMF sender a bit different.\n // https://github.com/w3c/ortc/issues/714\n if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) {\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = new window.RTCDtmfSender(this);\n } else if (this.track.kind === 'video') {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n // Edge currently only implements the RTCDtmfSender, not the\n // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2*\n if (window.RTCDtmfSender && !window.RTCDTMFSender) {\n window.RTCDTMFSender = window.RTCDtmfSender;\n }\n\n const RTCPeerConnectionShim = shimRTCPeerConnection(window,\n browserDetails.version);\n window.RTCPeerConnection = function RTCPeerConnection(config) {\n if (config && config.iceServers) {\n config.iceServers = filterIceServers(config.iceServers,\n browserDetails.version);\n utils.log('ICE servers after filtering:', config.iceServers);\n }\n return new RTCPeerConnectionShim(config);\n };\n window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype;\n}\n\nexport function shimReplaceTrack(window) {\n // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614\n if (window.RTCRtpSender &&\n !('replaceTrack' in window.RTCRtpSender.prototype)) {\n window.RTCRtpSender.prototype.replaceTrack =\n window.RTCRtpSender.prototype.setTrack;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n const MediaStreamTrack = window && window.MediaStreamTrack;\n\n navigator.getUserMedia = function(constraints, onSuccess, onError) {\n // Replace Firefox 44+'s deprecation warning with unprefixed version.\n utils.deprecated('navigator.getUserMedia',\n 'navigator.mediaDevices.getUserMedia');\n navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError);\n };\n\n if (!(browserDetails.version > 55 &&\n 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n\n const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n if (typeof c === 'object' && typeof c.audio === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c.audio, 'autoGainControl', 'mozAutoGainControl');\n remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeGetUserMedia(c);\n };\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) {\n const nativeGetSettings = MediaStreamTrack.prototype.getSettings;\n MediaStreamTrack.prototype.getSettings = function() {\n const obj = nativeGetSettings.apply(this, arguments);\n remap(obj, 'mozAutoGainControl', 'autoGainControl');\n remap(obj, 'mozNoiseSuppression', 'noiseSuppression');\n return obj;\n };\n }\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) {\n const nativeApplyConstraints =\n MediaStreamTrack.prototype.applyConstraints;\n MediaStreamTrack.prototype.applyConstraints = function(c) {\n if (this.kind === 'audio' && typeof c === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c, 'autoGainControl', 'mozAutoGainControl');\n remap(c, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeApplyConstraints.apply(this, [c]);\n };\n }\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window, preferredMediaSource) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n if (!(constraints && constraints.video)) {\n const err = new DOMException('getDisplayMedia without video ' +\n 'constraints is undefined');\n err.name = 'NotFoundError';\n // from https://heycam.github.io/webidl/#idl-DOMException-error-names\n err.code = 8;\n return Promise.reject(err);\n }\n if (constraints.video === true) {\n constraints.video = {mediaSource: preferredMediaSource};\n } else {\n constraints.video.mediaSource = preferredMediaSource;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCTrackEvent &&\n ('receiver' in window.RTCTrackEvent.prototype) &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (typeof window !== 'object' ||\n !(window.RTCPeerConnection || window.mozRTCPeerConnection)) {\n return; // probably media.peerconnection.enabled=false in about:config\n }\n if (!window.RTCPeerConnection && window.mozRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.mozRTCPeerConnection;\n }\n\n if (browserDetails.version < 53) {\n // shim away need for obsolete RTCIceCandidate/RTCSessionDescription.\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n\n const modernStatsTypes = {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n };\n\n const nativeGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n return nativeGetStats.apply(this, [selector || null])\n .then(stats => {\n if (browserDetails.version < 53 && !onSucc) {\n // Shim only promise getStats with spec-hyphens in type names\n // Leave callback version alone; misc old uses of forEach before Map\n try {\n stats.forEach(stat => {\n stat.type = modernStatsTypes[stat.type] || stat.type;\n });\n } catch (e) {\n if (e.name !== 'TypeError') {\n throw e;\n }\n // Avoid TypeError: \"type\" is read-only, in old versions. 34-43ish\n stats.forEach((stat, i) => {\n stats.set(i, Object.assign({}, stat, {\n type: modernStatsTypes[stat.type] || stat.type\n }));\n });\n }\n }\n return stats;\n })\n .then(onSucc, onErr);\n };\n}\n\nexport function shimSenderGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) {\n return;\n }\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n return this.track ? this._pc.getStats(this.track) :\n Promise.resolve(new Map());\n };\n}\n\nexport function shimReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) {\n return;\n }\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers = function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n return this._pc.getStats(this.track);\n };\n}\n\nexport function shimRemoveStream(window) {\n if (!window.RTCPeerConnection ||\n 'removeStream' in window.RTCPeerConnection.prototype) {\n return;\n }\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n utils.deprecated('removeStream', 'removeTrack');\n this.getSenders().forEach(sender => {\n if (sender.track && stream.getTracks().includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n}\n\nexport function shimRTCDataChannel(window) {\n // rename DataChannel to RTCDataChannel (native fix in FF60):\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851\n if (window.DataChannel && !window.RTCDataChannel) {\n window.RTCDataChannel = window.DataChannel;\n }\n}\n\nexport function shimAddTransceiver(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver;\n if (origAddTransceiver) {\n window.RTCPeerConnection.prototype.addTransceiver =\n function addTransceiver() {\n this.setParametersPromises = [];\n const initParameters = arguments[1];\n const shouldPerformCheck = initParameters &&\n 'sendEncodings' in initParameters;\n if (shouldPerformCheck) {\n // If sendEncodings params are provided, validate grammar\n initParameters.sendEncodings.forEach((encodingParam) => {\n if ('rid' in encodingParam) {\n const ridRegex = /^[a-z0-9]{0,16}$/i;\n if (!ridRegex.test(encodingParam.rid)) {\n throw new TypeError('Invalid RID value provided.');\n }\n }\n if ('scaleResolutionDownBy' in encodingParam) {\n if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) {\n throw new RangeError('scale_resolution_down_by must be >= 1.0');\n }\n }\n if ('maxFramerate' in encodingParam) {\n if (!(parseFloat(encodingParam.maxFramerate) >= 0)) {\n throw new RangeError('max_framerate must be >= 0.0');\n }\n }\n });\n }\n const transceiver = origAddTransceiver.apply(this, arguments);\n if (shouldPerformCheck) {\n // Check if the init options were applied. If not we do this in an\n // asynchronous way and save the promise reference in a global object.\n // This is an ugly hack, but at the same time is way more robust than\n // checking the sender parameters before and after the createOffer\n // Also note that after the createoffer we are not 100% sure that\n // the params were asynchronously applied so we might miss the\n // opportunity to recreate offer.\n const {sender} = transceiver;\n const params = sender.getParameters();\n if (!('encodings' in params) ||\n // Avoid being fooled by patched getParameters() below.\n (params.encodings.length === 1 &&\n Object.keys(params.encodings[0]).length === 0)) {\n params.encodings = initParameters.sendEncodings;\n sender.sendEncodings = initParameters.sendEncodings;\n this.setParametersPromises.push(sender.setParameters(params)\n .then(() => {\n delete sender.sendEncodings;\n }).catch(() => {\n delete sender.sendEncodings;\n })\n );\n }\n }\n return transceiver;\n };\n }\n}\n\nexport function shimGetParameters(window) {\n if (!(typeof window === 'object' && window.RTCRtpSender)) {\n return;\n }\n const origGetParameters = window.RTCRtpSender.prototype.getParameters;\n if (origGetParameters) {\n window.RTCRtpSender.prototype.getParameters =\n function getParameters() {\n const params = origGetParameters.apply(this, arguments);\n if (!('encodings' in params)) {\n params.encodings = [].concat(this.sendEncodings || [{}]);\n }\n return params;\n };\n }\n}\n\nexport function shimCreateOffer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer = function createOffer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateOffer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimCreateAnswer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer;\n window.RTCPeerConnection.prototype.createAnswer = function createAnswer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateAnswer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateAnswer.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n'use strict';\nimport * as utils from '../utils';\n\nexport function shimLocalStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n return this._localStreams;\n };\n }\n if (!('addStream' in window.RTCPeerConnection.prototype)) {\n const _addTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n // Try to emulate Chrome's behaviour of adding in audio-video order.\n // Safari orders by track id.\n stream.getAudioTracks().forEach(track => _addTrack.call(this, track,\n stream));\n stream.getVideoTracks().forEach(track => _addTrack.call(this, track,\n stream));\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, ...streams) {\n if (streams) {\n streams.forEach((stream) => {\n if (!this._localStreams) {\n this._localStreams = [stream];\n } else if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n });\n }\n return _addTrack.apply(this, arguments);\n };\n }\n if (!('removeStream' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n const index = this._localStreams.indexOf(stream);\n if (index === -1) {\n return;\n }\n this._localStreams.splice(index, 1);\n const tracks = stream.getTracks();\n this.getSenders().forEach(sender => {\n if (tracks.includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n }\n}\n\nexport function shimRemoteStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getRemoteStreams =\n function getRemoteStreams() {\n return this._remoteStreams ? this._remoteStreams : [];\n };\n }\n if (!('onaddstream' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', {\n get() {\n return this._onaddstream;\n },\n set(f) {\n if (this._onaddstream) {\n this.removeEventListener('addstream', this._onaddstream);\n this.removeEventListener('track', this._onaddstreampoly);\n }\n this.addEventListener('addstream', this._onaddstream = f);\n this.addEventListener('track', this._onaddstreampoly = (e) => {\n e.streams.forEach(stream => {\n if (!this._remoteStreams) {\n this._remoteStreams = [];\n }\n if (this._remoteStreams.includes(stream)) {\n return;\n }\n this._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n this.dispatchEvent(event);\n });\n });\n }\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n const pc = this;\n if (!this._onaddstreampoly) {\n this.addEventListener('track', this._onaddstreampoly = function(e) {\n e.streams.forEach(stream => {\n if (!pc._remoteStreams) {\n pc._remoteStreams = [];\n }\n if (pc._remoteStreams.indexOf(stream) >= 0) {\n return;\n }\n pc._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n pc.dispatchEvent(event);\n });\n });\n }\n return origSetRemoteDescription.apply(pc, arguments);\n };\n }\n}\n\nexport function shimCallbacksAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n const prototype = window.RTCPeerConnection.prototype;\n const origCreateOffer = prototype.createOffer;\n const origCreateAnswer = prototype.createAnswer;\n const setLocalDescription = prototype.setLocalDescription;\n const setRemoteDescription = prototype.setRemoteDescription;\n const addIceCandidate = prototype.addIceCandidate;\n\n prototype.createOffer =\n function createOffer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateOffer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n prototype.createAnswer =\n function createAnswer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateAnswer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n let withCallback = function(description, successCallback, failureCallback) {\n const promise = setLocalDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setLocalDescription = withCallback;\n\n withCallback = function(description, successCallback, failureCallback) {\n const promise = setRemoteDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setRemoteDescription = withCallback;\n\n withCallback = function(candidate, successCallback, failureCallback) {\n const promise = addIceCandidate.apply(this, [candidate]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.addIceCandidate = withCallback;\n}\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {\n // shim not needed in Safari 12.1\n const mediaDevices = navigator.mediaDevices;\n const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices);\n navigator.mediaDevices.getUserMedia = (constraints) => {\n return _getUserMedia(shimConstraints(constraints));\n };\n }\n\n if (!navigator.getUserMedia && navigator.mediaDevices &&\n navigator.mediaDevices.getUserMedia) {\n navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) {\n navigator.mediaDevices.getUserMedia(constraints)\n .then(cb, errcb);\n }.bind(navigator);\n }\n}\n\nexport function shimConstraints(constraints) {\n if (constraints && constraints.video !== undefined) {\n return Object.assign({},\n constraints,\n {video: utils.compactObject(constraints.video)}\n );\n }\n\n return constraints;\n}\n\nexport function shimRTCIceServerUrls(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // migrate from non-spec RTCIceServer.url to RTCIceServer.urls\n const OrigPeerConnection = window.RTCPeerConnection;\n window.RTCPeerConnection =\n function RTCPeerConnection(pcConfig, pcConstraints) {\n if (pcConfig && pcConfig.iceServers) {\n const newIceServers = [];\n for (let i = 0; i < pcConfig.iceServers.length; i++) {\n let server = pcConfig.iceServers[i];\n if (!server.hasOwnProperty('urls') &&\n server.hasOwnProperty('url')) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n server = JSON.parse(JSON.stringify(server));\n server.urls = server.url;\n delete server.url;\n newIceServers.push(server);\n } else {\n newIceServers.push(pcConfig.iceServers[i]);\n }\n }\n pcConfig.iceServers = newIceServers;\n }\n return new OrigPeerConnection(pcConfig, pcConstraints);\n };\n window.RTCPeerConnection.prototype = OrigPeerConnection.prototype;\n // wrap static methods. Currently just generateCertificate.\n if ('generateCertificate' in OrigPeerConnection) {\n Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', {\n get() {\n return OrigPeerConnection.generateCertificate;\n }\n });\n }\n}\n\nexport function shimTrackEventTransceiver(window) {\n // Add event.transceiver member over deprecated event.receiver\n if (typeof window === 'object' && window.RTCTrackEvent &&\n 'receiver' in window.RTCTrackEvent.prototype &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimCreateOfferLegacy(window) {\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer =\n function createOffer(offerOptions) {\n if (offerOptions) {\n if (typeof offerOptions.offerToReceiveAudio !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveAudio =\n !!offerOptions.offerToReceiveAudio;\n }\n const audioTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'audio');\n if (offerOptions.offerToReceiveAudio === false && audioTransceiver) {\n if (audioTransceiver.direction === 'sendrecv') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('sendonly');\n } else {\n audioTransceiver.direction = 'sendonly';\n }\n } else if (audioTransceiver.direction === 'recvonly') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('inactive');\n } else {\n audioTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveAudio === true &&\n !audioTransceiver) {\n this.addTransceiver('audio');\n }\n\n if (typeof offerOptions.offerToReceiveVideo !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveVideo =\n !!offerOptions.offerToReceiveVideo;\n }\n const videoTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'video');\n if (offerOptions.offerToReceiveVideo === false && videoTransceiver) {\n if (videoTransceiver.direction === 'sendrecv') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('sendonly');\n } else {\n videoTransceiver.direction = 'sendonly';\n }\n } else if (videoTransceiver.direction === 'recvonly') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('inactive');\n } else {\n videoTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveVideo === true &&\n !videoTransceiver) {\n this.addTransceiver('video');\n }\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimAudioContext(window) {\n if (typeof window !== 'object' || window.AudioContext) {\n return;\n }\n window.AudioContext = window.webkitAudioContext;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport SDPUtils from 'sdp';\nimport * as utils from './utils';\n\nexport function shimRTCIceCandidate(window) {\n // foundation is arbitrarily chosen as an indicator for full support for\n // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface\n if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in\n window.RTCIceCandidate.prototype)) {\n return;\n }\n\n const NativeRTCIceCandidate = window.RTCIceCandidate;\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n // Remove the a= which shouldn't be part of the candidate string.\n if (typeof args === 'object' && args.candidate &&\n args.candidate.indexOf('a=') === 0) {\n args = JSON.parse(JSON.stringify(args));\n args.candidate = args.candidate.substr(2);\n }\n\n if (args.candidate && args.candidate.length) {\n // Augment the native candidate with the parsed fields.\n const nativeCandidate = new NativeRTCIceCandidate(args);\n const parsedCandidate = SDPUtils.parseCandidate(args.candidate);\n const augmentedCandidate = Object.assign(nativeCandidate,\n parsedCandidate);\n\n // Add a serializer that does not serialize the extra attributes.\n augmentedCandidate.toJSON = function toJSON() {\n return {\n candidate: augmentedCandidate.candidate,\n sdpMid: augmentedCandidate.sdpMid,\n sdpMLineIndex: augmentedCandidate.sdpMLineIndex,\n usernameFragment: augmentedCandidate.usernameFragment,\n };\n };\n return augmentedCandidate;\n }\n return new NativeRTCIceCandidate(args);\n };\n window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype;\n\n // Hook up the augmented candidate in onicecandidate and\n // addEventListener('icecandidate', ...)\n utils.wrapPeerConnectionEvent(window, 'icecandidate', e => {\n if (e.candidate) {\n Object.defineProperty(e, 'candidate', {\n value: new window.RTCIceCandidate(e.candidate),\n writable: 'false'\n });\n }\n return e;\n });\n}\n\nexport function shimMaxMessageSize(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n if (!('sctp' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n }\n });\n }\n\n const sctpInDescription = function(description) {\n if (!description || !description.sdp) {\n return false;\n }\n const sections = SDPUtils.splitSections(description.sdp);\n sections.shift();\n return sections.some(mediaSection => {\n const mLine = SDPUtils.parseMLine(mediaSection);\n return mLine && mLine.kind === 'application'\n && mLine.protocol.indexOf('SCTP') !== -1;\n });\n };\n\n const getRemoteFirefoxVersion = function(description) {\n // TODO: Is there a better solution for detecting Firefox?\n const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\\d+)/);\n if (match === null || match.length < 2) {\n return -1;\n }\n const version = parseInt(match[1], 10);\n // Test for NaN (yes, this is ugly)\n return version !== version ? -1 : version;\n };\n\n const getCanSendMaxMessageSize = function(remoteIsFirefox) {\n // Every implementation we know can send at least 64 KiB.\n // Note: Although Chrome is technically able to send up to 256 KiB, the\n // data does not reach the other peer reliably.\n // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419\n let canSendMaxMessageSize = 65536;\n if (browserDetails.browser === 'firefox') {\n if (browserDetails.version < 57) {\n if (remoteIsFirefox === -1) {\n // FF < 57 will send in 16 KiB chunks using the deprecated PPID\n // fragmentation.\n canSendMaxMessageSize = 16384;\n } else {\n // However, other FF (and RAWRTC) can reassemble PPID-fragmented\n // messages. Thus, supporting ~2 GiB when sending.\n canSendMaxMessageSize = 2147483637;\n }\n } else if (browserDetails.version < 60) {\n // Currently, all FF >= 57 will reset the remote maximum message size\n // to the default value when a data channel is created at a later\n // stage. :(\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n canSendMaxMessageSize =\n browserDetails.version === 57 ? 65535 : 65536;\n } else {\n // FF >= 60 supports sending ~2 GiB\n canSendMaxMessageSize = 2147483637;\n }\n }\n return canSendMaxMessageSize;\n };\n\n const getMaxMessageSize = function(description, remoteIsFirefox) {\n // Note: 65536 bytes is the default value from the SDP spec. Also,\n // every implementation we know supports receiving 65536 bytes.\n let maxMessageSize = 65536;\n\n // FF 57 has a slightly incorrect default remote max message size, so\n // we need to adjust it here to avoid a failure when sending.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697\n if (browserDetails.browser === 'firefox'\n && browserDetails.version === 57) {\n maxMessageSize = 65535;\n }\n\n const match = SDPUtils.matchPrefix(description.sdp,\n 'a=max-message-size:');\n if (match.length > 0) {\n maxMessageSize = parseInt(match[0].substr(19), 10);\n } else if (browserDetails.browser === 'firefox' &&\n remoteIsFirefox !== -1) {\n // If the maximum message size is not present in the remote SDP and\n // both local and remote are Firefox, the remote peer can receive\n // ~2 GiB.\n maxMessageSize = 2147483637;\n }\n return maxMessageSize;\n };\n\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n this._sctp = null;\n // Chrome decided to not expose .sctp in plan-b mode.\n // As usual, adapter.js has to do an 'ugly worakaround'\n // to cover up the mess.\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) {\n const {sdpSemantics} = this.getConfiguration();\n if (sdpSemantics === 'plan-b') {\n Object.defineProperty(this, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n },\n enumerable: true,\n configurable: true,\n });\n }\n }\n\n if (sctpInDescription(arguments[0])) {\n // Check if the remote is FF.\n const isFirefox = getRemoteFirefoxVersion(arguments[0]);\n\n // Get the maximum message size the local peer is capable of sending\n const canSendMMS = getCanSendMaxMessageSize(isFirefox);\n\n // Get the maximum message size of the remote peer.\n const remoteMMS = getMaxMessageSize(arguments[0], isFirefox);\n\n // Determine final maximum message size\n let maxMessageSize;\n if (canSendMMS === 0 && remoteMMS === 0) {\n maxMessageSize = Number.POSITIVE_INFINITY;\n } else if (canSendMMS === 0 || remoteMMS === 0) {\n maxMessageSize = Math.max(canSendMMS, remoteMMS);\n } else {\n maxMessageSize = Math.min(canSendMMS, remoteMMS);\n }\n\n // Create a dummy RTCSctpTransport object and the 'maxMessageSize'\n // attribute.\n const sctp = {};\n Object.defineProperty(sctp, 'maxMessageSize', {\n get() {\n return maxMessageSize;\n }\n });\n this._sctp = sctp;\n }\n\n return origSetRemoteDescription.apply(this, arguments);\n };\n}\n\nexport function shimSendThrowTypeError(window) {\n if (!(window.RTCPeerConnection &&\n 'createDataChannel' in window.RTCPeerConnection.prototype)) {\n return;\n }\n\n // Note: Although Firefox >= 57 has a native implementation, the maximum\n // message size can be reset for all data channels at a later stage.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n\n function wrapDcSend(dc, pc) {\n const origDataChannelSend = dc.send;\n dc.send = function send() {\n const data = arguments[0];\n const length = data.length || data.size || data.byteLength;\n if (dc.readyState === 'open' &&\n pc.sctp && length > pc.sctp.maxMessageSize) {\n throw new TypeError('Message too large (can send a maximum of ' +\n pc.sctp.maxMessageSize + ' bytes)');\n }\n return origDataChannelSend.apply(dc, arguments);\n };\n }\n const origCreateDataChannel =\n window.RTCPeerConnection.prototype.createDataChannel;\n window.RTCPeerConnection.prototype.createDataChannel =\n function createDataChannel() {\n const dataChannel = origCreateDataChannel.apply(this, arguments);\n wrapDcSend(dataChannel, this);\n return dataChannel;\n };\n utils.wrapPeerConnectionEvent(window, 'datachannel', e => {\n wrapDcSend(e.channel, e.target);\n return e;\n });\n}\n\n\n/* shims RTCConnectionState by pretending it is the same as iceConnectionState.\n * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12\n * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect\n * since DTLS failures would be hidden. See\n * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827\n * for the Firefox tracking bug.\n */\nexport function shimConnectionState(window) {\n if (!window.RTCPeerConnection ||\n 'connectionState' in window.RTCPeerConnection.prototype) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n Object.defineProperty(proto, 'connectionState', {\n get() {\n return {\n completed: 'connected',\n checking: 'connecting'\n }[this.iceConnectionState] || this.iceConnectionState;\n },\n enumerable: true,\n configurable: true\n });\n Object.defineProperty(proto, 'onconnectionstatechange', {\n get() {\n return this._onconnectionstatechange || null;\n },\n set(cb) {\n if (this._onconnectionstatechange) {\n this.removeEventListener('connectionstatechange',\n this._onconnectionstatechange);\n delete this._onconnectionstatechange;\n }\n if (cb) {\n this.addEventListener('connectionstatechange',\n this._onconnectionstatechange = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n\n ['setLocalDescription', 'setRemoteDescription'].forEach((method) => {\n const origMethod = proto[method];\n proto[method] = function() {\n if (!this._connectionstatechangepoly) {\n this._connectionstatechangepoly = e => {\n const pc = e.target;\n if (pc._lastConnectionState !== pc.connectionState) {\n pc._lastConnectionState = pc.connectionState;\n const newEvent = new Event('connectionstatechange', e);\n pc.dispatchEvent(newEvent);\n }\n return e;\n };\n this.addEventListener('iceconnectionstatechange',\n this._connectionstatechangepoly);\n }\n return origMethod.apply(this, arguments);\n };\n });\n}\n\nexport function removeExtmapAllowMixed(window, browserDetails) {\n /* remove a=extmap-allow-mixed for webrtc.org < M71 */\n if (!window.RTCPeerConnection) {\n return;\n }\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) {\n return;\n }\n if (browserDetails.browser === 'safari' && browserDetails.version >= 605) {\n return;\n }\n const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription(desc) {\n if (desc && desc.sdp && desc.sdp.indexOf('\\na=extmap-allow-mixed') !== -1) {\n const sdp = desc.sdp.split('\\n').filter((line) => {\n return line.trim() !== 'a=extmap-allow-mixed';\n }).join('\\n');\n // Safari enforces read-only-ness of RTCSessionDescription fields.\n if (window.RTCSessionDescription &&\n desc instanceof window.RTCSessionDescription) {\n arguments[0] = new window.RTCSessionDescription({\n type: desc.type,\n sdp,\n });\n } else {\n desc.sdp = sdp;\n }\n }\n return nativeSRD.apply(this, arguments);\n };\n}\n\nexport function shimAddIceCandidateNullOrEmpty(window, browserDetails) {\n // Support for addIceCandidate(null or undefined)\n // as well as addIceCandidate({candidate: \"\", ...})\n // https://bugs.chromium.org/p/chromium/issues/detail?id=978582\n // Note: must be called before other polyfills which change the signature.\n if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) {\n return;\n }\n const nativeAddIceCandidate =\n window.RTCPeerConnection.prototype.addIceCandidate;\n if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) {\n return;\n }\n window.RTCPeerConnection.prototype.addIceCandidate =\n function addIceCandidate() {\n if (!arguments[0]) {\n if (arguments[1]) {\n arguments[1].apply(null);\n }\n return Promise.resolve();\n }\n // Firefox 68+ emits and processes {candidate: \"\", ...}, ignore\n // in older versions.\n // Native support for ignoring exists for Chrome M77+.\n // Safari ignores as well, exact version unknown but works in the same\n // version that also ignores addIceCandidate(null).\n if (((browserDetails.browser === 'chrome' && browserDetails.version < 78)\n || (browserDetails.browser === 'firefox'\n && browserDetails.version < 68)\n || (browserDetails.browser === 'safari'))\n && arguments[0] && arguments[0].candidate === '') {\n return Promise.resolve();\n }\n return nativeAddIceCandidate.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\nimport * as utils from './utils';\n\n // Browser shims.\nimport * as chromeShim from './chrome/chrome_shim';\nimport * as edgeShim from './edge/edge_shim';\nimport * as firefoxShim from './firefox/firefox_shim';\nimport * as safariShim from './safari/safari_shim';\nimport * as commonShim from './common_shim';\n\n// Shimming starts here.\nexport function adapterFactory({window} = {}, options = {\n shimChrome: true,\n shimFirefox: true,\n shimEdge: true,\n shimSafari: true,\n}) {\n // Utils.\n const logging = utils.log;\n const browserDetails = utils.detectBrowser(window);\n\n const adapter = {\n browserDetails,\n commonShim,\n extractVersion: utils.extractVersion,\n disableLog: utils.disableLog,\n disableWarnings: utils.disableWarnings\n };\n\n // Shim browser if found.\n switch (browserDetails.browser) {\n case 'chrome':\n if (!chromeShim || !chromeShim.shimPeerConnection ||\n !options.shimChrome) {\n logging('Chrome shim is not included in this adapter release.');\n return adapter;\n }\n if (browserDetails.version === null) {\n logging('Chrome shim can not determine version, not shimming.');\n return adapter;\n }\n logging('adapter.js shimming chrome.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = chromeShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n chromeShim.shimGetUserMedia(window, browserDetails);\n chromeShim.shimMediaStream(window, browserDetails);\n chromeShim.shimPeerConnection(window, browserDetails);\n chromeShim.shimOnTrack(window, browserDetails);\n chromeShim.shimAddTrackRemoveTrack(window, browserDetails);\n chromeShim.shimGetSendersWithDtmf(window, browserDetails);\n chromeShim.shimGetStats(window, browserDetails);\n chromeShim.shimSenderReceiverGetStats(window, browserDetails);\n chromeShim.fixNegotiationNeeded(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n case 'firefox':\n if (!firefoxShim || !firefoxShim.shimPeerConnection ||\n !options.shimFirefox) {\n logging('Firefox shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming firefox.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = firefoxShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n firefoxShim.shimGetUserMedia(window, browserDetails);\n firefoxShim.shimPeerConnection(window, browserDetails);\n firefoxShim.shimOnTrack(window, browserDetails);\n firefoxShim.shimRemoveStream(window, browserDetails);\n firefoxShim.shimSenderGetStats(window, browserDetails);\n firefoxShim.shimReceiverGetStats(window, browserDetails);\n firefoxShim.shimRTCDataChannel(window, browserDetails);\n firefoxShim.shimAddTransceiver(window, browserDetails);\n firefoxShim.shimGetParameters(window, browserDetails);\n firefoxShim.shimCreateOffer(window, browserDetails);\n firefoxShim.shimCreateAnswer(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'edge':\n if (!edgeShim || !edgeShim.shimPeerConnection || !options.shimEdge) {\n logging('MS edge shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming edge.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = edgeShim;\n\n edgeShim.shimGetUserMedia(window, browserDetails);\n edgeShim.shimGetDisplayMedia(window, browserDetails);\n edgeShim.shimPeerConnection(window, browserDetails);\n edgeShim.shimReplaceTrack(window, browserDetails);\n\n // the edge shim implements the full RTCIceCandidate object.\n\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'safari':\n if (!safariShim || !options.shimSafari) {\n logging('Safari shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming safari.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = safariShim;\n\n // Must be called before shimCallbackAPI.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n safariShim.shimRTCIceServerUrls(window, browserDetails);\n safariShim.shimCreateOfferLegacy(window, browserDetails);\n safariShim.shimCallbacksAPI(window, browserDetails);\n safariShim.shimLocalStreamsAPI(window, browserDetails);\n safariShim.shimRemoteStreamsAPI(window, browserDetails);\n safariShim.shimTrackEventTransceiver(window, browserDetails);\n safariShim.shimGetUserMedia(window, browserDetails);\n safariShim.shimAudioContext(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n default:\n logging('Unsupported browser!');\n break;\n }\n\n return adapter;\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n\n'use strict';\n\nimport {adapterFactory} from './adapter_factory.js';\n\nconst adapter =\n adapterFactory({window: typeof window === 'undefined' ? undefined : window});\nexport default adapter;\n","\n'use strict';\nimport * as utils from './utils.js';\nimport * as MediaFormatModule from './mediaformat.js';\nimport adapter from 'webrtc-adapter';\n\n/**\n * @class AudioTrackConstraints\n * @classDesc Constraints for creating an audio MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track.\n */\nexport class AudioTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.AudioSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Values could be \"mic\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Do not provide deviceId if source is not \"mic\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n this.deviceId = undefined;\n }\n}\n\n/**\n * @class VideoTrackConstraints\n * @classDesc Constraints for creating a video MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.VideoSourceInfo} source Source info of this video track.\n */\nexport class VideoTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.VideoSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Values could be \"camera\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Do not provide deviceId if source is not \"camera\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n\n this.deviceId = undefined;\n\n /**\n * @member {Owt.Base.Resolution} resolution\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.resolution = undefined;\n\n /**\n * @member {number} frameRate\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.frameRate = undefined;\n }\n}\n/**\n * @class StreamConstraints\n * @classDesc Constraints for creating a MediaStream from screen mic and camera.\n * @memberof Owt.Base\n * @constructor\n * @param {?Owt.Base.AudioTrackConstraints} audioConstraints\n * @param {?Owt.Base.VideoTrackConstraints} videoConstraints\n */\nexport class StreamConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(audioConstraints = false, videoConstraints = false) {\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.audio = audioConstraints;\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.video = videoConstraints;\n }\n}\n\n// eslint-disable-next-line require-jsdoc\nfunction isVideoConstrainsForScreenCast(constraints) {\n return (typeof constraints.video === 'object' && constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST);\n}\n\n/**\n * @class MediaStreamFactory\n * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself.\n * @memberof Owt.Base\n */\nexport class MediaStreamFactory {\n /**\n * @function createMediaStream\n * @static\n * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are \"screen-cast\".\n * @memberof Owt.Base.MediaStreamFactory\n * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened:\n * - One or more parameters cannot be satisfied.\n * - Specified device is busy.\n * - Cannot obtain necessary permission or operation is canceled by user.\n * - Video source is screen cast, while audio source is not.\n * - Audio source is screen cast, while video source is disabled.\n * @param {Owt.Base.StreamConstraints} constraints\n */\n static createMediaStream(constraints) {\n if (typeof constraints !== 'object' ||\n (!constraints.audio && !constraints.video)) {\n return Promise.reject(new TypeError('Invalid constrains'));\n }\n if (!isVideoConstrainsForScreenCast(constraints) &&\n (typeof constraints.audio === 'object') &&\n constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(\n new TypeError('Cannot share screen without video.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) && !utils.isChrome() &&\n !utils.isFirefox()) {\n return Promise.reject(\n new TypeError('Screen sharing only supports Chrome and Firefox.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) &&\n typeof constraints.audio === 'object' &&\n constraints.audio.source !==\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(new TypeError(\n 'Cannot capture video from screen cast while capture audio from'\n + ' other source.'));\n }\n\n // Check and convert constraints.\n if (!constraints.audio && !constraints.video) {\n return Promise.reject(new TypeError(\n 'At least one of audio and video must be requested.'));\n }\n const mediaConstraints = Object.create({});\n if (typeof constraints.audio === 'object' &&\n constraints.audio.source === MediaFormatModule.AudioSourceInfo.MIC) {\n mediaConstraints.audio = Object.create({});\n if (utils.isEdge()) {\n mediaConstraints.audio.deviceId = constraints.audio.deviceId;\n } else {\n mediaConstraints.audio.deviceId = {\n exact: constraints.audio.deviceId,\n };\n }\n } else {\n if (constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n mediaConstraints.audio = true;\n } else {\n mediaConstraints.audio = constraints.audio;\n }\n }\n if (typeof constraints.video === 'object') {\n mediaConstraints.video = Object.create({});\n if (typeof constraints.video.frameRate === 'number') {\n mediaConstraints.video.frameRate = constraints.video.frameRate;\n }\n if (constraints.video.resolution &&\n constraints.video.resolution.width &&\n constraints.video.resolution.height) {\n if (constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.width = constraints.video.resolution.width;\n mediaConstraints.video.height = constraints.video.resolution.height;\n } else {\n mediaConstraints.video.width = Object.create({});\n mediaConstraints.video.width.exact =\n constraints.video.resolution.width;\n mediaConstraints.video.height = Object.create({});\n mediaConstraints.video.height.exact =\n constraints.video.resolution.height;\n }\n }\n if (typeof constraints.video.deviceId === 'string') {\n mediaConstraints.video.deviceId = {exact: constraints.video.deviceId};\n }\n if (utils.isFirefox() &&\n constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.mediaSource = 'screen';\n }\n } else {\n mediaConstraints.video = constraints.video;\n }\n\n if (isVideoConstrainsForScreenCast(constraints)) {\n return navigator.mediaDevices.getDisplayMedia(mediaConstraints);\n } else {\n return navigator.mediaDevices.getUserMedia(mediaConstraints);\n }\n }\n}\n","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n\nexport * from './mediastream-factory.js';\nexport * from './mediaformat.js';","let logger;\nlet errorLogger;\n\nexport function setLogger() {\n /*eslint-disable */\n logger = console.log;\n errorLogger = console.error;\n /*eslint-enable */\n}\n\nexport function isEnable() {\n return logger != null;\n}\n\nexport function log(message, ...optionalParams) {\n if (logger) {\n logger(message, ...optionalParams);\n }\n}\nexport function error(message, ...optionalParams) {\n if (errorLogger) {\n errorLogger(message, ...optionalParams);\n }\n}\n","export default class Event {\n constructor(type) {\n this.listener = {};\n this.type = type | '';\n }\n\n on(event, fn) {\n if (!this.listener[event]) {\n this.listener[event] = [];\n }\n this.listener[event].push(fn);\n return true;\n }\n\n off(event, fn) {\n if (this.listener[event]) {\n var index = this.listener[event].indexOf(fn);\n if (index > -1) {\n this.listener[event].splice(index, 1);\n }\n return true;\n }\n return false;\n }\n\n offAll() {\n this.listener = {};\n }\n\n dispatch(event, data) {\n if (this.listener[event]) {\n this.listener[event].map((each) => {\n each.apply(null, [data]);\n });\n return true;\n }\n return false;\n }\n}\n","'use strict';\n\nmodule.exports = function bind(fn, thisArg) {\n return function wrap() {\n var args = new Array(arguments.length);\n for (var i = 0; i < args.length; i++) {\n args[i] = arguments[i];\n }\n return fn.apply(thisArg, args);\n };\n};\n","'use strict';\n\nvar bind = require('./helpers/bind');\n\n/*global toString:true*/\n\n// utils is a library of generic helper functions non-specific to axios\n\nvar toString = Object.prototype.toString;\n\n/**\n * Determine if a value is an Array\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Array, otherwise false\n */\nfunction isArray(val) {\n return toString.call(val) === '[object Array]';\n}\n\n/**\n * Determine if a value is undefined\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if the value is undefined, otherwise false\n */\nfunction isUndefined(val) {\n return typeof val === 'undefined';\n}\n\n/**\n * Determine if a value is a Buffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Buffer, otherwise false\n */\nfunction isBuffer(val) {\n return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor)\n && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val);\n}\n\n/**\n * Determine if a value is an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an ArrayBuffer, otherwise false\n */\nfunction isArrayBuffer(val) {\n return toString.call(val) === '[object ArrayBuffer]';\n}\n\n/**\n * Determine if a value is a FormData\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an FormData, otherwise false\n */\nfunction isFormData(val) {\n return (typeof FormData !== 'undefined') && (val instanceof FormData);\n}\n\n/**\n * Determine if a value is a view on an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false\n */\nfunction isArrayBufferView(val) {\n var result;\n if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) {\n result = ArrayBuffer.isView(val);\n } else {\n result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer);\n }\n return result;\n}\n\n/**\n * Determine if a value is a String\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a String, otherwise false\n */\nfunction isString(val) {\n return typeof val === 'string';\n}\n\n/**\n * Determine if a value is a Number\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Number, otherwise false\n */\nfunction isNumber(val) {\n return typeof val === 'number';\n}\n\n/**\n * Determine if a value is an Object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Object, otherwise false\n */\nfunction isObject(val) {\n return val !== null && typeof val === 'object';\n}\n\n/**\n * Determine if a value is a plain Object\n *\n * @param {Object} val The value to test\n * @return {boolean} True if value is a plain Object, otherwise false\n */\nfunction isPlainObject(val) {\n if (toString.call(val) !== '[object Object]') {\n return false;\n }\n\n var prototype = Object.getPrototypeOf(val);\n return prototype === null || prototype === Object.prototype;\n}\n\n/**\n * Determine if a value is a Date\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Date, otherwise false\n */\nfunction isDate(val) {\n return toString.call(val) === '[object Date]';\n}\n\n/**\n * Determine if a value is a File\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a File, otherwise false\n */\nfunction isFile(val) {\n return toString.call(val) === '[object File]';\n}\n\n/**\n * Determine if a value is a Blob\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Blob, otherwise false\n */\nfunction isBlob(val) {\n return toString.call(val) === '[object Blob]';\n}\n\n/**\n * Determine if a value is a Function\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Function, otherwise false\n */\nfunction isFunction(val) {\n return toString.call(val) === '[object Function]';\n}\n\n/**\n * Determine if a value is a Stream\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Stream, otherwise false\n */\nfunction isStream(val) {\n return isObject(val) && isFunction(val.pipe);\n}\n\n/**\n * Determine if a value is a URLSearchParams object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a URLSearchParams object, otherwise false\n */\nfunction isURLSearchParams(val) {\n return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams;\n}\n\n/**\n * Trim excess whitespace off the beginning and end of a string\n *\n * @param {String} str The String to trim\n * @returns {String} The String freed of excess whitespace\n */\nfunction trim(str) {\n return str.replace(/^\\s*/, '').replace(/\\s*$/, '');\n}\n\n/**\n * Determine if we're running in a standard browser environment\n *\n * This allows axios to run in a web worker, and react-native.\n * Both environments support XMLHttpRequest, but not fully standard globals.\n *\n * web workers:\n * typeof window -> undefined\n * typeof document -> undefined\n *\n * react-native:\n * navigator.product -> 'ReactNative'\n * nativescript\n * navigator.product -> 'NativeScript' or 'NS'\n */\nfunction isStandardBrowserEnv() {\n if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' ||\n navigator.product === 'NativeScript' ||\n navigator.product === 'NS')) {\n return false;\n }\n return (\n typeof window !== 'undefined' &&\n typeof document !== 'undefined'\n );\n}\n\n/**\n * Iterate over an Array or an Object invoking a function for each item.\n *\n * If `obj` is an Array callback will be called passing\n * the value, index, and complete array for each item.\n *\n * If 'obj' is an Object callback will be called passing\n * the value, key, and complete object for each property.\n *\n * @param {Object|Array} obj The object to iterate\n * @param {Function} fn The callback to invoke for each item\n */\nfunction forEach(obj, fn) {\n // Don't bother if no value provided\n if (obj === null || typeof obj === 'undefined') {\n return;\n }\n\n // Force an array if not already something iterable\n if (typeof obj !== 'object') {\n /*eslint no-param-reassign:0*/\n obj = [obj];\n }\n\n if (isArray(obj)) {\n // Iterate over array values\n for (var i = 0, l = obj.length; i < l; i++) {\n fn.call(null, obj[i], i, obj);\n }\n } else {\n // Iterate over object keys\n for (var key in obj) {\n if (Object.prototype.hasOwnProperty.call(obj, key)) {\n fn.call(null, obj[key], key, obj);\n }\n }\n }\n}\n\n/**\n * Accepts varargs expecting each argument to be an object, then\n * immutably merges the properties of each object and returns result.\n *\n * When multiple objects contain the same key the later object in\n * the arguments list will take precedence.\n *\n * Example:\n *\n * ```js\n * var result = merge({foo: 123}, {foo: 456});\n * console.log(result.foo); // outputs 456\n * ```\n *\n * @param {Object} obj1 Object to merge\n * @returns {Object} Result of all merge properties\n */\nfunction merge(/* obj1, obj2, obj3, ... */) {\n var result = {};\n function assignValue(val, key) {\n if (isPlainObject(result[key]) && isPlainObject(val)) {\n result[key] = merge(result[key], val);\n } else if (isPlainObject(val)) {\n result[key] = merge({}, val);\n } else if (isArray(val)) {\n result[key] = val.slice();\n } else {\n result[key] = val;\n }\n }\n\n for (var i = 0, l = arguments.length; i < l; i++) {\n forEach(arguments[i], assignValue);\n }\n return result;\n}\n\n/**\n * Extends object a by mutably adding to it the properties of object b.\n *\n * @param {Object} a The object to be extended\n * @param {Object} b The object to copy properties from\n * @param {Object} thisArg The object to bind function to\n * @return {Object} The resulting value of object a\n */\nfunction extend(a, b, thisArg) {\n forEach(b, function assignValue(val, key) {\n if (thisArg && typeof val === 'function') {\n a[key] = bind(val, thisArg);\n } else {\n a[key] = val;\n }\n });\n return a;\n}\n\n/**\n * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM)\n *\n * @param {string} content with BOM\n * @return {string} content value without BOM\n */\nfunction stripBOM(content) {\n if (content.charCodeAt(0) === 0xFEFF) {\n content = content.slice(1);\n }\n return content;\n}\n\nmodule.exports = {\n isArray: isArray,\n isArrayBuffer: isArrayBuffer,\n isBuffer: isBuffer,\n isFormData: isFormData,\n isArrayBufferView: isArrayBufferView,\n isString: isString,\n isNumber: isNumber,\n isObject: isObject,\n isPlainObject: isPlainObject,\n isUndefined: isUndefined,\n isDate: isDate,\n isFile: isFile,\n isBlob: isBlob,\n isFunction: isFunction,\n isStream: isStream,\n isURLSearchParams: isURLSearchParams,\n isStandardBrowserEnv: isStandardBrowserEnv,\n forEach: forEach,\n merge: merge,\n extend: extend,\n trim: trim,\n stripBOM: stripBOM\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction encode(val) {\n return encodeURIComponent(val).\n replace(/%3A/gi, ':').\n replace(/%24/g, '$').\n replace(/%2C/gi, ',').\n replace(/%20/g, '+').\n replace(/%5B/gi, '[').\n replace(/%5D/gi, ']');\n}\n\n/**\n * Build a URL by appending params to the end\n *\n * @param {string} url The base of the url (e.g., http://www.google.com)\n * @param {object} [params] The params to be appended\n * @returns {string} The formatted url\n */\nmodule.exports = function buildURL(url, params, paramsSerializer) {\n /*eslint no-param-reassign:0*/\n if (!params) {\n return url;\n }\n\n var serializedParams;\n if (paramsSerializer) {\n serializedParams = paramsSerializer(params);\n } else if (utils.isURLSearchParams(params)) {\n serializedParams = params.toString();\n } else {\n var parts = [];\n\n utils.forEach(params, function serialize(val, key) {\n if (val === null || typeof val === 'undefined') {\n return;\n }\n\n if (utils.isArray(val)) {\n key = key + '[]';\n } else {\n val = [val];\n }\n\n utils.forEach(val, function parseValue(v) {\n if (utils.isDate(v)) {\n v = v.toISOString();\n } else if (utils.isObject(v)) {\n v = JSON.stringify(v);\n }\n parts.push(encode(key) + '=' + encode(v));\n });\n });\n\n serializedParams = parts.join('&');\n }\n\n if (serializedParams) {\n var hashmarkIndex = url.indexOf('#');\n if (hashmarkIndex !== -1) {\n url = url.slice(0, hashmarkIndex);\n }\n\n url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams;\n }\n\n return url;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction InterceptorManager() {\n this.handlers = [];\n}\n\n/**\n * Add a new interceptor to the stack\n *\n * @param {Function} fulfilled The function to handle `then` for a `Promise`\n * @param {Function} rejected The function to handle `reject` for a `Promise`\n *\n * @return {Number} An ID used to remove interceptor later\n */\nInterceptorManager.prototype.use = function use(fulfilled, rejected) {\n this.handlers.push({\n fulfilled: fulfilled,\n rejected: rejected\n });\n return this.handlers.length - 1;\n};\n\n/**\n * Remove an interceptor from the stack\n *\n * @param {Number} id The ID that was returned by `use`\n */\nInterceptorManager.prototype.eject = function eject(id) {\n if (this.handlers[id]) {\n this.handlers[id] = null;\n }\n};\n\n/**\n * Iterate over all the registered interceptors\n *\n * This method is particularly useful for skipping over any\n * interceptors that may have become `null` calling `eject`.\n *\n * @param {Function} fn The function to call for each interceptor\n */\nInterceptorManager.prototype.forEach = function forEach(fn) {\n utils.forEach(this.handlers, function forEachHandler(h) {\n if (h !== null) {\n fn(h);\n }\n });\n};\n\nmodule.exports = InterceptorManager;\n","'use strict';\n\nvar utils = require('./../utils');\n\n/**\n * Transform the data for a request or a response\n *\n * @param {Object|String} data The data to be transformed\n * @param {Array} headers The headers for the request or response\n * @param {Array|Function} fns A single function or Array of functions\n * @returns {*} The resulting transformed data\n */\nmodule.exports = function transformData(data, headers, fns) {\n /*eslint no-param-reassign:0*/\n utils.forEach(fns, function transform(fn) {\n data = fn(data, headers);\n });\n\n return data;\n};\n","'use strict';\n\nmodule.exports = function isCancel(value) {\n return !!(value && value.__CANCEL__);\n};\n","'use strict';\n\nvar utils = require('../utils');\n\nmodule.exports = function normalizeHeaderName(headers, normalizedName) {\n utils.forEach(headers, function processHeader(value, name) {\n if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) {\n headers[normalizedName] = value;\n delete headers[name];\n }\n });\n};\n","'use strict';\n\n/**\n * Update an Error with the specified config, error code, and response.\n *\n * @param {Error} error The error to update.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The error.\n */\nmodule.exports = function enhanceError(error, config, code, request, response) {\n error.config = config;\n if (code) {\n error.code = code;\n }\n\n error.request = request;\n error.response = response;\n error.isAxiosError = true;\n\n error.toJSON = function toJSON() {\n return {\n // Standard\n message: this.message,\n name: this.name,\n // Microsoft\n description: this.description,\n number: this.number,\n // Mozilla\n fileName: this.fileName,\n lineNumber: this.lineNumber,\n columnNumber: this.columnNumber,\n stack: this.stack,\n // Axios\n config: this.config,\n code: this.code\n };\n };\n return error;\n};\n","'use strict';\n\nvar enhanceError = require('./enhanceError');\n\n/**\n * Create an Error with the specified message, config, error code, request and response.\n *\n * @param {string} message The error message.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The created error.\n */\nmodule.exports = function createError(message, config, code, request, response) {\n var error = new Error(message);\n return enhanceError(error, config, code, request, response);\n};\n","'use strict';\n\nvar createError = require('./createError');\n\n/**\n * Resolve or reject a Promise based on response status.\n *\n * @param {Function} resolve A function that resolves the promise.\n * @param {Function} reject A function that rejects the promise.\n * @param {object} response The response.\n */\nmodule.exports = function settle(resolve, reject, response) {\n var validateStatus = response.config.validateStatus;\n if (!response.status || !validateStatus || validateStatus(response.status)) {\n resolve(response);\n } else {\n reject(createError(\n 'Request failed with status code ' + response.status,\n response.config,\n null,\n response.request,\n response\n ));\n }\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs support document.cookie\n (function standardBrowserEnv() {\n return {\n write: function write(name, value, expires, path, domain, secure) {\n var cookie = [];\n cookie.push(name + '=' + encodeURIComponent(value));\n\n if (utils.isNumber(expires)) {\n cookie.push('expires=' + new Date(expires).toGMTString());\n }\n\n if (utils.isString(path)) {\n cookie.push('path=' + path);\n }\n\n if (utils.isString(domain)) {\n cookie.push('domain=' + domain);\n }\n\n if (secure === true) {\n cookie.push('secure');\n }\n\n document.cookie = cookie.join('; ');\n },\n\n read: function read(name) {\n var match = document.cookie.match(new RegExp('(^|;\\\\s*)(' + name + ')=([^;]*)'));\n return (match ? decodeURIComponent(match[3]) : null);\n },\n\n remove: function remove(name) {\n this.write(name, '', Date.now() - 86400000);\n }\n };\n })() :\n\n // Non standard browser env (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return {\n write: function write() {},\n read: function read() { return null; },\n remove: function remove() {}\n };\n })()\n);\n","'use strict';\n\n/**\n * Determines whether the specified URL is absolute\n *\n * @param {string} url The URL to test\n * @returns {boolean} True if the specified URL is absolute, otherwise false\n */\nmodule.exports = function isAbsoluteURL(url) {\n // A URL is considered absolute if it begins with \"://\" or \"//\" (protocol-relative URL).\n // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed\n // by any combination of letters, digits, plus, period, or hyphen.\n return /^([a-z][a-z\\d\\+\\-\\.]*:)?\\/\\//i.test(url);\n};\n","'use strict';\n\n/**\n * Creates a new URL by combining the specified URLs\n *\n * @param {string} baseURL The base URL\n * @param {string} relativeURL The relative URL\n * @returns {string} The combined URL\n */\nmodule.exports = function combineURLs(baseURL, relativeURL) {\n return relativeURL\n ? baseURL.replace(/\\/+$/, '') + '/' + relativeURL.replace(/^\\/+/, '')\n : baseURL;\n};\n","'use strict';\n\nvar isAbsoluteURL = require('../helpers/isAbsoluteURL');\nvar combineURLs = require('../helpers/combineURLs');\n\n/**\n * Creates a new URL by combining the baseURL with the requestedURL,\n * only when the requestedURL is not already an absolute URL.\n * If the requestURL is absolute, this function returns the requestedURL untouched.\n *\n * @param {string} baseURL The base URL\n * @param {string} requestedURL Absolute or relative URL to combine\n * @returns {string} The combined full path\n */\nmodule.exports = function buildFullPath(baseURL, requestedURL) {\n if (baseURL && !isAbsoluteURL(requestedURL)) {\n return combineURLs(baseURL, requestedURL);\n }\n return requestedURL;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\n// Headers whose duplicates are ignored by node\n// c.f. https://nodejs.org/api/http.html#http_message_headers\nvar ignoreDuplicateOf = [\n 'age', 'authorization', 'content-length', 'content-type', 'etag',\n 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since',\n 'last-modified', 'location', 'max-forwards', 'proxy-authorization',\n 'referer', 'retry-after', 'user-agent'\n];\n\n/**\n * Parse headers into an object\n *\n * ```\n * Date: Wed, 27 Aug 2014 08:58:49 GMT\n * Content-Type: application/json\n * Connection: keep-alive\n * Transfer-Encoding: chunked\n * ```\n *\n * @param {String} headers Headers needing to be parsed\n * @returns {Object} Headers parsed into an object\n */\nmodule.exports = function parseHeaders(headers) {\n var parsed = {};\n var key;\n var val;\n var i;\n\n if (!headers) { return parsed; }\n\n utils.forEach(headers.split('\\n'), function parser(line) {\n i = line.indexOf(':');\n key = utils.trim(line.substr(0, i)).toLowerCase();\n val = utils.trim(line.substr(i + 1));\n\n if (key) {\n if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) {\n return;\n }\n if (key === 'set-cookie') {\n parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]);\n } else {\n parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val;\n }\n }\n });\n\n return parsed;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs have full support of the APIs needed to test\n // whether the request URL is of the same origin as current location.\n (function standardBrowserEnv() {\n var msie = /(msie|trident)/i.test(navigator.userAgent);\n var urlParsingNode = document.createElement('a');\n var originURL;\n\n /**\n * Parse a URL to discover it's components\n *\n * @param {String} url The URL to be parsed\n * @returns {Object}\n */\n function resolveURL(url) {\n var href = url;\n\n if (msie) {\n // IE needs attribute set twice to normalize properties\n urlParsingNode.setAttribute('href', href);\n href = urlParsingNode.href;\n }\n\n urlParsingNode.setAttribute('href', href);\n\n // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils\n return {\n href: urlParsingNode.href,\n protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '',\n host: urlParsingNode.host,\n search: urlParsingNode.search ? urlParsingNode.search.replace(/^\\?/, '') : '',\n hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '',\n hostname: urlParsingNode.hostname,\n port: urlParsingNode.port,\n pathname: (urlParsingNode.pathname.charAt(0) === '/') ?\n urlParsingNode.pathname :\n '/' + urlParsingNode.pathname\n };\n }\n\n originURL = resolveURL(window.location.href);\n\n /**\n * Determine if a URL shares the same origin as the current location\n *\n * @param {String} requestURL The URL to test\n * @returns {boolean} True if URL shares the same origin, otherwise false\n */\n return function isURLSameOrigin(requestURL) {\n var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL;\n return (parsed.protocol === originURL.protocol &&\n parsed.host === originURL.host);\n };\n })() :\n\n // Non standard browser envs (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return function isURLSameOrigin() {\n return true;\n };\n })()\n);\n","'use strict';\n\nvar utils = require('./../utils');\nvar settle = require('./../core/settle');\nvar cookies = require('./../helpers/cookies');\nvar buildURL = require('./../helpers/buildURL');\nvar buildFullPath = require('../core/buildFullPath');\nvar parseHeaders = require('./../helpers/parseHeaders');\nvar isURLSameOrigin = require('./../helpers/isURLSameOrigin');\nvar createError = require('../core/createError');\n\nmodule.exports = function xhrAdapter(config) {\n return new Promise(function dispatchXhrRequest(resolve, reject) {\n var requestData = config.data;\n var requestHeaders = config.headers;\n\n if (utils.isFormData(requestData)) {\n delete requestHeaders['Content-Type']; // Let the browser set it\n }\n\n var request = new XMLHttpRequest();\n\n // HTTP basic authentication\n if (config.auth) {\n var username = config.auth.username || '';\n var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : '';\n requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password);\n }\n\n var fullPath = buildFullPath(config.baseURL, config.url);\n request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true);\n\n // Set the request timeout in MS\n request.timeout = config.timeout;\n\n // Listen for ready state\n request.onreadystatechange = function handleLoad() {\n if (!request || request.readyState !== 4) {\n return;\n }\n\n // The request errored out and we didn't get a response, this will be\n // handled by onerror instead\n // With one exception: request that using file: protocol, most browsers\n // will return status as 0 even though it's a successful request\n if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) {\n return;\n }\n\n // Prepare the response\n var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null;\n var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response;\n var response = {\n data: responseData,\n status: request.status,\n statusText: request.statusText,\n headers: responseHeaders,\n config: config,\n request: request\n };\n\n settle(resolve, reject, response);\n\n // Clean up request\n request = null;\n };\n\n // Handle browser request cancellation (as opposed to a manual cancellation)\n request.onabort = function handleAbort() {\n if (!request) {\n return;\n }\n\n reject(createError('Request aborted', config, 'ECONNABORTED', request));\n\n // Clean up request\n request = null;\n };\n\n // Handle low level network errors\n request.onerror = function handleError() {\n // Real errors are hidden from us by the browser\n // onerror should only fire if it's a network error\n reject(createError('Network Error', config, null, request));\n\n // Clean up request\n request = null;\n };\n\n // Handle timeout\n request.ontimeout = function handleTimeout() {\n var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded';\n if (config.timeoutErrorMessage) {\n timeoutErrorMessage = config.timeoutErrorMessage;\n }\n reject(createError(timeoutErrorMessage, config, 'ECONNABORTED',\n request));\n\n // Clean up request\n request = null;\n };\n\n // Add xsrf header\n // This is only done if running in a standard browser environment.\n // Specifically not if we're in a web worker, or react-native.\n if (utils.isStandardBrowserEnv()) {\n // Add xsrf header\n var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ?\n cookies.read(config.xsrfCookieName) :\n undefined;\n\n if (xsrfValue) {\n requestHeaders[config.xsrfHeaderName] = xsrfValue;\n }\n }\n\n // Add headers to the request\n if ('setRequestHeader' in request) {\n utils.forEach(requestHeaders, function setRequestHeader(val, key) {\n if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') {\n // Remove Content-Type if data is undefined\n delete requestHeaders[key];\n } else {\n // Otherwise add header to the request\n request.setRequestHeader(key, val);\n }\n });\n }\n\n // Add withCredentials to request if needed\n if (!utils.isUndefined(config.withCredentials)) {\n request.withCredentials = !!config.withCredentials;\n }\n\n // Add responseType to request if needed\n if (config.responseType) {\n try {\n request.responseType = config.responseType;\n } catch (e) {\n // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2.\n // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function.\n if (config.responseType !== 'json') {\n throw e;\n }\n }\n }\n\n // Handle progress if needed\n if (typeof config.onDownloadProgress === 'function') {\n request.addEventListener('progress', config.onDownloadProgress);\n }\n\n // Not all browsers support upload events\n if (typeof config.onUploadProgress === 'function' && request.upload) {\n request.upload.addEventListener('progress', config.onUploadProgress);\n }\n\n if (config.cancelToken) {\n // Handle cancellation\n config.cancelToken.promise.then(function onCanceled(cancel) {\n if (!request) {\n return;\n }\n\n request.abort();\n reject(cancel);\n // Clean up request\n request = null;\n });\n }\n\n if (!requestData) {\n requestData = null;\n }\n\n // Send the request\n request.send(requestData);\n });\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar normalizeHeaderName = require('./helpers/normalizeHeaderName');\n\nvar DEFAULT_CONTENT_TYPE = {\n 'Content-Type': 'application/x-www-form-urlencoded'\n};\n\nfunction setContentTypeIfUnset(headers, value) {\n if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) {\n headers['Content-Type'] = value;\n }\n}\n\nfunction getDefaultAdapter() {\n var adapter;\n if (typeof XMLHttpRequest !== 'undefined') {\n // For browsers use XHR adapter\n adapter = require('./adapters/xhr');\n } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') {\n // For node use HTTP adapter\n adapter = require('./adapters/http');\n }\n return adapter;\n}\n\nvar defaults = {\n adapter: getDefaultAdapter(),\n\n transformRequest: [function transformRequest(data, headers) {\n normalizeHeaderName(headers, 'Accept');\n normalizeHeaderName(headers, 'Content-Type');\n if (utils.isFormData(data) ||\n utils.isArrayBuffer(data) ||\n utils.isBuffer(data) ||\n utils.isStream(data) ||\n utils.isFile(data) ||\n utils.isBlob(data)\n ) {\n return data;\n }\n if (utils.isArrayBufferView(data)) {\n return data.buffer;\n }\n if (utils.isURLSearchParams(data)) {\n setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8');\n return data.toString();\n }\n if (utils.isObject(data)) {\n setContentTypeIfUnset(headers, 'application/json;charset=utf-8');\n return JSON.stringify(data);\n }\n return data;\n }],\n\n transformResponse: [function transformResponse(data) {\n /*eslint no-param-reassign:0*/\n if (typeof data === 'string') {\n try {\n data = JSON.parse(data);\n } catch (e) { /* Ignore */ }\n }\n return data;\n }],\n\n /**\n * A timeout in milliseconds to abort a request. If set to 0 (default) a\n * timeout is not created.\n */\n timeout: 0,\n\n xsrfCookieName: 'XSRF-TOKEN',\n xsrfHeaderName: 'X-XSRF-TOKEN',\n\n maxContentLength: -1,\n maxBodyLength: -1,\n\n validateStatus: function validateStatus(status) {\n return status >= 200 && status < 300;\n }\n};\n\ndefaults.headers = {\n common: {\n 'Accept': 'application/json, text/plain, */*'\n }\n};\n\nutils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) {\n defaults.headers[method] = {};\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE);\n});\n\nmodule.exports = defaults;\n","'use strict';\n\nvar utils = require('./../utils');\nvar transformData = require('./transformData');\nvar isCancel = require('../cancel/isCancel');\nvar defaults = require('../defaults');\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nfunction throwIfCancellationRequested(config) {\n if (config.cancelToken) {\n config.cancelToken.throwIfRequested();\n }\n}\n\n/**\n * Dispatch a request to the server using the configured adapter.\n *\n * @param {object} config The config that is to be used for the request\n * @returns {Promise} The Promise to be fulfilled\n */\nmodule.exports = function dispatchRequest(config) {\n throwIfCancellationRequested(config);\n\n // Ensure headers exist\n config.headers = config.headers || {};\n\n // Transform request data\n config.data = transformData(\n config.data,\n config.headers,\n config.transformRequest\n );\n\n // Flatten headers\n config.headers = utils.merge(\n config.headers.common || {},\n config.headers[config.method] || {},\n config.headers\n );\n\n utils.forEach(\n ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'],\n function cleanHeaderConfig(method) {\n delete config.headers[method];\n }\n );\n\n var adapter = config.adapter || defaults.adapter;\n\n return adapter(config).then(function onAdapterResolution(response) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n response.data = transformData(\n response.data,\n response.headers,\n config.transformResponse\n );\n\n return response;\n }, function onAdapterRejection(reason) {\n if (!isCancel(reason)) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n if (reason && reason.response) {\n reason.response.data = transformData(\n reason.response.data,\n reason.response.headers,\n config.transformResponse\n );\n }\n }\n\n return Promise.reject(reason);\n });\n};\n","'use strict';\n\nvar utils = require('../utils');\n\n/**\n * Config-specific merge-function which creates a new config-object\n * by merging two configuration objects together.\n *\n * @param {Object} config1\n * @param {Object} config2\n * @returns {Object} New object resulting from merging config2 to config1\n */\nmodule.exports = function mergeConfig(config1, config2) {\n // eslint-disable-next-line no-param-reassign\n config2 = config2 || {};\n var config = {};\n\n var valueFromConfig2Keys = ['url', 'method', 'data'];\n var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params'];\n var defaultToConfig2Keys = [\n 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer',\n 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName',\n 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress',\n 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent',\n 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding'\n ];\n var directMergeKeys = ['validateStatus'];\n\n function getMergedValue(target, source) {\n if (utils.isPlainObject(target) && utils.isPlainObject(source)) {\n return utils.merge(target, source);\n } else if (utils.isPlainObject(source)) {\n return utils.merge({}, source);\n } else if (utils.isArray(source)) {\n return source.slice();\n }\n return source;\n }\n\n function mergeDeepProperties(prop) {\n if 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\ No newline at end of file diff --git a/www/webrtc/index.html b/www/webrtc/index.html index fbc4c2d2..81fd23be 100644 --- a/www/webrtc/index.html +++ b/www/webrtc/index.html @@ -1,34 +1,97 @@ - - - - PeerConnection PRANSWER Demo - - - - - -
-

ip_address

- -
- - - - - - - + + + ZLM RTC demo + + + + +
+
+ + + +
+ +
+ +

+ + +

+ +

+ + +

+ + + +
+
+ + + + + + + + \ No newline at end of file diff --git a/www/webrtc/js/adapter.js b/www/webrtc/js/adapter.js deleted file mode 100644 index 2562bfd7..00000000 --- a/www/webrtc/js/adapter.js +++ /dev/null @@ -1,2475 +0,0 @@ -(function(f){if(typeof exports==="object"&&typeof module!=="undefined"){module.exports=f()}else if(typeof define==="function"&&define.amd){define([],f)}else{var g;if(typeof window!=="undefined"){g=window}else if(typeof global!=="undefined"){g=global}else if(typeof self!=="undefined"){g=self}else{g=this}g.adapter = f()}})(function(){var define,module,exports;return (function e(t,n,r){function s(o,u){if(!n[o]){if(!t[o]){var a=typeof require=="function"&&require;if(!u&&a)return a(o,!0);if(i)return i(o,!0);var f=new Error("Cannot find module '"+o+"'");throw f.code="MODULE_NOT_FOUND",f}var l=n[o]={exports:{}};t[o][0].call(l.exports,function(e){var n=t[o][1][e];return s(n?n:e)},l,l.exports,e,t,n,r)}return n[o].exports}var i=typeof require=="function"&&require;for(var o=0;o 0 && typeof selector === 'function') { - return origGetStats(selector, successCallback); - } - - var fixChromeStats_ = function(response) { - var standardReport = {}; - var reports = response.result(); - reports.forEach(function(report) { - var standardStats = { - id: report.id, - timestamp: report.timestamp, - type: report.type - }; - report.names().forEach(function(name) { - standardStats[name] = report.stat(name); - }); - standardReport[standardStats.id] = standardStats; - }); - - return standardReport; - }; - - if (arguments.length >= 2) { - var successCallbackWrapper_ = function(response) { - args[1](fixChromeStats_(response)); - }; - - return origGetStats.apply(this, [successCallbackWrapper_, - arguments[0]]); - } - - // promise-support - return new Promise(function(resolve, reject) { - if (args.length === 1 && typeof selector === 'object') { - origGetStats.apply(self, - [function(response) { - resolve.apply(null, [fixChromeStats_(response)]); - }, reject]); - } else { - origGetStats.apply(self, [resolve, reject]); - } - }); - }; - - return pc; - }; - window.RTCPeerConnection.prototype = webkitRTCPeerConnection.prototype; - - // wrap static methods. Currently just generateCertificate. - if (webkitRTCPeerConnection.generateCertificate) { - Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', { - get: function() { - return webkitRTCPeerConnection.generateCertificate; - } - }); - } - - // add promise support - ['createOffer', 'createAnswer'].forEach(function(method) { - var nativeMethod = webkitRTCPeerConnection.prototype[method]; - webkitRTCPeerConnection.prototype[method] = function() { - var self = this; - if (arguments.length < 1 || (arguments.length === 1 && - typeof(arguments[0]) === 'object')) { - var opts = arguments.length === 1 ? arguments[0] : undefined; - return new Promise(function(resolve, reject) { - nativeMethod.apply(self, [resolve, reject, opts]); - }); - } - return nativeMethod.apply(this, arguments); - }; - }); - - ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] - .forEach(function(method) { - var nativeMethod = webkitRTCPeerConnection.prototype[method]; - webkitRTCPeerConnection.prototype[method] = function() { - var args = arguments; - var self = this; - args[0] = new ((method === 'addIceCandidate')? - RTCIceCandidate : RTCSessionDescription)(args[0]); - return new Promise(function(resolve, reject) { - nativeMethod.apply(self, [args[0], - function() { - resolve(); - if (args.length >= 2) { - args[1].apply(null, []); - } - }, - function(err) { - reject(err); - if (args.length >= 3) { - args[2].apply(null, [err]); - } - }] - ); - }); - }; - }); - }, - - // Attach a media stream to an element. - attachMediaStream: function(element, stream) { - logging('DEPRECATED, attachMediaStream will soon be removed.'); - if (browserDetails.version >= 43) { - element.srcObject = stream; - } else if (typeof element.src !== 'undefined') { - element.src = URL.createObjectURL(stream); - } else { - logging('Error attaching stream to element.'); - } - }, - - reattachMediaStream: function(to, from) { - logging('DEPRECATED, reattachMediaStream will soon be removed.'); - if (browserDetails.version >= 43) { - to.srcObject = from.srcObject; - } else { - to.src = from.src; - } - } -}; - - -// Expose public methods. -module.exports = { - shimOnTrack: chromeShim.shimOnTrack, - shimSourceObject: chromeShim.shimSourceObject, - shimPeerConnection: chromeShim.shimPeerConnection, - shimGetUserMedia: require('./getusermedia'), - attachMediaStream: chromeShim.attachMediaStream, - reattachMediaStream: chromeShim.reattachMediaStream -}; - -},{"../utils.js":9,"./getusermedia":3}],3:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; -var logging = require('../utils.js').log; - -// Expose public methods. -module.exports = function() { - var constraintsToChrome_ = function(c) { - if (typeof c !== 'object' || c.mandatory || c.optional) { - return c; - } - var cc = {}; - Object.keys(c).forEach(function(key) { - if (key === 'require' || key === 'advanced' || key === 'mediaSource') { - return; - } - var r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]}; - if (r.exact !== undefined && typeof r.exact === 'number') { - r.min = r.max = r.exact; - } - var oldname_ = function(prefix, name) { - if (prefix) { - return prefix + name.charAt(0).toUpperCase() + name.slice(1); - } - return (name === 'deviceId') ? 'sourceId' : name; - }; - if (r.ideal !== undefined) { - cc.optional = cc.optional || []; - var oc = {}; - if (typeof r.ideal === 'number') { - oc[oldname_('min', key)] = r.ideal; - cc.optional.push(oc); - oc = {}; - oc[oldname_('max', key)] = r.ideal; - cc.optional.push(oc); - } else { - oc[oldname_('', key)] = r.ideal; - cc.optional.push(oc); - } - } - if (r.exact !== undefined && typeof r.exact !== 'number') { - cc.mandatory = cc.mandatory || {}; - cc.mandatory[oldname_('', key)] = r.exact; - } else { - ['min', 'max'].forEach(function(mix) { - if (r[mix] !== undefined) { - cc.mandatory = cc.mandatory || {}; - cc.mandatory[oldname_(mix, key)] = r[mix]; - } - }); - } - }); - if (c.advanced) { - cc.optional = (cc.optional || []).concat(c.advanced); - } - return cc; - }; - - var getUserMedia_ = function(constraints, onSuccess, onError) { - constraints = JSON.parse(JSON.stringify(constraints)); - if (constraints.audio) { - constraints.audio = constraintsToChrome_(constraints.audio); - } - if (constraints.video) { - constraints.video = constraintsToChrome_(constraints.video); - } - logging('chrome: ' + JSON.stringify(constraints)); - return navigator.webkitGetUserMedia(constraints, onSuccess, onError); - }; - navigator.getUserMedia = getUserMedia_; - - // Returns the result of getUserMedia as a Promise. - var getUserMediaPromise_ = function(constraints) { - return new Promise(function(resolve, reject) { - navigator.getUserMedia(constraints, resolve, reject); - }); - }; - - if (!navigator.mediaDevices) { - navigator.mediaDevices = { - getUserMedia: getUserMediaPromise_, - enumerateDevices: function() { - return new Promise(function(resolve) { - var kinds = {audio: 'audioinput', video: 'videoinput'}; - return MediaStreamTrack.getSources(function(devices) { - resolve(devices.map(function(device) { - return {label: device.label, - kind: kinds[device.kind], - deviceId: device.id, - groupId: ''}; - })); - }); - }); - } - }; - } - - // A shim for getUserMedia method on the mediaDevices object. - // TODO(KaptenJansson) remove once implemented in Chrome stable. - if (!navigator.mediaDevices.getUserMedia) { - navigator.mediaDevices.getUserMedia = function(constraints) { - return getUserMediaPromise_(constraints); - }; - } else { - // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia - // function which returns a Promise, it does not accept spec-style - // constraints. - var origGetUserMedia = navigator.mediaDevices.getUserMedia. - bind(navigator.mediaDevices); - navigator.mediaDevices.getUserMedia = function(c) { - if (c) { - logging('spec: ' + JSON.stringify(c)); // whitespace for alignment - c.audio = constraintsToChrome_(c.audio); - c.video = constraintsToChrome_(c.video); - logging('chrome: ' + JSON.stringify(c)); - } - return origGetUserMedia(c); - }.bind(this); - } - - // Dummy devicechange event methods. - // TODO(KaptenJansson) remove once implemented in Chrome stable. - if (typeof navigator.mediaDevices.addEventListener === 'undefined') { - navigator.mediaDevices.addEventListener = function() { - logging('Dummy mediaDevices.addEventListener called.'); - }; - } - if (typeof navigator.mediaDevices.removeEventListener === 'undefined') { - navigator.mediaDevices.removeEventListener = function() { - logging('Dummy mediaDevices.removeEventListener called.'); - }; - } -}; - -},{"../utils.js":9}],4:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; - -// SDP helpers. -var SDPUtils = {}; - -// Generate an alphanumeric identifier for cname or mids. -// TODO: use UUIDs instead? https://gist.github.com/jed/982883 -SDPUtils.generateIdentifier = function() { - return Math.random().toString(36).substr(2, 10); -}; - -// The RTCP CNAME used by all peerconnections from the same JS. -SDPUtils.localCName = SDPUtils.generateIdentifier(); - -// Splits SDP into lines, dealing with both CRLF and LF. -SDPUtils.splitLines = function(blob) { - return blob.trim().split('\n').map(function(line) { - return line.trim(); - }); -}; -// Splits SDP into sessionpart and mediasections. Ensures CRLF. -SDPUtils.splitSections = function(blob) { - var parts = blob.split('\nm='); - return parts.map(function(part, index) { - return (index > 0 ? 'm=' + part : part).trim() + '\r\n'; - }); -}; - -// Returns lines that start with a certain prefix. -SDPUtils.matchPrefix = function(blob, prefix) { - return SDPUtils.splitLines(blob).filter(function(line) { - return line.indexOf(prefix) === 0; - }); -}; - -// Parses an ICE candidate line. Sample input: -// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8 -// rport 55996" -SDPUtils.parseCandidate = function(line) { - var parts; - // Parse both variants. - if (line.indexOf('a=candidate:') === 0) { - parts = line.substring(12).split(' '); - } else { - parts = line.substring(10).split(' '); - } - - var candidate = { - foundation: parts[0], - component: parts[1], - protocol: parts[2].toLowerCase(), - priority: parseInt(parts[3], 10), - ip: parts[4], - port: parseInt(parts[5], 10), - // skip parts[6] == 'typ' - type: parts[7] - }; - - for (var i = 8; i < parts.length; i += 2) { - switch (parts[i]) { - case 'raddr': - candidate.relatedAddress = parts[i + 1]; - break; - case 'rport': - candidate.relatedPort = parseInt(parts[i + 1], 10); - break; - case 'tcptype': - candidate.tcpType = parts[i + 1]; - break; - default: // Unknown extensions are silently ignored. - break; - } - } - return candidate; -}; - -// Translates a candidate object into SDP candidate attribute. -SDPUtils.writeCandidate = function(candidate) { - var sdp = []; - sdp.push(candidate.foundation); - sdp.push(candidate.component); - sdp.push(candidate.protocol.toUpperCase()); - sdp.push(candidate.priority); - sdp.push(candidate.ip); - sdp.push(candidate.port); - - var type = candidate.type; - sdp.push('typ'); - sdp.push(type); - if (type !== 'host' && candidate.relatedAddress && - candidate.relatedPort) { - sdp.push('raddr'); - sdp.push(candidate.relatedAddress); // was: relAddr - sdp.push('rport'); - sdp.push(candidate.relatedPort); // was: relPort - } - if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') { - sdp.push('tcptype'); - sdp.push(candidate.tcpType); - } - return 'candidate:' + sdp.join(' '); -}; - -// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input: -// a=rtpmap:111 opus/48000/2 -SDPUtils.parseRtpMap = function(line) { - var parts = line.substr(9).split(' '); - var parsed = { - payloadType: parseInt(parts.shift(), 10) // was: id - }; - - parts = parts[0].split('/'); - - parsed.name = parts[0]; - parsed.clockRate = parseInt(parts[1], 10); // was: clockrate - // was: channels - parsed.numChannels = parts.length === 3 ? parseInt(parts[2], 10) : 1; - return parsed; -}; - -// Generate an a=rtpmap line from RTCRtpCodecCapability or -// RTCRtpCodecParameters. -SDPUtils.writeRtpMap = function(codec) { - var pt = codec.payloadType; - if (codec.preferredPayloadType !== undefined) { - pt = codec.preferredPayloadType; - } - return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate + - (codec.numChannels !== 1 ? '/' + codec.numChannels : '') + '\r\n'; -}; - -// Parses an a=extmap line (headerextension from RFC 5285). Sample input: -// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset -SDPUtils.parseExtmap = function(line) { - var parts = line.substr(9).split(' '); - return { - id: parseInt(parts[0], 10), - uri: parts[1] - }; -}; - -// Generates a=extmap line from RTCRtpHeaderExtensionParameters or -// RTCRtpHeaderExtension. -SDPUtils.writeExtmap = function(headerExtension) { - return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) + - ' ' + headerExtension.uri + '\r\n'; -}; - -// Parses an ftmp line, returns dictionary. Sample input: -// a=fmtp:96 vbr=on;cng=on -// Also deals with vbr=on; cng=on -SDPUtils.parseFmtp = function(line) { - var parsed = {}; - var kv; - var parts = line.substr(line.indexOf(' ') + 1).split(';'); - for (var j = 0; j < parts.length; j++) { - kv = parts[j].trim().split('='); - parsed[kv[0].trim()] = kv[1]; - } - return parsed; -}; - -// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters. -SDPUtils.writeFmtp = function(codec) { - var line = ''; - var pt = codec.payloadType; - if (codec.preferredPayloadType !== undefined) { - pt = codec.preferredPayloadType; - } - if (codec.parameters && Object.keys(codec.parameters).length) { - var params = []; - Object.keys(codec.parameters).forEach(function(param) { - params.push(param + '=' + codec.parameters[param]); - }); - line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\r\n'; - } - return line; -}; - -// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input: -// a=rtcp-fb:98 nack rpsi -SDPUtils.parseRtcpFb = function(line) { - var parts = line.substr(line.indexOf(' ') + 1).split(' '); - return { - type: parts.shift(), - parameter: parts.join(' ') - }; -}; -// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters. -SDPUtils.writeRtcpFb = function(codec) { - var lines = ''; - var pt = codec.payloadType; - if (codec.preferredPayloadType !== undefined) { - pt = codec.preferredPayloadType; - } - if (codec.rtcpFeedback && codec.rtcpFeedback.length) { - // FIXME: special handling for trr-int? - codec.rtcpFeedback.forEach(function(fb) { - lines += 'a=rtcp-fb:' + pt + ' ' + fb.type + ' ' + fb.parameter + - '\r\n'; - }); - } - return lines; -}; - -// Parses an RFC 5576 ssrc media attribute. Sample input: -// a=ssrc:3735928559 cname:something -SDPUtils.parseSsrcMedia = function(line) { - var sp = line.indexOf(' '); - var parts = { - ssrc: parseInt(line.substr(7, sp - 7), 10) - }; - var colon = line.indexOf(':', sp); - if (colon > -1) { - parts.attribute = line.substr(sp + 1, colon - sp - 1); - parts.value = line.substr(colon + 1); - } else { - parts.attribute = line.substr(sp + 1); - } - return parts; -}; - -// Extracts DTLS parameters from SDP media section or sessionpart. -// FIXME: for consistency with other functions this should only -// get the fingerprint line as input. See also getIceParameters. -SDPUtils.getDtlsParameters = function(mediaSection, sessionpart) { - var lines = SDPUtils.splitLines(mediaSection); - // Search in session part, too. - lines = lines.concat(SDPUtils.splitLines(sessionpart)); - var fpLine = lines.filter(function(line) { - return line.indexOf('a=fingerprint:') === 0; - })[0].substr(14); - // Note: a=setup line is ignored since we use the 'auto' role. - var dtlsParameters = { - role: 'auto', - fingerprints: [{ - algorithm: fpLine.split(' ')[0], - value: fpLine.split(' ')[1] - }] - }; - return dtlsParameters; -}; - -// Serializes DTLS parameters to SDP. -SDPUtils.writeDtlsParameters = function(params, setupType) { - var sdp = 'a=setup:' + setupType + '\r\n'; - params.fingerprints.forEach(function(fp) { - sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\r\n'; - }); - return sdp; -}; -// Parses ICE information from SDP media section or sessionpart. -// FIXME: for consistency with other functions this should only -// get the ice-ufrag and ice-pwd lines as input. -SDPUtils.getIceParameters = function(mediaSection, sessionpart) { - var lines = SDPUtils.splitLines(mediaSection); - // Search in session part, too. - lines = lines.concat(SDPUtils.splitLines(sessionpart)); - var iceParameters = { - usernameFragment: lines.filter(function(line) { - return line.indexOf('a=ice-ufrag:') === 0; - })[0].substr(12), - password: lines.filter(function(line) { - return line.indexOf('a=ice-pwd:') === 0; - })[0].substr(10) - }; - return iceParameters; -}; - -// Serializes ICE parameters to SDP. -SDPUtils.writeIceParameters = function(params) { - return 'a=ice-ufrag:' + params.usernameFragment + '\r\n' + - 'a=ice-pwd:' + params.password + '\r\n'; -}; - -// Parses the SDP media section and returns RTCRtpParameters. -SDPUtils.parseRtpParameters = function(mediaSection) { - var description = { - codecs: [], - headerExtensions: [], - fecMechanisms: [], - rtcp: [] - }; - var lines = SDPUtils.splitLines(mediaSection); - var mline = lines[0].split(' '); - for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..] - var pt = mline[i]; - var rtpmapline = SDPUtils.matchPrefix( - mediaSection, 'a=rtpmap:' + pt + ' ')[0]; - if (rtpmapline) { - var codec = SDPUtils.parseRtpMap(rtpmapline); - var fmtps = SDPUtils.matchPrefix( - mediaSection, 'a=fmtp:' + pt + ' '); - // Only the first a=fmtp: is considered. - codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {}; - codec.rtcpFeedback = SDPUtils.matchPrefix( - mediaSection, 'a=rtcp-fb:' + pt + ' ') - .map(SDPUtils.parseRtcpFb); - description.codecs.push(codec); - // parse FEC mechanisms from rtpmap lines. - switch (codec.name.toUpperCase()) { - case 'RED': - case 'ULPFEC': - description.fecMechanisms.push(codec.name.toUpperCase()); - break; - default: // only RED and ULPFEC are recognized as FEC mechanisms. - break; - } - } - } - SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) { - description.headerExtensions.push(SDPUtils.parseExtmap(line)); - }); - // FIXME: parse rtcp. - return description; -}; - -// Generates parts of the SDP media section describing the capabilities / -// parameters. -SDPUtils.writeRtpDescription = function(kind, caps) { - var sdp = ''; - - // Build the mline. - sdp += 'm=' + kind + ' '; - sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs. - sdp += ' UDP/TLS/RTP/SAVPF '; - sdp += caps.codecs.map(function(codec) { - if (codec.preferredPayloadType !== undefined) { - return codec.preferredPayloadType; - } - return codec.payloadType; - }).join(' ') + '\r\n'; - - sdp += 'c=IN IP4 0.0.0.0\r\n'; - sdp += 'a=rtcp:9 IN IP4 0.0.0.0\r\n'; - - // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb. - caps.codecs.forEach(function(codec) { - sdp += SDPUtils.writeRtpMap(codec); - sdp += SDPUtils.writeFmtp(codec); - sdp += SDPUtils.writeRtcpFb(codec); - }); - // FIXME: add headerExtensions, fecMechanismş and rtcp. - sdp += 'a=rtcp-mux\r\n'; - return sdp; -}; - -// Parses the SDP media section and returns an array of -// RTCRtpEncodingParameters. -SDPUtils.parseRtpEncodingParameters = function(mediaSection) { - var encodingParameters = []; - var description = SDPUtils.parseRtpParameters(mediaSection); - var hasRed = description.fecMechanisms.indexOf('RED') !== -1; - var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1; - - // filter a=ssrc:... cname:, ignore PlanB-msid - var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') - .map(function(line) { - return SDPUtils.parseSsrcMedia(line); - }) - .filter(function(parts) { - return parts.attribute === 'cname'; - }); - var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc; - var secondarySsrc; - - var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID') - .map(function(line) { - var parts = line.split(' '); - parts.shift(); - return parts.map(function(part) { - return parseInt(part, 10); - }); - }); - if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) { - secondarySsrc = flows[0][1]; - } - - description.codecs.forEach(function(codec) { - if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) { - var encParam = { - ssrc: primarySsrc, - codecPayloadType: parseInt(codec.parameters.apt, 10), - rtx: { - ssrc: secondarySsrc - } - }; - encodingParameters.push(encParam); - if (hasRed) { - encParam = JSON.parse(JSON.stringify(encParam)); - encParam.fec = { - ssrc: secondarySsrc, - mechanism: hasUlpfec ? 'red+ulpfec' : 'red' - }; - encodingParameters.push(encParam); - } - } - }); - if (encodingParameters.length === 0 && primarySsrc) { - encodingParameters.push({ - ssrc: primarySsrc - }); - } - - // we support both b=AS and b=TIAS but interpret AS as TIAS. - var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b='); - if (bandwidth.length) { - if (bandwidth[0].indexOf('b=TIAS:') === 0) { - bandwidth = parseInt(bandwidth[0].substr(7), 10); - } else if (bandwidth[0].indexOf('b=AS:') === 0) { - bandwidth = parseInt(bandwidth[0].substr(5), 10); - } - encodingParameters.forEach(function(params) { - params.maxBitrate = bandwidth; - }); - } - return encodingParameters; -}; - -SDPUtils.writeSessionBoilerplate = function() { - // FIXME: sess-id should be an NTP timestamp. - return 'v=0\r\n' + - 'o=thisisadapterortc 8169639915646943137 2 IN IP4 127.0.0.1\r\n' + - 's=-\r\n' + - 't=0 0\r\n'; -}; - -SDPUtils.writeMediaSection = function(transceiver, caps, type, stream) { - var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps); - - // Map ICE parameters (ufrag, pwd) to SDP. - sdp += SDPUtils.writeIceParameters( - transceiver.iceGatherer.getLocalParameters()); - - // Map DTLS parameters to SDP. - sdp += SDPUtils.writeDtlsParameters( - transceiver.dtlsTransport.getLocalParameters(), - type === 'offer' ? 'actpass' : 'active'); - - sdp += 'a=mid:' + transceiver.mid + '\r\n'; - - if (transceiver.rtpSender && transceiver.rtpReceiver) { - sdp += 'a=sendrecv\r\n'; - } else if (transceiver.rtpSender) { - sdp += 'a=sendonly\r\n'; - } else if (transceiver.rtpReceiver) { - sdp += 'a=recvonly\r\n'; - } else { - sdp += 'a=inactive\r\n'; - } - - // FIXME: for RTX there might be multiple SSRCs. Not implemented in Edge yet. - if (transceiver.rtpSender) { - var msid = 'msid:' + stream.id + ' ' + - transceiver.rtpSender.track.id + '\r\n'; - sdp += 'a=' + msid; - sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + - ' ' + msid; - } - // FIXME: this should be written by writeRtpDescription. - sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc + - ' cname:' + SDPUtils.localCName + '\r\n'; - return sdp; -}; - -// Gets the direction from the mediaSection or the sessionpart. -SDPUtils.getDirection = function(mediaSection, sessionpart) { - // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv. - var lines = SDPUtils.splitLines(mediaSection); - for (var i = 0; i < lines.length; i++) { - switch (lines[i]) { - case 'a=sendrecv': - case 'a=sendonly': - case 'a=recvonly': - case 'a=inactive': - return lines[i].substr(2); - default: - // FIXME: What should happen here? - } - } - if (sessionpart) { - return SDPUtils.getDirection(sessionpart); - } - return 'sendrecv'; -}; - -// Expose public methods. -module.exports = SDPUtils; - -},{}],5:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; - -var SDPUtils = require('./edge_sdp'); -var logging = require('../utils').log; - -var edgeShim = { - shimPeerConnection: function() { - if (window.RTCIceGatherer) { - // ORTC defines an RTCIceCandidate object but no constructor. - // Not implemented in Edge. - if (!window.RTCIceCandidate) { - window.RTCIceCandidate = function(args) { - return args; - }; - } - // ORTC does not have a session description object but - // other browsers (i.e. Chrome) that will support both PC and ORTC - // in the future might have this defined already. - if (!window.RTCSessionDescription) { - window.RTCSessionDescription = function(args) { - return args; - }; - } - } - - window.RTCPeerConnection = function(config) { - var self = this; - - var _eventTarget = document.createDocumentFragment(); - ['addEventListener', 'removeEventListener', 'dispatchEvent'] - .forEach(function(method) { - self[method] = _eventTarget[method].bind(_eventTarget); - }); - - this.onicecandidate = null; - this.onaddstream = null; - this.ontrack = null; - this.onremovestream = null; - this.onsignalingstatechange = null; - this.oniceconnectionstatechange = null; - this.onnegotiationneeded = null; - this.ondatachannel = null; - - this.localStreams = []; - this.remoteStreams = []; - this.getLocalStreams = function() { - return self.localStreams; - }; - this.getRemoteStreams = function() { - return self.remoteStreams; - }; - - this.localDescription = new RTCSessionDescription({ - type: '', - sdp: '' - }); - this.remoteDescription = new RTCSessionDescription({ - type: '', - sdp: '' - }); - this.signalingState = 'stable'; - this.iceConnectionState = 'new'; - this.iceGatheringState = 'new'; - - this.iceOptions = { - gatherPolicy: 'all', - iceServers: [] - }; - if (config && config.iceTransportPolicy) { - switch (config.iceTransportPolicy) { - case 'all': - case 'relay': - this.iceOptions.gatherPolicy = config.iceTransportPolicy; - break; - case 'none': - // FIXME: remove once implementation and spec have added this. - throw new TypeError('iceTransportPolicy "none" not supported'); - default: - // don't set iceTransportPolicy. - break; - } - } - if (config && config.iceServers) { - // Edge does not like - // 1) stun: - // 2) turn: that does not have all of turn:host:port?transport=udp - this.iceOptions.iceServers = config.iceServers.filter(function(server) { - if (server && server.urls) { - server.urls = server.urls.filter(function(url) { - return url.indexOf('turn:') === 0 && - url.indexOf('transport=udp') !== -1; - })[0]; - return !!server.urls; - } - return false; - }); - } - - // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ... - // everything that is needed to describe a SDP m-line. - this.transceivers = []; - - // since the iceGatherer is currently created in createOffer but we - // must not emit candidates until after setLocalDescription we buffer - // them in this array. - this._localIceCandidatesBuffer = []; - }; - - window.RTCPeerConnection.prototype._emitBufferedCandidates = function() { - var self = this; - var sections = SDPUtils.splitSections(self.localDescription.sdp); - // FIXME: need to apply ice candidates in a way which is async but - // in-order - this._localIceCandidatesBuffer.forEach(function(event) { - var end = !event.candidate || Object.keys(event.candidate).length === 0; - if (end) { - for (var j = 1; j < sections.length; j++) { - if (sections[j].indexOf('\r\na=end-of-candidates\r\n') === -1) { - sections[j] += 'a=end-of-candidates\r\n'; - } - } - } else if (event.candidate.candidate.indexOf('typ endOfCandidates') - === -1) { - sections[event.candidate.sdpMLineIndex + 1] += - 'a=' + event.candidate.candidate + '\r\n'; - } - self.localDescription.sdp = sections.join(''); - self.dispatchEvent(event); - if (self.onicecandidate !== null) { - self.onicecandidate(event); - } - if (!event.candidate && self.iceGatheringState !== 'complete') { - var complete = self.transceivers.every(function(transceiver) { - return transceiver.iceGatherer && - transceiver.iceGatherer.state === 'completed'; - }); - if (complete) { - self.iceGatheringState = 'complete'; - } - } - }); - this._localIceCandidatesBuffer = []; - }; - - window.RTCPeerConnection.prototype.addStream = function(stream) { - // Clone is necessary for local demos mostly, attaching directly - // to two different senders does not work (build 10547). - this.localStreams.push(stream.clone()); - this._maybeFireNegotiationNeeded(); - }; - - window.RTCPeerConnection.prototype.removeStream = function(stream) { - var idx = this.localStreams.indexOf(stream); - if (idx > -1) { - this.localStreams.splice(idx, 1); - this._maybeFireNegotiationNeeded(); - } - }; - - // Determines the intersection of local and remote capabilities. - window.RTCPeerConnection.prototype._getCommonCapabilities = - function(localCapabilities, remoteCapabilities) { - var commonCapabilities = { - codecs: [], - headerExtensions: [], - fecMechanisms: [] - }; - localCapabilities.codecs.forEach(function(lCodec) { - for (var i = 0; i < remoteCapabilities.codecs.length; i++) { - var rCodec = remoteCapabilities.codecs[i]; - if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() && - lCodec.clockRate === rCodec.clockRate && - lCodec.numChannels === rCodec.numChannels) { - // push rCodec so we reply with offerer payload type - commonCapabilities.codecs.push(rCodec); - - // FIXME: also need to determine intersection between - // .rtcpFeedback and .parameters - break; - } - } - }); - - localCapabilities.headerExtensions - .forEach(function(lHeaderExtension) { - for (var i = 0; i < remoteCapabilities.headerExtensions.length; - i++) { - var rHeaderExtension = remoteCapabilities.headerExtensions[i]; - if (lHeaderExtension.uri === rHeaderExtension.uri) { - commonCapabilities.headerExtensions.push(rHeaderExtension); - break; - } - } - }); - - // FIXME: fecMechanisms - return commonCapabilities; - }; - - // Create ICE gatherer, ICE transport and DTLS transport. - window.RTCPeerConnection.prototype._createIceAndDtlsTransports = - function(mid, sdpMLineIndex) { - var self = this; - var iceGatherer = new RTCIceGatherer(self.iceOptions); - var iceTransport = new RTCIceTransport(iceGatherer); - iceGatherer.onlocalcandidate = function(evt) { - var event = new Event('icecandidate'); - event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex}; - - var cand = evt.candidate; - var end = !cand || Object.keys(cand).length === 0; - // Edge emits an empty object for RTCIceCandidateComplete‥ - if (end) { - // polyfill since RTCIceGatherer.state is not implemented in - // Edge 10547 yet. - if (iceGatherer.state === undefined) { - iceGatherer.state = 'completed'; - } - - // Emit a candidate with type endOfCandidates to make the samples - // work. Edge requires addIceCandidate with this empty candidate - // to start checking. The real solution is to signal - // end-of-candidates to the other side when getting the null - // candidate but some apps (like the samples) don't do that. - event.candidate.candidate = - 'candidate:1 1 udp 1 0.0.0.0 9 typ endOfCandidates'; - } else { - // RTCIceCandidate doesn't have a component, needs to be added - cand.component = iceTransport.component === 'RTCP' ? 2 : 1; - event.candidate.candidate = SDPUtils.writeCandidate(cand); - } - - var complete = self.transceivers.every(function(transceiver) { - return transceiver.iceGatherer && - transceiver.iceGatherer.state === 'completed'; - }); - - // Emit candidate if localDescription is set. - // Also emits null candidate when all gatherers are complete. - switch (self.iceGatheringState) { - case 'new': - self._localIceCandidatesBuffer.push(event); - if (end && complete) { - self._localIceCandidatesBuffer.push( - new Event('icecandidate')); - } - break; - case 'gathering': - self._emitBufferedCandidates(); - self.dispatchEvent(event); - if (self.onicecandidate !== null) { - self.onicecandidate(event); - } - if (complete) { - self.dispatchEvent(new Event('icecandidate')); - if (self.onicecandidate !== null) { - self.onicecandidate(new Event('icecandidate')); - } - self.iceGatheringState = 'complete'; - } - break; - case 'complete': - // should not happen... currently! - break; - default: // no-op. - break; - } - }; - iceTransport.onicestatechange = function() { - self._updateConnectionState(); - }; - - var dtlsTransport = new RTCDtlsTransport(iceTransport); - dtlsTransport.ondtlsstatechange = function() { - self._updateConnectionState(); - }; - dtlsTransport.onerror = function() { - // onerror does not set state to failed by itself. - dtlsTransport.state = 'failed'; - self._updateConnectionState(); - }; - - return { - iceGatherer: iceGatherer, - iceTransport: iceTransport, - dtlsTransport: dtlsTransport - }; - }; - - // Start the RTP Sender and Receiver for a transceiver. - window.RTCPeerConnection.prototype._transceive = function(transceiver, - send, recv) { - var params = this._getCommonCapabilities(transceiver.localCapabilities, - transceiver.remoteCapabilities); - if (send && transceiver.rtpSender) { - params.encodings = transceiver.sendEncodingParameters; - params.rtcp = { - cname: SDPUtils.localCName - }; - if (transceiver.recvEncodingParameters.length) { - params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc; - } - transceiver.rtpSender.send(params); - } - if (recv && transceiver.rtpReceiver) { - params.encodings = transceiver.recvEncodingParameters; - params.rtcp = { - cname: transceiver.cname - }; - if (transceiver.sendEncodingParameters.length) { - params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc; - } - transceiver.rtpReceiver.receive(params); - } - }; - - window.RTCPeerConnection.prototype.setLocalDescription = - function(description) { - var self = this; - var sections; - var sessionpart; - if (description.type === 'offer') { - // FIXME: What was the purpose of this empty if statement? - // if (!this._pendingOffer) { - // } else { - if (this._pendingOffer) { - // VERY limited support for SDP munging. Limited to: - // * changing the order of codecs - sections = SDPUtils.splitSections(description.sdp); - sessionpart = sections.shift(); - sections.forEach(function(mediaSection, sdpMLineIndex) { - var caps = SDPUtils.parseRtpParameters(mediaSection); - self._pendingOffer[sdpMLineIndex].localCapabilities = caps; - }); - this.transceivers = this._pendingOffer; - delete this._pendingOffer; - } - } else if (description.type === 'answer') { - sections = SDPUtils.splitSections(self.remoteDescription.sdp); - sessionpart = sections.shift(); - sections.forEach(function(mediaSection, sdpMLineIndex) { - var transceiver = self.transceivers[sdpMLineIndex]; - var iceGatherer = transceiver.iceGatherer; - var iceTransport = transceiver.iceTransport; - var dtlsTransport = transceiver.dtlsTransport; - var localCapabilities = transceiver.localCapabilities; - var remoteCapabilities = transceiver.remoteCapabilities; - var rejected = mediaSection.split('\n', 1)[0] - .split(' ', 2)[1] === '0'; - - if (!rejected) { - var remoteIceParameters = SDPUtils.getIceParameters( - mediaSection, sessionpart); - iceTransport.start(iceGatherer, remoteIceParameters, - 'controlled'); - - var remoteDtlsParameters = SDPUtils.getDtlsParameters( - mediaSection, sessionpart); - dtlsTransport.start(remoteDtlsParameters); - - // Calculate intersection of capabilities. - var params = self._getCommonCapabilities(localCapabilities, - remoteCapabilities); - - // Start the RTCRtpSender. The RTCRtpReceiver for this - // transceiver has already been started in setRemoteDescription. - self._transceive(transceiver, - params.codecs.length > 0, - false); - } - }); - } - - this.localDescription = { - type: description.type, - sdp: description.sdp - }; - switch (description.type) { - case 'offer': - this._updateSignalingState('have-local-offer'); - break; - case 'answer': - this._updateSignalingState('stable'); - break; - default: - throw new TypeError('unsupported type "' + description.type + - '"'); - } - - // If a success callback was provided, emit ICE candidates after it - // has been executed. Otherwise, emit callback after the Promise is - // resolved. - var hasCallback = arguments.length > 1 && - typeof arguments[1] === 'function'; - if (hasCallback) { - var cb = arguments[1]; - window.setTimeout(function() { - cb(); - if (self.iceGatheringState === 'new') { - self.iceGatheringState = 'gathering'; - } - self._emitBufferedCandidates(); - }, 0); - } - var p = Promise.resolve(); - p.then(function() { - if (!hasCallback) { - if (self.iceGatheringState === 'new') { - self.iceGatheringState = 'gathering'; - } - // Usually candidates will be emitted earlier. - window.setTimeout(self._emitBufferedCandidates.bind(self), 500); - } - }); - return p; - }; - - window.RTCPeerConnection.prototype.setRemoteDescription = - function(description) { - var self = this; - var stream = new MediaStream(); - var receiverList = []; - var sections = SDPUtils.splitSections(description.sdp); - var sessionpart = sections.shift(); - sections.forEach(function(mediaSection, sdpMLineIndex) { - var lines = SDPUtils.splitLines(mediaSection); - var mline = lines[0].substr(2).split(' '); - var kind = mline[0]; - var rejected = mline[1] === '0'; - var direction = SDPUtils.getDirection(mediaSection, sessionpart); - - var transceiver; - var iceGatherer; - var iceTransport; - var dtlsTransport; - var rtpSender; - var rtpReceiver; - var sendEncodingParameters; - var recvEncodingParameters; - var localCapabilities; - - var track; - // FIXME: ensure the mediaSection has rtcp-mux set. - var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection); - var remoteIceParameters; - var remoteDtlsParameters; - if (!rejected) { - remoteIceParameters = SDPUtils.getIceParameters(mediaSection, - sessionpart); - remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection, - sessionpart); - } - recvEncodingParameters = - SDPUtils.parseRtpEncodingParameters(mediaSection); - - var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:'); - if (mid.length) { - mid = mid[0].substr(6); - } else { - mid = SDPUtils.generateIdentifier(); - } - - var cname; - // Gets the first SSRC. Note that with RTX there might be multiple - // SSRCs. - var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:') - .map(function(line) { - return SDPUtils.parseSsrcMedia(line); - }) - .filter(function(obj) { - return obj.attribute === 'cname'; - })[0]; - if (remoteSsrc) { - cname = remoteSsrc.value; - } - - var isComplete = SDPUtils.matchPrefix(mediaSection, - 'a=end-of-candidates').length > 0; - var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:') - .map(function(cand) { - return SDPUtils.parseCandidate(cand); - }) - .filter(function(cand) { - return cand.component === '1'; - }); - if (description.type === 'offer' && !rejected) { - var transports = self._createIceAndDtlsTransports(mid, - sdpMLineIndex); - if (isComplete) { - transports.iceTransport.setRemoteCandidates(cands); - } - - localCapabilities = RTCRtpReceiver.getCapabilities(kind); - sendEncodingParameters = [{ - ssrc: (2 * sdpMLineIndex + 2) * 1001 - }]; - - rtpReceiver = new RTCRtpReceiver(transports.dtlsTransport, kind); - - track = rtpReceiver.track; - receiverList.push([track, rtpReceiver]); - // FIXME: not correct when there are multiple streams but that is - // not currently supported in this shim. - stream.addTrack(track); - - // FIXME: look at direction. - if (self.localStreams.length > 0 && - self.localStreams[0].getTracks().length >= sdpMLineIndex) { - // FIXME: actually more complicated, needs to match types etc - var localtrack = self.localStreams[0] - .getTracks()[sdpMLineIndex]; - rtpSender = new RTCRtpSender(localtrack, - transports.dtlsTransport); - } - - self.transceivers[sdpMLineIndex] = { - iceGatherer: transports.iceGatherer, - iceTransport: transports.iceTransport, - dtlsTransport: transports.dtlsTransport, - localCapabilities: localCapabilities, - remoteCapabilities: remoteCapabilities, - rtpSender: rtpSender, - rtpReceiver: rtpReceiver, - kind: kind, - mid: mid, - cname: cname, - sendEncodingParameters: sendEncodingParameters, - recvEncodingParameters: recvEncodingParameters - }; - // Start the RTCRtpReceiver now. The RTPSender is started in - // setLocalDescription. - self._transceive(self.transceivers[sdpMLineIndex], - false, - direction === 'sendrecv' || direction === 'sendonly'); - } else if (description.type === 'answer' && !rejected) { - transceiver = self.transceivers[sdpMLineIndex]; - iceGatherer = transceiver.iceGatherer; - iceTransport = transceiver.iceTransport; - dtlsTransport = transceiver.dtlsTransport; - rtpSender = transceiver.rtpSender; - rtpReceiver = transceiver.rtpReceiver; - sendEncodingParameters = transceiver.sendEncodingParameters; - localCapabilities = transceiver.localCapabilities; - - self.transceivers[sdpMLineIndex].recvEncodingParameters = - recvEncodingParameters; - self.transceivers[sdpMLineIndex].remoteCapabilities = - remoteCapabilities; - self.transceivers[sdpMLineIndex].cname = cname; - - if (isComplete) { - iceTransport.setRemoteCandidates(cands); - } - iceTransport.start(iceGatherer, remoteIceParameters, - 'controlling'); - dtlsTransport.start(remoteDtlsParameters); - - self._transceive(transceiver, - direction === 'sendrecv' || direction === 'recvonly', - direction === 'sendrecv' || direction === 'sendonly'); - - if (rtpReceiver && - (direction === 'sendrecv' || direction === 'sendonly')) { - track = rtpReceiver.track; - receiverList.push([track, rtpReceiver]); - stream.addTrack(track); - } else { - // FIXME: actually the receiver should be created later. - delete transceiver.rtpReceiver; - } - } - }); - - this.remoteDescription = { - type: description.type, - sdp: description.sdp - }; - switch (description.type) { - case 'offer': - this._updateSignalingState('have-remote-offer'); - break; - case 'answer': - this._updateSignalingState('stable'); - break; - default: - throw new TypeError('unsupported type "' + description.type + - '"'); - } - if (stream.getTracks().length) { - self.remoteStreams.push(stream); - window.setTimeout(function() { - var event = new Event('addstream'); - event.stream = stream; - self.dispatchEvent(event); - if (self.onaddstream !== null) { - window.setTimeout(function() { - self.onaddstream(event); - }, 0); - } - - receiverList.forEach(function(item) { - var track = item[0]; - var receiver = item[1]; - var trackEvent = new Event('track'); - trackEvent.track = track; - trackEvent.receiver = receiver; - trackEvent.streams = [stream]; - self.dispatchEvent(event); - if (self.ontrack !== null) { - window.setTimeout(function() { - self.ontrack(trackEvent); - }, 0); - } - }); - }, 0); - } - if (arguments.length > 1 && typeof arguments[1] === 'function') { - window.setTimeout(arguments[1], 0); - } - return Promise.resolve(); - }; - - window.RTCPeerConnection.prototype.close = function() { - this.transceivers.forEach(function(transceiver) { - /* not yet - if (transceiver.iceGatherer) { - transceiver.iceGatherer.close(); - } - */ - if (transceiver.iceTransport) { - transceiver.iceTransport.stop(); - } - if (transceiver.dtlsTransport) { - transceiver.dtlsTransport.stop(); - } - if (transceiver.rtpSender) { - transceiver.rtpSender.stop(); - } - if (transceiver.rtpReceiver) { - transceiver.rtpReceiver.stop(); - } - }); - // FIXME: clean up tracks, local streams, remote streams, etc - this._updateSignalingState('closed'); - }; - - // Update the signaling state. - window.RTCPeerConnection.prototype._updateSignalingState = - function(newState) { - this.signalingState = newState; - var event = new Event('signalingstatechange'); - this.dispatchEvent(event); - if (this.onsignalingstatechange !== null) { - this.onsignalingstatechange(event); - } - }; - - // Determine whether to fire the negotiationneeded event. - window.RTCPeerConnection.prototype._maybeFireNegotiationNeeded = - function() { - // Fire away (for now). - var event = new Event('negotiationneeded'); - this.dispatchEvent(event); - if (this.onnegotiationneeded !== null) { - this.onnegotiationneeded(event); - } - }; - - // Update the connection state. - window.RTCPeerConnection.prototype._updateConnectionState = function() { - var self = this; - var newState; - var states = { - 'new': 0, - closed: 0, - connecting: 0, - checking: 0, - connected: 0, - completed: 0, - failed: 0 - }; - this.transceivers.forEach(function(transceiver) { - states[transceiver.iceTransport.state]++; - states[transceiver.dtlsTransport.state]++; - }); - // ICETransport.completed and connected are the same for this purpose. - states.connected += states.completed; - - newState = 'new'; - if (states.failed > 0) { - newState = 'failed'; - } else if (states.connecting > 0 || states.checking > 0) { - newState = 'connecting'; - } else if (states.disconnected > 0) { - newState = 'disconnected'; - } else if (states.new > 0) { - newState = 'new'; - } else if (states.connected > 0 || states.completed > 0) { - newState = 'connected'; - } - - if (newState !== self.iceConnectionState) { - self.iceConnectionState = newState; - var event = new Event('iceconnectionstatechange'); - this.dispatchEvent(event); - if (this.oniceconnectionstatechange !== null) { - this.oniceconnectionstatechange(event); - } - } - }; - - window.RTCPeerConnection.prototype.createOffer = function() { - var self = this; - if (this._pendingOffer) { - throw new Error('createOffer called while there is a pending offer.'); - } - var offerOptions; - if (arguments.length === 1 && typeof arguments[0] !== 'function') { - offerOptions = arguments[0]; - } else if (arguments.length === 3) { - offerOptions = arguments[2]; - } - - var tracks = []; - var numAudioTracks = 0; - var numVideoTracks = 0; - // Default to sendrecv. - if (this.localStreams.length) { - numAudioTracks = this.localStreams[0].getAudioTracks().length; - numVideoTracks = this.localStreams[0].getVideoTracks().length; - } - // Determine number of audio and video tracks we need to send/recv. - if (offerOptions) { - // Reject Chrome legacy constraints. - if (offerOptions.mandatory || offerOptions.optional) { - throw new TypeError( - 'Legacy mandatory/optional constraints not supported.'); - } - if (offerOptions.offerToReceiveAudio !== undefined) { - numAudioTracks = offerOptions.offerToReceiveAudio; - } - if (offerOptions.offerToReceiveVideo !== undefined) { - numVideoTracks = offerOptions.offerToReceiveVideo; - } - } - if (this.localStreams.length) { - // Push local streams. - this.localStreams[0].getTracks().forEach(function(track) { - tracks.push({ - kind: track.kind, - track: track, - wantReceive: track.kind === 'audio' ? - numAudioTracks > 0 : numVideoTracks > 0 - }); - if (track.kind === 'audio') { - numAudioTracks--; - } else if (track.kind === 'video') { - numVideoTracks--; - } - }); - } - // Create M-lines for recvonly streams. - while (numAudioTracks > 0 || numVideoTracks > 0) { - if (numAudioTracks > 0) { - tracks.push({ - kind: 'audio', - wantReceive: true - }); - numAudioTracks--; - } - if (numVideoTracks > 0) { - tracks.push({ - kind: 'video', - wantReceive: true - }); - numVideoTracks--; - } - } - - var sdp = SDPUtils.writeSessionBoilerplate(); - var transceivers = []; - tracks.forEach(function(mline, sdpMLineIndex) { - // For each track, create an ice gatherer, ice transport, - // dtls transport, potentially rtpsender and rtpreceiver. - var track = mline.track; - var kind = mline.kind; - var mid = SDPUtils.generateIdentifier(); - - var transports = self._createIceAndDtlsTransports(mid, sdpMLineIndex); - - var localCapabilities = RTCRtpSender.getCapabilities(kind); - var rtpSender; - var rtpReceiver; - - // generate an ssrc now, to be used later in rtpSender.send - var sendEncodingParameters = [{ - ssrc: (2 * sdpMLineIndex + 1) * 1001 - }]; - if (track) { - rtpSender = new RTCRtpSender(track, transports.dtlsTransport); - } - - if (mline.wantReceive) { - rtpReceiver = new RTCRtpReceiver(transports.dtlsTransport, kind); - } - - transceivers[sdpMLineIndex] = { - iceGatherer: transports.iceGatherer, - iceTransport: transports.iceTransport, - dtlsTransport: transports.dtlsTransport, - localCapabilities: localCapabilities, - remoteCapabilities: null, - rtpSender: rtpSender, - rtpReceiver: rtpReceiver, - kind: kind, - mid: mid, - sendEncodingParameters: sendEncodingParameters, - recvEncodingParameters: null - }; - var transceiver = transceivers[sdpMLineIndex]; - sdp += SDPUtils.writeMediaSection(transceiver, - transceiver.localCapabilities, 'offer', self.localStreams[0]); - }); - - this._pendingOffer = transceivers; - var desc = new RTCSessionDescription({ - type: 'offer', - sdp: sdp - }); - if (arguments.length && typeof arguments[0] === 'function') { - window.setTimeout(arguments[0], 0, desc); - } - return Promise.resolve(desc); - }; - - window.RTCPeerConnection.prototype.createAnswer = function() { - var self = this; - - var sdp = SDPUtils.writeSessionBoilerplate(); - this.transceivers.forEach(function(transceiver) { - // Calculate intersection of capabilities. - var commonCapabilities = self._getCommonCapabilities( - transceiver.localCapabilities, - transceiver.remoteCapabilities); - - sdp += SDPUtils.writeMediaSection(transceiver, commonCapabilities, - 'answer', self.localStreams[0]); - }); - - var desc = new RTCSessionDescription({ - type: 'answer', - sdp: sdp - }); - if (arguments.length && typeof arguments[0] === 'function') { - window.setTimeout(arguments[0], 0, desc); - } - return Promise.resolve(desc); - }; - - window.RTCPeerConnection.prototype.addIceCandidate = function(candidate) { - var mLineIndex = candidate.sdpMLineIndex; - if (candidate.sdpMid) { - for (var i = 0; i < this.transceivers.length; i++) { - if (this.transceivers[i].mid === candidate.sdpMid) { - mLineIndex = i; - break; - } - } - } - var transceiver = this.transceivers[mLineIndex]; - if (transceiver) { - var cand = Object.keys(candidate.candidate).length > 0 ? - SDPUtils.parseCandidate(candidate.candidate) : {}; - // Ignore Chrome's invalid candidates since Edge does not like them. - if (cand.protocol === 'tcp' && cand.port === 0) { - return; - } - // Ignore RTCP candidates, we assume RTCP-MUX. - if (cand.component !== '1') { - return; - } - // A dirty hack to make samples work. - if (cand.type === 'endOfCandidates') { - cand = {}; - } - transceiver.iceTransport.addRemoteCandidate(cand); - - // update the remoteDescription. - var sections = SDPUtils.splitSections(this.remoteDescription.sdp); - sections[mLineIndex + 1] += (cand.type ? candidate.candidate.trim() - : 'a=end-of-candidates') + '\r\n'; - this.remoteDescription.sdp = sections.join(''); - } - if (arguments.length > 1 && typeof arguments[1] === 'function') { - window.setTimeout(arguments[1], 0); - } - return Promise.resolve(); - }; - - window.RTCPeerConnection.prototype.getStats = function() { - var promises = []; - this.transceivers.forEach(function(transceiver) { - ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport', - 'dtlsTransport'].forEach(function(method) { - if (transceiver[method]) { - promises.push(transceiver[method].getStats()); - } - }); - }); - var cb = arguments.length > 1 && typeof arguments[1] === 'function' && - arguments[1]; - return new Promise(function(resolve) { - var results = {}; - Promise.all(promises).then(function(res) { - res.forEach(function(result) { - Object.keys(result).forEach(function(id) { - results[id] = result[id]; - }); - }); - if (cb) { - window.setTimeout(cb, 0, results); - } - resolve(results); - }); - }); - }; - }, - - // Attach a media stream to an element. - attachMediaStream: function(element, stream) { - logging('DEPRECATED, attachMediaStream will soon be removed.'); - element.srcObject = stream; - }, - - reattachMediaStream: function(to, from) { - logging('DEPRECATED, reattachMediaStream will soon be removed.'); - to.srcObject = from.srcObject; - } -}; - -// Expose public methods. -module.exports = { - shimPeerConnection: edgeShim.shimPeerConnection, - attachMediaStream: edgeShim.attachMediaStream, - reattachMediaStream: edgeShim.reattachMediaStream -}; - -},{"../utils":9,"./edge_sdp":4}],6:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; - -var logging = require('../utils').log; -var browserDetails = require('../utils').browserDetails; - -var firefoxShim = { - shimOnTrack: function() { - if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in - window.RTCPeerConnection.prototype)) { - Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', { - get: function() { - return this._ontrack; - }, - set: function(f) { - if (this._ontrack) { - this.removeEventListener('track', this._ontrack); - this.removeEventListener('addstream', this._ontrackpoly); - } - this.addEventListener('track', this._ontrack = f); - this.addEventListener('addstream', this._ontrackpoly = function(e) { - e.stream.getTracks().forEach(function(track) { - var event = new Event('track'); - event.track = track; - event.receiver = {track: track}; - event.streams = [e.stream]; - this.dispatchEvent(event); - }.bind(this)); - }.bind(this)); - } - }); - } - }, - - shimSourceObject: function() { - // Firefox has supported mozSrcObject since FF22, unprefixed in 42. - if (typeof window === 'object') { - if (window.HTMLMediaElement && - !('srcObject' in window.HTMLMediaElement.prototype)) { - // Shim the srcObject property, once, when HTMLMediaElement is found. - Object.defineProperty(window.HTMLMediaElement.prototype, 'srcObject', { - get: function() { - return this.mozSrcObject; - }, - set: function(stream) { - this.mozSrcObject = stream; - } - }); - } - } - }, - - shimPeerConnection: function() { - // The RTCPeerConnection object. - if (!window.RTCPeerConnection) { - window.RTCPeerConnection = function(pcConfig, pcConstraints) { - if (browserDetails.version < 38) { - // .urls is not supported in FF < 38. - // create RTCIceServers with a single url. - if (pcConfig && pcConfig.iceServers) { - var newIceServers = []; - for (var i = 0; i < pcConfig.iceServers.length; i++) { - var server = pcConfig.iceServers[i]; - if (server.hasOwnProperty('urls')) { - for (var j = 0; j < server.urls.length; j++) { - var newServer = { - url: server.urls[j] - }; - if (server.urls[j].indexOf('turn') === 0) { - newServer.username = server.username; - newServer.credential = server.credential; - } - newIceServers.push(newServer); - } - } else { - newIceServers.push(pcConfig.iceServers[i]); - } - } - pcConfig.iceServers = newIceServers; - } - } - return new mozRTCPeerConnection(pcConfig, pcConstraints); - }; - window.RTCPeerConnection.prototype = mozRTCPeerConnection.prototype; - - // wrap static methods. Currently just generateCertificate. - if (mozRTCPeerConnection.generateCertificate) { - Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', { - get: function() { - return mozRTCPeerConnection.generateCertificate; - } - }); - } - - window.RTCSessionDescription = mozRTCSessionDescription; - window.RTCIceCandidate = mozRTCIceCandidate; - } - - // shim away need for obsolete RTCIceCandidate/RTCSessionDescription. - ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'] - .forEach(function(method) { - var nativeMethod = RTCPeerConnection.prototype[method]; - RTCPeerConnection.prototype[method] = function() { - arguments[0] = new ((method === 'addIceCandidate')? - RTCIceCandidate : RTCSessionDescription)(arguments[0]); - return nativeMethod.apply(this, arguments); - }; - }); - }, - - shimGetUserMedia: function() { - // getUserMedia constraints shim. - var getUserMedia_ = function(constraints, onSuccess, onError) { - var constraintsToFF37_ = function(c) { - if (typeof c !== 'object' || c.require) { - return c; - } - var require = []; - Object.keys(c).forEach(function(key) { - if (key === 'require' || key === 'advanced' || - key === 'mediaSource') { - return; - } - var r = c[key] = (typeof c[key] === 'object') ? - c[key] : {ideal: c[key]}; - if (r.min !== undefined || - r.max !== undefined || r.exact !== undefined) { - require.push(key); - } - if (r.exact !== undefined) { - if (typeof r.exact === 'number') { - r. min = r.max = r.exact; - } else { - c[key] = r.exact; - } - delete r.exact; - } - if (r.ideal !== undefined) { - c.advanced = c.advanced || []; - var oc = {}; - if (typeof r.ideal === 'number') { - oc[key] = {min: r.ideal, max: r.ideal}; - } else { - oc[key] = r.ideal; - } - c.advanced.push(oc); - delete r.ideal; - if (!Object.keys(r).length) { - delete c[key]; - } - } - }); - if (require.length) { - c.require = require; - } - return c; - }; - constraints = JSON.parse(JSON.stringify(constraints)); - if (browserDetails.version < 38) { - logging('spec: ' + JSON.stringify(constraints)); - if (constraints.audio) { - constraints.audio = constraintsToFF37_(constraints.audio); - } - if (constraints.video) { - constraints.video = constraintsToFF37_(constraints.video); - } - logging('ff37: ' + JSON.stringify(constraints)); - } - return navigator.mozGetUserMedia(constraints, onSuccess, onError); - }; - - navigator.getUserMedia = getUserMedia_; - - // Returns the result of getUserMedia as a Promise. - var getUserMediaPromise_ = function(constraints) { - return new Promise(function(resolve, reject) { - navigator.getUserMedia(constraints, resolve, reject); - }); - }; - - // Shim for mediaDevices on older versions. - if (!navigator.mediaDevices) { - navigator.mediaDevices = {getUserMedia: getUserMediaPromise_, - addEventListener: function() { }, - removeEventListener: function() { } - }; - } - navigator.mediaDevices.enumerateDevices = - navigator.mediaDevices.enumerateDevices || function() { - return new Promise(function(resolve) { - var infos = [ - {kind: 'audioinput', deviceId: 'default', label: '', groupId: ''}, - {kind: 'videoinput', deviceId: 'default', label: '', groupId: ''} - ]; - resolve(infos); - }); - }; - - if (browserDetails.version < 41) { - // Work around http://bugzil.la/1169665 - var orgEnumerateDevices = - navigator.mediaDevices.enumerateDevices.bind(navigator.mediaDevices); - navigator.mediaDevices.enumerateDevices = function() { - return orgEnumerateDevices().then(undefined, function(e) { - if (e.name === 'NotFoundError') { - return []; - } - throw e; - }); - }; - } - }, - - // Attach a media stream to an element. - attachMediaStream: function(element, stream) { - logging('DEPRECATED, attachMediaStream will soon be removed.'); - element.srcObject = stream; - }, - - reattachMediaStream: function(to, from) { - logging('DEPRECATED, reattachMediaStream will soon be removed.'); - to.srcObject = from.srcObject; - } -}; - -// Expose public methods. -module.exports = { - shimOnTrack: firefoxShim.shimOnTrack, - shimSourceObject: firefoxShim.shimSourceObject, - shimPeerConnection: firefoxShim.shimPeerConnection, - shimGetUserMedia: require('./getusermedia'), - attachMediaStream: firefoxShim.attachMediaStream, - reattachMediaStream: firefoxShim.reattachMediaStream -}; - -},{"../utils":9,"./getusermedia":7}],7:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; - -var logging = require('../utils').log; -var browserDetails = require('../utils').browserDetails; - -// Expose public methods. -module.exports = function() { - // getUserMedia constraints shim. - var getUserMedia_ = function(constraints, onSuccess, onError) { - var constraintsToFF37_ = function(c) { - if (typeof c !== 'object' || c.require) { - return c; - } - var require = []; - Object.keys(c).forEach(function(key) { - if (key === 'require' || key === 'advanced' || key === 'mediaSource') { - return; - } - var r = c[key] = (typeof c[key] === 'object') ? - c[key] : {ideal: c[key]}; - if (r.min !== undefined || - r.max !== undefined || r.exact !== undefined) { - require.push(key); - } - if (r.exact !== undefined) { - if (typeof r.exact === 'number') { - r. min = r.max = r.exact; - } else { - c[key] = r.exact; - } - delete r.exact; - } - if (r.ideal !== undefined) { - c.advanced = c.advanced || []; - var oc = {}; - if (typeof r.ideal === 'number') { - oc[key] = {min: r.ideal, max: r.ideal}; - } else { - oc[key] = r.ideal; - } - c.advanced.push(oc); - delete r.ideal; - if (!Object.keys(r).length) { - delete c[key]; - } - } - }); - if (require.length) { - c.require = require; - } - return c; - }; - constraints = JSON.parse(JSON.stringify(constraints)); - if (browserDetails.version < 38) { - logging('spec: ' + JSON.stringify(constraints)); - if (constraints.audio) { - constraints.audio = constraintsToFF37_(constraints.audio); - } - if (constraints.video) { - constraints.video = constraintsToFF37_(constraints.video); - } - logging('ff37: ' + JSON.stringify(constraints)); - } - return navigator.mozGetUserMedia(constraints, onSuccess, onError); - }; - - navigator.getUserMedia = getUserMedia_; - - // Returns the result of getUserMedia as a Promise. - var getUserMediaPromise_ = function(constraints) { - return new Promise(function(resolve, reject) { - navigator.getUserMedia(constraints, resolve, reject); - }); - }; - - // Shim for mediaDevices on older versions. - if (!navigator.mediaDevices) { - navigator.mediaDevices = {getUserMedia: getUserMediaPromise_, - addEventListener: function() { }, - removeEventListener: function() { } - }; - } - navigator.mediaDevices.enumerateDevices = - navigator.mediaDevices.enumerateDevices || function() { - return new Promise(function(resolve) { - var infos = [ - {kind: 'audioinput', deviceId: 'default', label: '', groupId: ''}, - {kind: 'videoinput', deviceId: 'default', label: '', groupId: ''} - ]; - resolve(infos); - }); - }; - - if (browserDetails.version < 41) { - // Work around http://bugzil.la/1169665 - var orgEnumerateDevices = - navigator.mediaDevices.enumerateDevices.bind(navigator.mediaDevices); - navigator.mediaDevices.enumerateDevices = function() { - return orgEnumerateDevices().then(undefined, function(e) { - if (e.name === 'NotFoundError') { - return []; - } - throw e; - }); - }; - } -}; - -},{"../utils":9}],8:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ -'use strict'; -var safariShim = { - // TODO: DrAlex, should be here, double check against LayoutTests - // shimOnTrack: function() { }, - - // TODO: DrAlex - // attachMediaStream: function(element, stream) { }, - // reattachMediaStream: function(to, from) { }, - - // TODO: once the back-end for the mac port is done, add. - // TODO: check for webkitGTK+ - // shimPeerConnection: function() { }, - - shimGetUserMedia: function() { - navigator.getUserMedia = navigator.webkitGetUserMedia; - } -}; - -// Expose public methods. -module.exports = { - shimGetUserMedia: safariShim.shimGetUserMedia - // TODO - // shimOnTrack: safariShim.shimOnTrack, - // shimPeerConnection: safariShim.shimPeerConnection, - // attachMediaStream: safariShim.attachMediaStream, - // reattachMediaStream: safariShim.reattachMediaStream -}; - -},{}],9:[function(require,module,exports){ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - /* eslint-env node */ -'use strict'; - -var logDisabled_ = false; - -// Utility methods. -var utils = { - disableLog: function(bool) { - if (typeof bool !== 'boolean') { - return new Error('Argument type: ' + typeof bool + - '. Please use a boolean.'); - } - logDisabled_ = bool; - return (bool) ? 'adapter.js logging disabled' : - 'adapter.js logging enabled'; - }, - - log: function() { - if (typeof window === 'object') { - if (logDisabled_) { - return; - } - if (typeof console !== 'undefined' && typeof console.log === 'function') { - console.log.apply(console, arguments); - } - } - }, - - /** - * Extract browser version out of the provided user agent string. - * - * @param {!string} uastring userAgent string. - * @param {!string} expr Regular expression used as match criteria. - * @param {!number} pos position in the version string to be returned. - * @return {!number} browser version. - */ - extractVersion: function(uastring, expr, pos) { - var match = uastring.match(expr); - return match && match.length >= pos && parseInt(match[pos], 10); - }, - - /** - * Browser detector. - * - * @return {object} result containing browser, version and minVersion - * properties. - */ - detectBrowser: function() { - // Returned result object. - var result = {}; - result.browser = null; - result.version = null; - result.minVersion = null; - - // Fail early if it's not a browser - if (typeof window === 'undefined' || !window.navigator) { - result.browser = 'Not a browser.'; - return result; - } - - // Firefox. - if (navigator.mozGetUserMedia) { - result.browser = 'firefox'; - result.version = this.extractVersion(navigator.userAgent, - /Firefox\/([0-9]+)\./, 1); - result.minVersion = 31; - - // all webkit-based browsers - } else if (navigator.webkitGetUserMedia) { - // Chrome, Chromium, Webview, Opera, all use the chrome shim for now - if (window.webkitRTCPeerConnection) { - result.browser = 'chrome'; - result.version = this.extractVersion(navigator.userAgent, - /Chrom(e|ium)\/([0-9]+)\./, 2); - result.minVersion = 38; - - // Safari or unknown webkit-based - // for the time being Safari has support for MediaStreams but not webRTC - } else { - // Safari UA substrings of interest for reference: - // - webkit version: AppleWebKit/602.1.25 (also used in Op,Cr) - // - safari UI version: Version/9.0.3 (unique to Safari) - // - safari UI webkit version: Safari/601.4.4 (also used in Op,Cr) - // - // if the webkit version and safari UI webkit versions are equals, - // ... this is a stable version. - // - // only the internal webkit version is important today to know if - // media streams are supported - // - if (navigator.userAgent.match(/Version\/(\d+).(\d+)/)) { - result.browser = 'safari'; - result.version = this.extractVersion(navigator.userAgent, - /AppleWebKit\/([0-9]+)\./, 1); - result.minVersion = 602; - - // unknown webkit-based browser - } else { - result.browser = 'Unsupported webkit-based browser ' + - 'with GUM support but no WebRTC support.'; - return result; - } - } - - // Edge. - } else if (navigator.mediaDevices && - navigator.userAgent.match(/Edge\/(\d+).(\d+)$/)) { - result.browser = 'edge'; - result.version = this.extractVersion(navigator.userAgent, - /Edge\/(\d+).(\d+)$/, 2); - result.minVersion = 10547; - - // Default fallthrough: not supported. - } else { - result.browser = 'Not a supported browser.'; - return result; - } - - // Warn if version is less than minVersion. - if (result.version < result.minVersion) { - utils.log('Browser: ' + result.browser + ' Version: ' + result.version + - ' < minimum supported version: ' + result.minVersion + - '\n some things might not work!'); - } - - return result; - } -}; - -// Export. -module.exports = { - log: utils.log, - disableLog: utils.disableLog, - browserDetails: utils.detectBrowser(), - extractVersion: utils.extractVersion -}; - -},{}]},{},[1])(1) -}); \ No newline at end of file diff --git a/www/webrtc/js/common.js b/www/webrtc/js/common.js deleted file mode 100644 index d3de5988..00000000 --- a/www/webrtc/js/common.js +++ /dev/null @@ -1,12 +0,0 @@ -function trace(text) { - // This function is used for logging. - if (text[text.length - 1] === '\n') { - text = text.substring(0, text.length - 1); - } - if (window.performance) { - var now = (window.performance.now() / 1000).toFixed(3); - console.log(now + ': ' + text); - } else { - console.log(text); - } -} diff --git a/www/webrtc/js/main.js b/www/webrtc/js/main.js deleted file mode 100644 index 20c6b2da..00000000 --- a/www/webrtc/js/main.js +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. - */ - -'use strict'; - - -var vid2 = document.getElementById('vid2'); -var btn1 = document.getElementById('btn1'); -var btn3 = document.getElementById('btn3'); -var input1 = document.getElementById('input1'); - -btn1.addEventListener('click', start); - -btn3.addEventListener('click', stop); - -btn1.disabled = false; -btn3.disabled = true; - - -var pc2 = null; -var xmlhttp = null; - -function start() { - btn1.disabled = true; - btn3.disabled = false; - trace('Starting Call'); - - var servers = null; - var addr = input1.value; - pc2 = new RTCPeerConnection(servers); - trace('Created remote peer connection object pc2'); - pc2.onicecandidate = iceCallback; - pc2.onaddstream = gotRemoteStream; - - if (window.XMLHttpRequest) - { - // IE7+, Firefox, Chrome, Opera, Safari �����ִ�д��� - xmlhttp=new XMLHttpRequest(); - } - else - { - // IE6, IE5 �����ִ�д��� - xmlhttp=new ActiveXObject("Microsoft.XMLHTTP"); - } - - xmlhttp.onreadystatechange=function() - { - if (xmlhttp.readyState==4 && xmlhttp.status==200) - { - var res = xmlhttp.responseText; - gotoffer(res); - } - } - xmlhttp.open("GET",addr,true); - xmlhttp.send(); - -} - -function onCreateSessionDescriptionError(error) { - trace('Failed to create session description: ' + error.toString()); - stop(); -} - -function onCreateAnswerError(error) { - trace('Failed to set createAnswer: ' + error.toString()); - stop(); -} - -function onSetLocalDescriptionError(error) { - trace('Failed to set setLocalDescription: ' + error.toString()); - stop(); -} - -function onSetLocalDescriptionSuccess() { - trace('localDescription success.'); -} - -function gotoffer(offer) { - - trace('Offer from server \n' + offer); - //??????offer sdp????????RTCSessionDescription???? - var desc = new RTCSessionDescription(); - desc.sdp = offer; - desc.type = 'offer'; - pc2.setRemoteDescription(desc); - // Since the 'remote' side has no media stream we need - // to pass in the right constraints in order for it to - // accept the incoming offer of audio and video. - pc2.createAnswer().then( - gotDescription2, - onCreateSessionDescriptionError - ); -} - -function gotDescription2(desc) { - // Provisional answer, set a=inactive & set sdp type to pranswer. - /*desc.sdp = desc.sdp.replace(/a=recvonly/g, 'a=inactive'); - desc.type = 'pranswer';*/ - - pc2.setLocalDescription(desc).then( - onSetLocalDescriptionSuccess, - onSetLocalDescriptionError - ); - trace('Pranswer from pc2 \n' + desc.sdp); - - //conn.send(JSON.stringify(desc)); - // send desc.sdp to server -} - -function stop() { - trace('Ending Call' + '\n\n'); - pc2.close(); - pc2 = null; -} - -function gotRemoteStream(e) { - vid2.srcObject = e.stream; - trace('Received remote stream'); -} - -function iceCallback(event) { - if (event.candidate) { - trace('Remote ICE candidate: \n ' + event.candidate.candidate); - //conn.send(JSON.stringify(event.candidate)); - } - else { - // All ICE candidates have been sent - } -} From 356a8d159606652c447220e047160c38acaa85ad Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 31 Mar 2021 18:32:43 +0800 Subject: [PATCH 036/218] =?UTF-8?q?=E5=AE=9E=E7=8E=B0=E9=83=A8=E5=88=86ans?= =?UTF-8?q?wer=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/CMakeLists.txt | 2 +- webrtc/Sdp.cpp | 105 ++++++++++++++++++++++-- webrtc/Sdp.h | 18 ++--- webrtc/offer-simulcast.sdp | 156 +++++++++++++++++++++++++++++++++++ webrtc/offer.sdp | 161 +++++++++++++++++++++++++++++++++++++ 5 files changed, 423 insertions(+), 19 deletions(-) create mode 100644 webrtc/offer-simulcast.sdp create mode 100644 webrtc/offer.sdp diff --git a/webrtc/CMakeLists.txt b/webrtc/CMakeLists.txt index 03039aea..db7dd3a8 100644 --- a/webrtc/CMakeLists.txt +++ b/webrtc/CMakeLists.txt @@ -10,7 +10,7 @@ else () endif () include_directories(./) -file(GLOB SRC_LIST ./*.*) +file(GLOB SRC_LIST ./*.cpp ./*.h ./*.hpp) add_library(webrtc ${SRC_LIST}) set(LINK_LIB_LIST ${LINK_LIB_LIST} PARENT_SCOPE) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 5f40da33..ccedc67f 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -3,12 +3,9 @@ // #include "Sdp.h" -#include "assert.h" #include "Common/Parser.h" #include -#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); - using onCreateSdpItem = function; static map sdpItemCreator; @@ -1148,7 +1145,6 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, - DtlsRole role, RtpDirection direction, const SdpAttrFingerprint &fingerprint) { video.setDefaultSetting(TrackVideo); @@ -1157,9 +1153,9 @@ void RtcConfigure::setDefaultSetting(string ice_ufrag, video.ice_ufrag = audio.ice_ufrag = application.ice_ufrag = ice_ufrag; video.ice_pwd = audio.ice_pwd = application.ice_pwd = ice_ufrag; - video.role = audio.role = application.role = role; video.direction = audio.direction = application.direction = direction; video.fingerprint = audio.fingerprint = application.fingerprint = fingerprint; + application.enable = false; } void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType type) { @@ -1203,5 +1199,100 @@ shared_ptr RtcConfigure::createOffer(){ } shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ - return nullptr; -} \ No newline at end of file + shared_ptr ret = std::make_shared(); + ret->version = offer.version; + ret->origin.parse("- 0 0 IN IP4 0.0.0.0"); + ret->session_name = "zlmediakit_webrtc_session"; + ret->session_info = "zlmediakit_webrtc_session"; + ret->connection.parse("IN IP4 0.0.0.0"); + ret->msid_semantic.parse("WMS *"); + matchMedia(ret, TrackVideo, offer.media, video); + matchMedia(ret, TrackAudio, offer.media, audio); + matchMedia(ret, TrackApplication, offer.media, application); + return ret; +} + +void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure){ + if (!configure.enable) { + return; + } + for (auto &codec : configure.preferred_codec) { + for (auto &media : medias) { + if (media.type != type) { + continue; + } + if (media.ice_lite && configure.ice_lite) { + WarnL << "offer sdp开启了ice_lite模式，但是answer sdp配置为不支持"; + continue; + } + + const RtcCodecPlan *plan_ptr = nullptr; + for (auto &plan : media.plan) { + if (getCodecId(plan.codec) != codec) { + continue; + } + //命中偏好的编码格式 + if (!onMatchCodecPlan(plan, codec)) { + continue; + } + plan_ptr = &plan; + } + if (!plan_ptr) { + continue; + } + RtcMedia answer_media; + answer_media.type = media.type; + answer_media.mid = media.type; + answer_media.proto = media.proto; + answer_media.rtcp_mux = media.rtcp_mux && configure.rtcp_mux; + answer_media.rtcp_rsize = media.rtcp_rsize && configure.rtcp_rsize; + answer_media.ice_trickle = media.ice_trickle && configure.ice_trickle; + answer_media.ice_renomination = media.ice_renomination && configure.ice_renomination; + switch (media.role) { + case DtlsRole::actpass : { + answer_media.role = DtlsRole::passive; + break; + } + case DtlsRole::active : { + answer_media.role = DtlsRole::passive; + break; + } + case DtlsRole::passive : { + answer_media.role = DtlsRole::active; + break; + } + default: continue; + } + + switch (media.direction) { + case RtpDirection::sendonly : { + if (configure.direction != RtpDirection::recvonly && + configure.direction != RtpDirection::sendrecv) { + //我们不支持接收 + continue; + } + answer_media.direction = RtpDirection::recvonly; + break; + } + case RtpDirection::recvonly : { + if (configure.direction != RtpDirection::sendonly && + configure.direction != RtpDirection::sendrecv) { + //我们不支持发送 + continue; + } + answer_media.direction = RtpDirection::sendonly; + break; + } + case RtpDirection::sendrecv : { + //对方支持发送接收，那么最终能力根据配置来决定 + answer_media.direction = configure.direction; + break; + } + default: continue; + } + answer_media.plan.emplace_back(*plan_ptr); + ret->media.emplace_back(answer_media); + } + } + +} diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index a3ef62bc..602ce691 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -7,10 +7,13 @@ #include #include +#include "assert.h" #include "Extension/Frame.h" using namespace std; using namespace mediakit; +#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); + //https://datatracker.ietf.org/doc/rfc4566/?include_text=1 //https://blog.csdn.net/aggresss/article/details/109850434 //https://aggresss.blog.csdn.net/article/details/106436703 @@ -545,15 +548,6 @@ public: ////////////////////////////////////////////////////////////////// -//ssrc类型 -enum class RtcSSRCType { - rtp = 0, - rtx, - sim_low, - sim_mid, - ssrc_high -}; - //ssrc相关信息 class RtcSSRC{ public: @@ -672,7 +666,6 @@ public: string ice_ufrag; string ice_pwd; - DtlsRole role{DtlsRole::invalid}; RtpDirection direction{RtpDirection::invalid}; SdpAttrFingerprint fingerprint; @@ -690,13 +683,16 @@ public: void setDefaultSetting(string ice_ufrag, string ice_pwd, - DtlsRole role, RtpDirection direction, const SdpAttrFingerprint &fingerprint); void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid); shared_ptr createOffer(); shared_ptr createAnswer(const RtcSession &offer); + +private: + void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); + bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec) { return true; } }; diff --git a/webrtc/offer-simulcast.sdp b/webrtc/offer-simulcast.sdp new file mode 100644 index 00000000..91701061 --- /dev/null +++ b/webrtc/offer-simulcast.sdp @@ -0,0 +1,156 @@ +v=0 +o=- 99044479296054817 2 IN IP4 127.0.0.1 +s=- +t=0 0 +a=group:BUNDLE 0 1 +a=extmap-allow-mixed +a=msid-semantic: WMS +m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 +c=IN IP4 0.0.0.0 +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:s95k +a=ice-pwd:JFAzK9rInMLPOosVj4Z7lC+Q +a=ice-options:trickle +a=fingerprint:sha-256 50:76:BB:FA:A8:D3:B7:19:81:C7:8D:19:60:C7:A9:7E:0E:5F:AA:3F:FE:68:60:98:B1:93:95:BC:A9:29:E4:D1 +a=setup:actpass +a=mid:0 +a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level +a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id +a=sendrecv +a=msid:- 83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 +a=rtcp-mux +a=rtpmap:111 opus/48000/2 +a=rtcp-fb:111 transport-cc +a=fmtp:111 minptime=10;useinbandfec=1 +a=rtpmap:103 ISAC/16000 +a=rtpmap:104 ISAC/32000 +a=rtpmap:9 G722/8000 +a=rtpmap:0 PCMU/8000 +a=rtpmap:8 PCMA/8000 +a=rtpmap:106 CN/32000 +a=rtpmap:105 CN/16000 +a=rtpmap:13 CN/8000 +a=rtpmap:110 telephone-event/48000 +a=rtpmap:112 telephone-event/32000 +a=rtpmap:113 telephone-event/16000 +a=rtpmap:126 telephone-event/8000 +a=ssrc:1320289824 cname:YOYQAT5W3y5SSliB +a=ssrc:1320289824 msid:- 83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 +a=ssrc:1320289824 mslabel:- +a=ssrc:1320289824 label:83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 +m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 +c=IN IP4 0.0.0.0 +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:s95k +a=ice-pwd:JFAzK9rInMLPOosVj4Z7lC+Q +a=ice-options:trickle +a=fingerprint:sha-256 50:76:BB:FA:A8:D3:B7:19:81:C7:8D:19:60:C7:A9:7E:0E:5F:AA:3F:FE:68:60:98:B1:93:95:BC:A9:29:E4:D1 +a=setup:actpass +a=mid:1 +a=extmap:14 urn:ietf:params:rtp-hdrext:toffset +a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:13 urn:3gpp:video-orientation +a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type +a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing +a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space +a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id +a=sendrecv +a=msid:- cd8e4810-dadf-4f11-8799-eed2560fe196 +a=rtcp-mux +a=rtcp-rsize +a=rtpmap:96 VP8/90000 +a=rtcp-fb:96 goog-remb +a=rtcp-fb:96 transport-cc +a=rtcp-fb:96 ccm fir +a=rtcp-fb:96 nack +a=rtcp-fb:96 nack pli +a=rtpmap:97 rtx/90000 +a=fmtp:97 apt=96 +a=rtpmap:98 VP9/90000 +a=rtcp-fb:98 goog-remb +a=rtcp-fb:98 transport-cc +a=rtcp-fb:98 ccm fir +a=rtcp-fb:98 nack +a=rtcp-fb:98 nack pli +a=fmtp:98 profile-id=0 +a=rtpmap:99 rtx/90000 +a=fmtp:99 apt=98 +a=rtpmap:100 VP9/90000 +a=rtcp-fb:100 goog-remb +a=rtcp-fb:100 transport-cc +a=rtcp-fb:100 ccm fir +a=rtcp-fb:100 nack +a=rtcp-fb:100 nack pli +a=fmtp:100 profile-id=2 +a=rtpmap:101 rtx/90000 +a=fmtp:101 apt=100 +a=rtpmap:102 H264/90000 +a=rtcp-fb:102 goog-remb +a=rtcp-fb:102 transport-cc +a=rtcp-fb:102 ccm fir +a=rtcp-fb:102 nack +a=rtcp-fb:102 nack pli +a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f +a=rtpmap:121 rtx/90000 +a=fmtp:121 apt=102 +a=rtpmap:127 H264/90000 +a=rtcp-fb:127 goog-remb +a=rtcp-fb:127 transport-cc +a=rtcp-fb:127 ccm fir +a=rtcp-fb:127 nack +a=rtcp-fb:127 nack pli +a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f +a=rtpmap:120 rtx/90000 +a=fmtp:120 apt=127 +a=rtpmap:125 H264/90000 +a=rtcp-fb:125 goog-remb +a=rtcp-fb:125 transport-cc +a=rtcp-fb:125 ccm fir +a=rtcp-fb:125 nack +a=rtcp-fb:125 nack pli +a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f +a=rtpmap:107 rtx/90000 +a=fmtp:107 apt=125 +a=rtpmap:108 H264/90000 +a=rtcp-fb:108 goog-remb +a=rtcp-fb:108 transport-cc +a=rtcp-fb:108 ccm fir +a=rtcp-fb:108 nack +a=rtcp-fb:108 nack pli +a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f +a=rtpmap:109 rtx/90000 +a=fmtp:109 apt=108 +a=rtpmap:124 H264/90000 +a=rtcp-fb:124 goog-remb +a=rtcp-fb:124 transport-cc +a=rtcp-fb:124 ccm fir +a=rtcp-fb:124 nack +a=rtcp-fb:124 nack pli +a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f +a=rtpmap:119 rtx/90000 +a=fmtp:119 apt=124 +a=rtpmap:123 H264/90000 +a=rtcp-fb:123 goog-remb +a=rtcp-fb:123 transport-cc +a=rtcp-fb:123 ccm fir +a=rtcp-fb:123 nack +a=rtcp-fb:123 nack pli +a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f +a=rtpmap:118 rtx/90000 +a=fmtp:118 apt=123 +a=rtpmap:114 red/90000 +a=rtpmap:115 rtx/90000 +a=fmtp:115 apt=114 +a=rtpmap:116 ulpfec/90000 +a=rid:q send +a=rid:h send +a=rid:f send +a=simulcast:send q;h;f \ No newline at end of file diff --git a/webrtc/offer.sdp b/webrtc/offer.sdp new file mode 100644 index 00000000..762150d8 --- /dev/null +++ b/webrtc/offer.sdp @@ -0,0 +1,161 @@ +v=0 +o=- 257973874652185302 2 IN IP4 127.0.0.1 +s=- +t=0 0 +a=group:BUNDLE 0 1 +a=extmap-allow-mixed +a=msid-semantic: WMS +m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 +c=IN IP4 0.0.0.0 +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:w2IN +a=ice-pwd:X7kCoPoI2NqW8kxuV9LHRR78 +a=ice-options:trickle +a=fingerprint:sha-256 7A:A7:A4:9A:BC:37:64:68:9C:48:E5:E9:9B:97:BD:88:17:3E:E5:44:29:4D:6D:BB:AB:2C:85:B8:DE:7A:15:B1 +a=setup:actpass +a=mid:0 +a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level +a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id +a=sendrecv +a=msid:- 56049f63-4b19-45c4-aa0a-8895049b5430 +a=rtcp-mux +a=rtpmap:111 opus/48000/2 +a=rtcp-fb:111 transport-cc +a=fmtp:111 minptime=10;useinbandfec=1 +a=rtpmap:103 ISAC/16000 +a=rtpmap:104 ISAC/32000 +a=rtpmap:9 G722/8000 +a=rtpmap:0 PCMU/8000 +a=rtpmap:8 PCMA/8000 +a=rtpmap:106 CN/32000 +a=rtpmap:105 CN/16000 +a=rtpmap:13 CN/8000 +a=rtpmap:110 telephone-event/48000 +a=rtpmap:112 telephone-event/32000 +a=rtpmap:113 telephone-event/16000 +a=rtpmap:126 telephone-event/8000 +a=ssrc:3304267696 cname:sCv+hHL1+2UbfMTB +a=ssrc:3304267696 msid:- 56049f63-4b19-45c4-aa0a-8895049b5430 +a=ssrc:3304267696 mslabel:- +a=ssrc:3304267696 label:56049f63-4b19-45c4-aa0a-8895049b5430 +m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 +c=IN IP4 0.0.0.0 +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:w2IN +a=ice-pwd:X7kCoPoI2NqW8kxuV9LHRR78 +a=ice-options:trickle +a=fingerprint:sha-256 7A:A7:A4:9A:BC:37:64:68:9C:48:E5:E9:9B:97:BD:88:17:3E:E5:44:29:4D:6D:BB:AB:2C:85:B8:DE:7A:15:B1 +a=setup:actpass +a=mid:1 +a=extmap:14 urn:ietf:params:rtp-hdrext:toffset +a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:13 urn:3gpp:video-orientation +a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type +a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing +a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space +a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id +a=sendrecv +a=msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 +a=rtcp-mux +a=rtcp-rsize +a=rtpmap:96 VP8/90000 +a=rtcp-fb:96 goog-remb +a=rtcp-fb:96 transport-cc +a=rtcp-fb:96 ccm fir +a=rtcp-fb:96 nack +a=rtcp-fb:96 nack pli +a=rtpmap:97 rtx/90000 +a=fmtp:97 apt=96 +a=rtpmap:98 VP9/90000 +a=rtcp-fb:98 goog-remb +a=rtcp-fb:98 transport-cc +a=rtcp-fb:98 ccm fir +a=rtcp-fb:98 nack +a=rtcp-fb:98 nack pli +a=fmtp:98 profile-id=0 +a=rtpmap:99 rtx/90000 +a=fmtp:99 apt=98 +a=rtpmap:100 VP9/90000 +a=rtcp-fb:100 goog-remb +a=rtcp-fb:100 transport-cc +a=rtcp-fb:100 ccm fir +a=rtcp-fb:100 nack +a=rtcp-fb:100 nack pli +a=fmtp:100 profile-id=2 +a=rtpmap:101 rtx/90000 +a=fmtp:101 apt=100 +a=rtpmap:102 H264/90000 +a=rtcp-fb:102 goog-remb +a=rtcp-fb:102 transport-cc +a=rtcp-fb:102 ccm fir +a=rtcp-fb:102 nack +a=rtcp-fb:102 nack pli +a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f +a=rtpmap:121 rtx/90000 +a=fmtp:121 apt=102 +a=rtpmap:127 H264/90000 +a=rtcp-fb:127 goog-remb +a=rtcp-fb:127 transport-cc +a=rtcp-fb:127 ccm fir +a=rtcp-fb:127 nack +a=rtcp-fb:127 nack pli +a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f +a=rtpmap:120 rtx/90000 +a=fmtp:120 apt=127 +a=rtpmap:125 H264/90000 +a=rtcp-fb:125 goog-remb +a=rtcp-fb:125 transport-cc +a=rtcp-fb:125 ccm fir +a=rtcp-fb:125 nack +a=rtcp-fb:125 nack pli +a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f +a=rtpmap:107 rtx/90000 +a=fmtp:107 apt=125 +a=rtpmap:108 H264/90000 +a=rtcp-fb:108 goog-remb +a=rtcp-fb:108 transport-cc +a=rtcp-fb:108 ccm fir +a=rtcp-fb:108 nack +a=rtcp-fb:108 nack pli +a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f +a=rtpmap:109 rtx/90000 +a=fmtp:109 apt=108 +a=rtpmap:124 H264/90000 +a=rtcp-fb:124 goog-remb +a=rtcp-fb:124 transport-cc +a=rtcp-fb:124 ccm fir +a=rtcp-fb:124 nack +a=rtcp-fb:124 nack pli +a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f +a=rtpmap:119 rtx/90000 +a=fmtp:119 apt=124 +a=rtpmap:123 H264/90000 +a=rtcp-fb:123 goog-remb +a=rtcp-fb:123 transport-cc +a=rtcp-fb:123 ccm fir +a=rtcp-fb:123 nack +a=rtcp-fb:123 nack pli +a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f +a=rtpmap:118 rtx/90000 +a=fmtp:118 apt=123 +a=rtpmap:114 red/90000 +a=rtpmap:115 rtx/90000 +a=fmtp:115 apt=114 +a=rtpmap:116 ulpfec/90000 +a=ssrc-group:FID 1128910219 3552306261 +a=ssrc:1128910219 cname:sCv+hHL1+2UbfMTB +a=ssrc:1128910219 msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 +a=ssrc:1128910219 mslabel:- +a=ssrc:1128910219 label:a73b3f5f-007a-4a46-ac8b-582ca7fee460 +a=ssrc:3552306261 cname:sCv+hHL1+2UbfMTB +a=ssrc:3552306261 msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 +a=ssrc:3552306261 mslabel:- +a=ssrc:3552306261 label:a73b3f5f-007a-4a46-ac8b-582ca7fee460 \ No newline at end of file From 868b379a12db418b9302f06f9f4fab3247270b8f Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 10:09:50 +0800 Subject: [PATCH 037/218] =?UTF-8?q?=E5=AE=8C=E5=96=84answer=20sdp=E4=BB=A3?= =?UTF-8?q?=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 113 ++++++++++++++++++++++++++++--------- webrtc/Sdp.h | 1 + webrtc/WebRtcTransport.cpp | 72 +++++++++++++++-------- webrtc/WebRtcTransport.h | 1 + www/webrtc/ZLMRTCClient.js | 2 +- 5 files changed, 140 insertions(+), 49 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index ccedc67f..60c2bad0 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1105,6 +1105,15 @@ void RtcSession::checkValid() const{ } } +RtcMedia *RtcSession::getMedia(TrackType type){ + for(auto &m : media){ + if(m.type == type){ + return &m; + } + } + return nullptr; +} + void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ enable = true; rtcp_mux = true; @@ -1119,13 +1128,13 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ ice_trickle = true; ice_renomination = false; switch (type) { - case TrackVideo: { + case TrackAudio: { preferred_codec = {CodecAAC, CodecOpus, CodecG711U, CodecG711A}; rtcp_fb = {"transport-cc"}; extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"}; break; } - case TrackAudio: { + case TrackVideo: { preferred_codec = {CodecH264, CodecH265}; rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"}; extmap = {"2 urn:ietf:params:rtp-hdrext:toffset", @@ -1152,7 +1161,7 @@ void RtcConfigure::setDefaultSetting(string ice_ufrag, application.setDefaultSetting(TrackApplication); video.ice_ufrag = audio.ice_ufrag = application.ice_ufrag = ice_ufrag; - video.ice_pwd = audio.ice_pwd = application.ice_pwd = ice_ufrag; + video.ice_pwd = audio.ice_pwd = application.ice_pwd = ice_pwd; video.direction = audio.direction = application.direction = direction; video.fingerprint = audio.fingerprint = application.fingerprint = fingerprint; application.enable = false; @@ -1217,17 +1226,16 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const return; } for (auto &codec : configure.preferred_codec) { - for (auto &media : medias) { - if (media.type != type) { + for (auto &offer_media : medias) { + if (offer_media.type != type) { continue; } - if (media.ice_lite && configure.ice_lite) { + if (offer_media.ice_lite && configure.ice_lite) { WarnL << "offer sdp开启了ice_lite模式，但是answer sdp配置为不支持"; continue; } - - const RtcCodecPlan *plan_ptr = nullptr; - for (auto &plan : media.plan) { + const RtcCodecPlan *offer_plan_ptr = nullptr; + for (auto &plan : offer_media.plan) { if (getCodecId(plan.codec) != codec) { continue; } @@ -1235,24 +1243,27 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const if (!onMatchCodecPlan(plan, codec)) { continue; } - plan_ptr = &plan; + //找到中意的codec + offer_plan_ptr = &plan; + break; } - if (!plan_ptr) { + if (!offer_plan_ptr) { + //offer中该媒体的所有的codec都不支持 continue; } RtcMedia answer_media; - answer_media.type = media.type; - answer_media.mid = media.type; - answer_media.proto = media.proto; - answer_media.rtcp_mux = media.rtcp_mux && configure.rtcp_mux; - answer_media.rtcp_rsize = media.rtcp_rsize && configure.rtcp_rsize; - answer_media.ice_trickle = media.ice_trickle && configure.ice_trickle; - answer_media.ice_renomination = media.ice_renomination && configure.ice_renomination; - switch (media.role) { - case DtlsRole::actpass : { - answer_media.role = DtlsRole::passive; - break; - } + answer_media.type = offer_media.type; + answer_media.mid = offer_media.mid; + answer_media.proto = offer_media.proto; + answer_media.rtcp_mux = offer_media.rtcp_mux && configure.rtcp_mux; + answer_media.rtcp_rsize = offer_media.rtcp_rsize && configure.rtcp_rsize; + answer_media.ice_trickle = offer_media.ice_trickle && configure.ice_trickle; + answer_media.ice_renomination = offer_media.ice_renomination && configure.ice_renomination; + answer_media.ice_ufrag = configure.ice_ufrag; + answer_media.ice_pwd = configure.ice_pwd; + answer_media.fingerprint = configure.fingerprint; + switch (offer_media.role) { + case DtlsRole::actpass : case DtlsRole::active : { answer_media.role = DtlsRole::passive; break; @@ -1264,7 +1275,7 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const default: continue; } - switch (media.direction) { + switch (offer_media.direction) { case RtpDirection::sendonly : { if (configure.direction != RtpDirection::recvonly && configure.direction != RtpDirection::sendrecv) { @@ -1290,9 +1301,59 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const } default: continue; } - answer_media.plan.emplace_back(*plan_ptr); + answer_media.plan.emplace_back(*offer_plan_ptr); + if (configure.support_red || configure.support_rtx || configure.support_ulpfec) { + for (auto &plan : offer_media.plan) { + if (!strcasecmp(plan.codec.data(), "rtx")) { + if (configure.support_rtx && atoi(plan.getFmtp("apt").data()) == offer_plan_ptr->pt) { + answer_media.plan.emplace_back(plan); + } + continue; + } + if (!strcasecmp(plan.codec.data(), "red")) { + if (configure.support_red) { + answer_media.plan.emplace_back(plan); + } + continue; + } + if (!strcasecmp(plan.codec.data(), "ulpfec")) { + if (configure.support_ulpfec) { + answer_media.plan.emplace_back(plan); + } + continue; + } + } + } + + //这是我们支持的扩展 + unordered_set extmap_set; + for (auto &ext : configure.extmap) { + SdpAttrExtmap ext_cfg; + ext_cfg.parse(ext); + extmap_set.emplace(ext_cfg.ext); + } + + //对方和我方都支持的扩展，那么我们都支持 + for (auto &ext : offer_media.extmap) { + if (extmap_set.find(ext.ext) != extmap_set.end()) { + answer_media.extmap.emplace_back(ext); + } + } + //我们支持的rtcp类型 + unordered_set rtcp_fb_set; + for (auto &fp : configure.rtcp_fb) { + rtcp_fb_set.emplace(fp); + } + vector offer_rtcp_fb; + for (auto &fp : offer_plan_ptr->rtcp_fb) { + if (rtcp_fb_set.find(fp) != rtcp_fb_set.end()) { + //对方该rtcp被我们支持 + offer_rtcp_fb.emplace_back(fp); + } + } + answer_media.plan[0].rtcp_fb.swap(offer_rtcp_fb); ret->media.emplace_back(answer_media); + return; } } - } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 602ce691..97c5f020 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -644,6 +644,7 @@ public: void loadFrom(const string &sdp); void checkValid() const; string toString() const; + RtcMedia *getMedia(TrackType type); }; class RtcConfigure { diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index c96c1e73..3c58628e 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -24,7 +24,11 @@ void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) { InfoL; - dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); + if (_answer_sdp->media[0].role == DtlsRole::passive) { + dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); + } else { + dtls_transport_->Run(RTC::DtlsTransport::Role::CLIENT); + } } void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) { @@ -47,6 +51,7 @@ void WebRtcTransport::OnDtlsTransportConnected( std::string &remoteCert) { InfoL; srtp_session_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); + srtp_session_recv_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen); onDtlsConnected(); } @@ -62,14 +67,45 @@ void WebRtcTransport::onWrite(const char *buf, size_t len){ onWrite(buf, len, (struct sockaddr_in *)tuple); } +string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ + auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str); + for (auto &finger_prints : transport->GetLocalFingerprints()) { + if (finger_prints.algorithm == algorithm) { + return finger_prints.value; + } + } + throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str); +} + std::string WebRtcTransport::getAnswerSdp(const string &offer){ InfoL << offer; _offer_sdp = std::make_shared(); _offer_sdp->loadFrom(offer); + SdpAttrFingerprint fingerprint; + fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; + fingerprint.hash = getFingerprint(fingerprint.algorithm, dtls_transport_); + RtcConfigure configure; + configure.setDefaultSetting(ice_server_->GetUsernameFragment(), ice_server_->GetPassword(), RtpDirection::recvonly, fingerprint); _answer_sdp = configure.createAnswer(*_offer_sdp); - return _answer_sdp->toString(); + + //设置远端dtls签名 + RTC::DtlsTransport::Fingerprint remote_fingerprint; + remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm); + remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash; + dtls_transport_->SetRemoteFingerprint(remote_fingerprint); + + if (!_offer_sdp->group.mids.empty()) { + for (auto &m : _answer_sdp->media) { + _answer_sdp->group.mids.emplace_back(m.mid); + } + } else { + throw std::invalid_argument("支持group BUNDLE模式"); + } + auto str = _answer_sdp->toString(); + InfoL << str; + return str; } std::string WebRtcTransport::getOfferSdp() { @@ -78,14 +114,6 @@ std::string WebRtcTransport::getOfferSdp() { remote_fingerprint.value = ""; dtls_transport_->SetRemoteFingerprint(remote_fingerprint); - string finger_print_sha256; - auto finger_prints = dtls_transport_->GetLocalFingerprints(); - for (size_t i = 0; i < finger_prints.size(); i++) { - if (finger_prints[i].algorithm == RTC::DtlsTransport::FingerprintAlgorithm::SHA256) { - finger_print_sha256 = finger_prints[i].value; - } - } - char sdp[1024 * 10] = {0}; auto ssrc = getSSRC(); auto ip = getIP(); @@ -118,7 +146,7 @@ std::string WebRtcTransport::getOfferSdp() { "a=candidate:%s 1 udp %u %s %u typ %s\r\n", ip.c_str(), port, pt, ip.c_str(), ice_server_->GetUsernameFragment().c_str(),ice_server_->GetPassword().c_str(), - finger_print_sha256.c_str(), pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); + "", pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); return sdp; } @@ -137,6 +165,16 @@ bool is_rtcp(char *buf) { } void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple) { + if (is_rtp(buf)) { + RtpHeader *header = (RtpHeader *) buf; + InfoL << "rtp:" << header->dumpString(len); + return; + } + if (is_rtcp(buf)) { + RtcpHeader *header = (RtcpHeader *) buf; +// InfoL << "rtcp:" << header->dumpString(); + return; + } if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len); if (packet == nullptr) { @@ -147,17 +185,7 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTup return; } if (is_dtls(buf)) { - dtls_transport_->ProcessDtlsData((uint8_t *)buf, len); - return; - } - if (is_rtp(buf)) { - RtpHeader *header = (RtpHeader *) buf; -// InfoL << "rtp:" << header->dumpString(len); - return; - } - if (is_rtcp(buf)) { - RtcpHeader *header = (RtcpHeader *) buf; -// InfoL << "rtcp:" << header->dumpString(); + dtls_transport_->ProcessDtlsData((uint8_t *) buf, len); return; } } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 29ca7c86..67a2083a 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -77,6 +77,7 @@ private: std::shared_ptr ice_server_; std::shared_ptr dtls_transport_; std::shared_ptr srtp_session_; + std::shared_ptr srtp_session_recv_; RtcSession::Ptr _offer_sdp; RtcSession::Ptr _answer_sdp; }; diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js index 233ed60b..76557c32 100644 --- a/www/webrtc/ZLMRTCClient.js +++ b/www/webrtc/ZLMRTCClient.js @@ -7357,7 +7357,7 @@ var ZLMRTCClient = (function (exports) { 'Content-Type': 'text/plain;charset=utf-8' } }).then(response => { - let ret = JSON.parse(response.data); + let ret = response.data; if (ret.code != 0) { // mean failed for offer/anwser exchange From 83208b0b311dad3bb84ad066458fe98cd8985507 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 10:35:31 +0800 Subject: [PATCH 038/218] =?UTF-8?q?=E5=AE=8C=E5=96=84?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 44 +++++++++++++++++++++++++++++------- webrtc/Sdp.h | 4 ++++ webrtc/WebRtcTransport.cpp | 2 +- webrtc/answer.sdp | 46 ++++++++++++++++++++++++++++++++++++++ 4 files changed, 87 insertions(+), 9 deletions(-) create mode 100644 webrtc/answer.sdp diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 60c2bad0..eca66a61 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -468,6 +468,33 @@ string SdpAttrRtcp::toString() const { return SdpItem::toString(); } +void SdpAttrIceOption::parse(const string &str){ + auto vec = split(str, " "); + for (auto &v : vec) { + if (!strcasecmp(v.data(), "trickle")) { + trickle = true; + continue; + } + if (!strcasecmp(v.data(), "renomination")) { + renomination = true; + continue; + } + } +} + +string SdpAttrIceOption::toString() const{ + if (value.empty()) { + if (trickle && renomination) { + value = "trickle renomination"; + } else if (trickle) { + value = "trickle"; + } else if (renomination) { + value = "renomination"; + } + } + return value; +} + void SdpAttrFingerprint::parse(const string &str) { auto vec = split(str, " "); if (vec.size() != 2) { @@ -751,9 +778,10 @@ void RtcSession::loadFrom(const string &str) { rtc_media.ice_pwd = media.getStringItem('a', "ice-pwd"); rtc_media.role = media.getItemClass('a', "setup").role; rtc_media.fingerprint = media.getItemClass('a', "fingerprint"); - rtc_media.ice_trickle = media.getItem('a', "ice-trickle").operator bool(); rtc_media.ice_lite = media.getItem('a', "ice-lite").operator bool(); - rtc_media.ice_renomination = media.getItem('a', "ice-renomination").operator bool(); + auto ice_options = media.getItemClass('a', "ice-options"); + rtc_media.ice_trickle = ice_options.trickle; + rtc_media.ice_renomination = ice_options.renomination; rtc_media.candidate = media.getAllItem('a', "candidate"); if (mline.type == TrackType::TrackApplication) { @@ -922,15 +950,15 @@ string RtcSession::toString() const{ sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.ice_pwd))); sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.fingerprint))); sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.role))); - if (m.ice_trickle) { - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-trickle"))); + if (m.ice_trickle || m.ice_renomination) { + auto attr = std::make_shared(); + attr->trickle = m.ice_trickle; + attr->renomination = m.ice_renomination; + sdp_media.items.emplace_back(wrapSdpAttr(attr)); } if (m.ice_lite) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-lite"))); } - if (m.ice_renomination) { - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-renomination"))); - } for (auto &ext : m.extmap) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(ext))); } @@ -1215,8 +1243,8 @@ shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ ret->session_info = "zlmediakit_webrtc_session"; ret->connection.parse("IN IP4 0.0.0.0"); ret->msid_semantic.parse("WMS *"); - matchMedia(ret, TrackVideo, offer.media, video); matchMedia(ret, TrackAudio, offer.media, audio); + matchMedia(ret, TrackVideo, offer.media, video); matchMedia(ret, TrackApplication, offer.media, application); return ret; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 97c5f020..fa167fbc 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -282,6 +282,10 @@ public: class SdpAttrIceOption : public SdpItem { public: //a=ice-options:trickle + bool trickle{false}; + bool renomination{false}; + void parse(const string &str) override; + string toString() const override; const char* getKey() const override { return "ice-options";} }; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 3c58628e..4c43f97d 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -78,7 +78,7 @@ string getFingerprint(const string &algorithm_str, const std::shared_ptr(); _offer_sdp->loadFrom(offer); diff --git a/webrtc/answer.sdp b/webrtc/answer.sdp new file mode 100644 index 00000000..82df3355 --- /dev/null +++ b/webrtc/answer.sdp @@ -0,0 +1,46 @@ + v=0 +o=- 0 0 IN IP4 0.0.0.0 +s=zlmediakit_webrtc_session +i=zlmediakit_webrtc_session +t=0 0 +c=IN IP4 0.0.0.0 +a=msid-semantic: WMS * +a=group:BUNDLE 0 1 +m=audio 0 UDP/TLS/RTP/SAVPF 111 +a=mid:0 +c=IN IP4 0.0.0.0 +a=ice-ufrag:PdK3 +a=ice-pwd:PdK31zZ1sKhPU2ACzP5av37f +a=fingerprint:sha-256 D1:4F:7B:70:86:99:BE:1E:7E:B8:C7:0F:13:BC:78:C4:6A:F9:2C:F6:F3:5E:42:11:0F:1C:74:18:F6:DD:95:75 +a=setup:passive +a=ice-options:trickle +a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level +a=recvonly +a=rtcp-mux +a=rtpmap:111 opus/48000/2 +a=rtcp-fb:111 transport-cc +a=fmtp:111 minptime=10;useinbandfec=1 +m=video 0 UDP/TLS/RTP/SAVPF 102 121 +a=mid:1 +c=IN IP4 0.0.0.0 +a=ice-ufrag:PdK3 +a=ice-pwd:PdK31zZ1sKhPU2ACzP5av37f +a=fingerprint:sha-256 D1:4F:7B:70:86:99:BE:1E:7E:B8:C7:0F:13:BC:78:C4:6A:F9:2C:F6:F3:5E:42:11:0F:1C:74:18:F6:DD:95:75 +a=setup:passive +a=ice-options:trickle +a=extmap:14 urn:ietf:params:rtp-hdrext:toffset +a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:13 urn:3gpp:video-orientation +a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +a=recvonly +a=rtcp-mux +a=rtpmap:102 H264/90000 +a=rtcp-fb:102 goog-remb +a=rtcp-fb:102 transport-cc +a=rtcp-fb:102 ccm fir +a=rtcp-fb:102 nack +a=rtcp-fb:102 nack pli +a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f +a=rtpmap:121 rtx/90000 +a=fmtp:121 apt=102 \ No newline at end of file From 4853a5af4821ef486d1ba512c4bf352146aa08eb Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 11:02:09 +0800 Subject: [PATCH 039/218] =?UTF-8?q?=E5=AE=8C=E5=96=84answer=E4=BB=A3?= =?UTF-8?q?=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 34 +++++++++++++++++++--------------- webrtc/Sdp.h | 4 ++++ 2 files changed, 23 insertions(+), 15 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index eca66a61..0230f914 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -924,12 +924,14 @@ string RtcSession::toString() const{ sdp.items.emplace_back(std::make_shared >(session_info)); } sdp.items.emplace_back(std::make_shared(time)); - sdp.items.emplace_back(std::make_shared(connection)); + if(connection.empty()){ + sdp.items.emplace_back(std::make_shared(connection)); + } if (!bandwidth.empty()) { sdp.items.emplace_back(std::make_shared(bandwidth)); } - sdp.items.emplace_back(wrapSdpAttr(std::make_shared(msid_semantic))); sdp.items.emplace_back(wrapSdpAttr(std::make_shared(group))); + sdp.items.emplace_back(wrapSdpAttr(std::make_shared(msid_semantic))); for (auto &m : media) { sdp.medias.emplace_back(); auto &sdp_media = sdp.medias.back(); @@ -944,18 +946,22 @@ string RtcSession::toString() const{ mline->fmts.emplace_back(m.sctp_port); } sdp_media.items.emplace_back(std::move(mline)); - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.mid))); sdp_media.items.emplace_back(std::make_shared(m.addr)); + if (!m.rtcp_addr.empty()) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.rtcp_addr))); + } + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.ice_ufrag))); sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.ice_pwd))); - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.fingerprint))); - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.role))); if (m.ice_trickle || m.ice_renomination) { auto attr = std::make_shared(); attr->trickle = m.ice_trickle; attr->renomination = m.ice_renomination; sdp_media.items.emplace_back(wrapSdpAttr(attr)); } + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.fingerprint))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.role))); + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.mid))); if (m.ice_lite) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-lite"))); } @@ -965,9 +971,6 @@ string RtcSession::toString() const{ if (m.direction != RtpDirection::invalid) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.direction))); } - if (m.rtcp_addr.port) { - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.rtcp_addr))); - } if (m.rtcp_mux) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("rtcp-mux"))); } @@ -1145,7 +1148,7 @@ RtcMedia *RtcSession::getMedia(TrackType type){ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ enable = true; rtcp_mux = true; - rtcp_rsize = false; + rtcp_rsize = true; group_bundle = true; unified_plan = false; support_rtx = true; @@ -1238,11 +1241,9 @@ shared_ptr RtcConfigure::createOffer(){ shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ shared_ptr ret = std::make_shared(); ret->version = offer.version; - ret->origin.parse("- 0 0 IN IP4 0.0.0.0"); - ret->session_name = "zlmediakit_webrtc_session"; - ret->session_info = "zlmediakit_webrtc_session"; - ret->connection.parse("IN IP4 0.0.0.0"); - ret->msid_semantic.parse("WMS *"); + ret->origin = offer.origin; + ret->session_name = offer.session_name; + ret->msid_semantic = offer.msid_semantic; matchMedia(ret, TrackAudio, offer.media, audio); matchMedia(ret, TrackVideo, offer.media, video); matchMedia(ret, TrackApplication, offer.media, application); @@ -1283,19 +1284,22 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const answer_media.type = offer_media.type; answer_media.mid = offer_media.mid; answer_media.proto = offer_media.proto; + answer_media.port = offer_media.port; answer_media.rtcp_mux = offer_media.rtcp_mux && configure.rtcp_mux; answer_media.rtcp_rsize = offer_media.rtcp_rsize && configure.rtcp_rsize; + answer_media.rtcp_addr = offer_media.rtcp_addr; answer_media.ice_trickle = offer_media.ice_trickle && configure.ice_trickle; answer_media.ice_renomination = offer_media.ice_renomination && configure.ice_renomination; answer_media.ice_ufrag = configure.ice_ufrag; answer_media.ice_pwd = configure.ice_pwd; answer_media.fingerprint = configure.fingerprint; + answer_media.ice_lite = configure.ice_lite; switch (offer_media.role) { - case DtlsRole::actpass : case DtlsRole::active : { answer_media.role = DtlsRole::passive; break; } + case DtlsRole::actpass : case DtlsRole::passive : { answer_media.role = DtlsRole::active; break; diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index fa167fbc..895610ff 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -159,6 +159,7 @@ public: void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "c";} + bool empty() const {return address.empty();} }; class SdpBandwidth : public SdpItem { @@ -261,6 +262,9 @@ public: void parse(const string &str) override;; string toString() const override; const char* getKey() const override { return "rtcp";} + bool empty() const { + return address.empty() || !port; + } }; class SdpAttrIceUfrag : public SdpItem { From fd23a87cfb0eb2818bbefa7116d48b5efedadb47 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 11:42:40 +0800 Subject: [PATCH 040/218] =?UTF-8?q?answer=20sdp=E9=80=9A=E8=BF=87chrome?= =?UTF-8?q?=E6=A0=A1=E9=AA=8C?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 3 +- webrtc/WebRtcTransport.cpp | 2 +- webrtc/answer.sdp | 42 ++++++++++--------- webrtc/janus_answer.sdp | 69 +++++++++++++++++++++++++++++++ webrtc/janus_offer.sdp | 83 ++++++++++++++++++++++++++++++++++++++ www/webrtc/ZLMRTCClient.js | 16 ++++---- 6 files changed, 185 insertions(+), 30 deletions(-) create mode 100644 webrtc/janus_answer.sdp create mode 100644 webrtc/janus_offer.sdp diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 0230f914..ea0e8cf4 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1176,7 +1176,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ break; } case TrackApplication: { - enable = false; + enable = true; break; } default: break; @@ -1195,7 +1195,6 @@ void RtcConfigure::setDefaultSetting(string ice_ufrag, video.ice_pwd = audio.ice_pwd = application.ice_pwd = ice_pwd; video.direction = audio.direction = application.direction = direction; video.fingerprint = audio.fingerprint = application.fingerprint = fingerprint; - application.enable = false; } void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType type) { diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 4c43f97d..439b9c28 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -104,7 +104,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){ throw std::invalid_argument("支持group BUNDLE模式"); } auto str = _answer_sdp->toString(); - InfoL << str; + InfoL << "\r\n" << str; return str; } diff --git a/webrtc/answer.sdp b/webrtc/answer.sdp index 82df3355..32045375 100644 --- a/webrtc/answer.sdp +++ b/webrtc/answer.sdp @@ -1,33 +1,35 @@ - v=0 -o=- 0 0 IN IP4 0.0.0.0 -s=zlmediakit_webrtc_session -i=zlmediakit_webrtc_session +v=0 +o=- 2712902958023202213 2 IN IP4 127.0.0.1 +s=- t=0 0 -c=IN IP4 0.0.0.0 -a=msid-semantic: WMS * a=group:BUNDLE 0 1 -m=audio 0 UDP/TLS/RTP/SAVPF 111 -a=mid:0 +a=msid-semantic: WMS +m=audio 9 UDP/TLS/RTP/SAVPF 111 c=IN IP4 0.0.0.0 -a=ice-ufrag:PdK3 -a=ice-pwd:PdK31zZ1sKhPU2ACzP5av37f -a=fingerprint:sha-256 D1:4F:7B:70:86:99:BE:1E:7E:B8:C7:0F:13:BC:78:C4:6A:F9:2C:F6:F3:5E:42:11:0F:1C:74:18:F6:DD:95:75 -a=setup:passive +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:68ws +a=ice-pwd:68wscAOadU4rdMwZ4Ow14M5u a=ice-options:trickle +a=fingerprint:sha-256 BF:F8:C2:35:19:2E:C3:17:44:5A:BA:DF:08:C4:65:87:AB:0C:85:B9:91:69:80:8E:55:D9:2F:EF:5C:D6:F3:A2 +a=setup:active +a=mid:0 +a=ice-lite a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=recvonly a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 -m=video 0 UDP/TLS/RTP/SAVPF 102 121 -a=mid:1 +m=video 9 UDP/TLS/RTP/SAVPF 102 121 c=IN IP4 0.0.0.0 -a=ice-ufrag:PdK3 -a=ice-pwd:PdK31zZ1sKhPU2ACzP5av37f -a=fingerprint:sha-256 D1:4F:7B:70:86:99:BE:1E:7E:B8:C7:0F:13:BC:78:C4:6A:F9:2C:F6:F3:5E:42:11:0F:1C:74:18:F6:DD:95:75 -a=setup:passive +a=rtcp:9 IN IP4 0.0.0.0 +a=ice-ufrag:68ws +a=ice-pwd:68wscAOadU4rdMwZ4Ow14M5u a=ice-options:trickle +a=fingerprint:sha-256 BF:F8:C2:35:19:2E:C3:17:44:5A:BA:DF:08:C4:65:87:AB:0C:85:B9:91:69:80:8E:55:D9:2F:EF:5C:D6:F3:A2 +a=setup:active +a=mid:1 +a=ice-lite a=extmap:14 urn:ietf:params:rtp-hdrext:toffset a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:13 urn:3gpp:video-orientation @@ -35,6 +37,7 @@ a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extension a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=recvonly a=rtcp-mux +a=rtcp-rsize a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc @@ -43,4 +46,5 @@ a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:121 rtx/90000 -a=fmtp:121 apt=102 \ No newline at end of file +a=fmtp:121 apt=102 + diff --git a/webrtc/janus_answer.sdp b/webrtc/janus_answer.sdp new file mode 100644 index 00000000..345c3de7 --- /dev/null +++ b/webrtc/janus_answer.sdp @@ -0,0 +1,69 @@ +v=0 +o=mozilla...THIS_IS_SDPARTA-87.0 5715694060540081921 0 IN IP4 0.0.0.0 +s=- +t=0 0 +a=fingerprint:sha-256 9C:9C:D9:4B:2B:82:84:95:C2:23:BC:3A:0B:BE:3B:8B:D5:6A:DA:A6:B2:F4:43:5E:C1:07:48:F8:AF:CE:FC:3D +a=group:BUNDLE video data +a=ice-options:trickle +a=msid-semantic:WMS * +m=video 9 UDP/TLS/RTP/SAVPF 120 122 121 123 126 127 97 98 +c=IN IP4 0.0.0.0 +a=sendrecv +a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:5 urn:ietf:params:rtp-hdrext:toffset +a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1 +a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1 +a=fmtp:120 max-fs=12288;max-fr=60 +a=fmtp:122 apt=120 +a=fmtp:121 max-fs=12288;max-fr=60 +a=fmtp:123 apt=121 +a=fmtp:127 apt=126 +a=fmtp:98 apt=97 +a=ice-pwd:518cb4fc626f82bef2ada4e9221dfb50 +a=ice-ufrag:d5484fed +a=mid:video +a=msid:{9cd17d8b-f96d-4663-866b-72ddd3b8b54b} {3dcd399d-4933-41d5-89fd-87070cfc3cf6} +a=rtcp-fb:120 nack +a=rtcp-fb:120 nack pli +a=rtcp-fb:120 ccm fir +a=rtcp-fb:120 goog-remb +a=rtcp-fb:120 transport-cc +a=rtcp-fb:121 nack +a=rtcp-fb:121 nack pli +a=rtcp-fb:121 ccm fir +a=rtcp-fb:121 goog-remb +a=rtcp-fb:121 transport-cc +a=rtcp-fb:126 nack +a=rtcp-fb:126 nack pli +a=rtcp-fb:126 ccm fir +a=rtcp-fb:126 goog-remb +a=rtcp-fb:126 transport-cc +a=rtcp-fb:97 nack +a=rtcp-fb:97 nack pli +a=rtcp-fb:97 ccm fir +a=rtcp-fb:97 goog-remb +a=rtcp-fb:97 transport-cc +a=rtcp-mux +a=rtpmap:120 VP8/90000 +a=rtpmap:122 rtx/90000 +a=rtpmap:121 VP9/90000 +a=rtpmap:123 rtx/90000 +a=rtpmap:126 H264/90000 +a=rtpmap:127 rtx/90000 +a=rtpmap:97 H264/90000 +a=rtpmap:98 rtx/90000 +a=setup:active +a=ssrc:697402393 cname:{f1ba68ef-ad59-4fff-a1f3-dbbb0cda3118} +a=ssrc:258662392 cname:{f1ba68ef-ad59-4fff-a1f3-dbbb0cda3118} +a=ssrc-group:FID 697402393 258662392 +m=application 9 UDP/DTLS/SCTP webrtc-datachannel +c=IN IP4 0.0.0.0 +a=sendrecv +a=ice-pwd:518cb4fc626f82bef2ada4e9221dfb50 +a=ice-ufrag:d5484fed +a=mid:data +a=setup:active +a=sctp-port:5000 +a=max-message-size:1073741823 diff --git a/webrtc/janus_offer.sdp b/webrtc/janus_offer.sdp new file mode 100644 index 00000000..f4f454ed --- /dev/null +++ b/webrtc/janus_offer.sdp @@ -0,0 +1,83 @@ +v=0 +o=mozilla...THIS_IS_SDPARTA-87.0 1617247876363377 1 IN IP4 23.101.8.213 +s=- +t=0 0 +a=group:BUNDLE video data +a=msid-semantic: WMS janus +a=ice-lite +a=sendrecv +m=video 9 UDP/TLS/RTP/SAVPF 120 121 126 97 122 123 127 98 +c=IN IP4 23.101.8.213 +a=sendrecv +a=mid:video +a=rtcp-mux +a=ice-ufrag:ePgh +a=ice-pwd:ZURiB67/xs69E76aOa7JDw +a=ice-options:trickle +a=fingerprint:sha-256 EF:7A:50:9C:05:8C:EF:84:4D:72:B2:74:30:BA:FD:82:76:D1:C3:FE:0C:A0:10:43:B8:6C:B2:ED:B3:F7:77:8B +a=setup:actpass +a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:5 urn:ietf:params:rtp-hdrext:toffset +a=extmap:6/recvonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1 +a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1 +a=fmtp:120 max-fs=12288;max-fr=60 +a=fmtp:121 max-fs=12288;max-fr=60 +a=rtcp-fb:120 nack +a=rtcp-fb:120 nack pli +a=rtcp-fb:120 ccm fir +a=rtcp-fb:120 goog-remb +a=rtcp-fb:120 transport-cc +a=rtcp-fb:121 nack +a=rtcp-fb:121 nack pli +a=rtcp-fb:121 ccm fir +a=rtcp-fb:121 goog-remb +a=rtcp-fb:121 transport-cc +a=rtcp-fb:126 nack +a=rtcp-fb:126 nack pli +a=rtcp-fb:126 ccm fir +a=rtcp-fb:126 goog-remb +a=rtcp-fb:126 transport-cc +a=rtcp-fb:97 nack +a=rtcp-fb:97 nack pli +a=rtcp-fb:97 ccm fir +a=rtcp-fb:97 goog-remb +a=rtcp-fb:97 transport-cc +a=rtpmap:120 VP8/90000 +a=rtpmap:121 VP9/90000 +a=rtpmap:126 H264/90000 +a=rtpmap:97 H264/90000 +a=rtpmap:122 rtx/90000 +a=fmtp:122 apt=120 +a=rtpmap:123 rtx/90000 +a=fmtp:123 apt=121 +a=rtpmap:127 rtx/90000 +a=fmtp:127 apt=126 +a=rtpmap:98 rtx/90000 +a=fmtp:98 apt=97 +a=ssrc-group:FID 3581519470 2258376012 +a=msid:janus janusv0 +a=ssrc:3581519470 cname:janus +a=ssrc:3581519470 msid:janus janusv0 +a=ssrc:3581519470 mslabel:janus +a=ssrc:3581519470 label:janusv0 +a=ssrc:2258376012 cname:janus +a=ssrc:2258376012 msid:janus janusv0 +a=ssrc:2258376012 mslabel:janus +a=ssrc:2258376012 label:janusv0 +a=candidate:1 1 udp 2013266431 23.101.8.213 41901 typ host +a=end-of-candidates +m=application 9 UDP/DTLS/SCTP webrtc-datachannel +c=IN IP4 23.101.8.213 +a=sendrecv +a=sctp-port:5000 +a=mid:data +a=ice-ufrag:ePgh +a=ice-pwd:ZURiB67/xs69E76aOa7JDw +a=ice-options:trickle +a=fingerprint:sha-256 EF:7A:50:9C:05:8C:EF:84:4D:72:B2:74:30:BA:FD:82:76:D1:C3:FE:0C:A0:10:43:B8:6C:B2:ED:B3:F7:77:8B +a=setup:actpass +a=candidate:1 1 udp 2013266431 23.101.8.213 41901 typ host +a=end-of-candidates diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js index 76557c32..b1e049ab 100644 --- a/www/webrtc/ZLMRTCClient.js +++ b/www/webrtc/ZLMRTCClient.js @@ -4,7 +4,7 @@ var ZLMRTCClient = (function (exports) { const Events$1 = { WEBRTC_NOT_SUPPORT: 'WEBRTC_NOT_SUPPORT', WEBRTC_ICE_CANDIDATE_ERROR: 'WEBRTC_ICE_CANDIDATE_ERROR', - WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED: 'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED', + WEBRTC_OFFER_answer_EXCHANGE_FAILED: 'WEBRTC_OFFER_answer_EXCHANGE_FAILED', WEBRTC_ON_REMOTE_STREAMS: 'WEBRTC_ON_REMOTE_STREAMS', WEBRTC_ON_LOCAL_STREAM: 'WEBRTC_ON_LOCAL_STREAM' }; @@ -7360,16 +7360,16 @@ var ZLMRTCClient = (function (exports) { let ret = response.data; if (ret.code != 0) { - // mean failed for offer/anwser exchange - this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); + // mean failed for offer/answer exchange + this.dispatch(Events$1.WEBRTC_OFFER_answer_EXCHANGE_FAILED, ret); return; } - let anwser = {}; - anwser.sdp = ret.sdp; - anwser.type = 'anwser'; - log(this.TAG, 'anwser:', ret.sdp); - this.pc.setRemoteDescription(anwser).then(() => { + let answer = {}; + answer.sdp = ret.sdp; + answer.type = 'answer'; + log(this.TAG, 'answer:', ret.sdp); + this.pc.setRemoteDescription(answer).then(() => { log(this.TAG, 'set remote sucess'); }).catch(e => { error(this.TAG, e); From 6a95562dca570f289a6cd0e2d52e53b463f3d136 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 11:59:35 +0800 Subject: [PATCH 041/218] =?UTF-8?q?=E7=A8=8D=E5=BE=AE=E8=B0=83=E6=95=B4?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 4 +-- webrtc/WebRtcTransport.cpp | 60 ++++++++++++-------------------------- webrtc/WebRtcTransport.h | 2 -- 3 files changed, 20 insertions(+), 46 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index ea0e8cf4..173299d4 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1148,7 +1148,7 @@ RtcMedia *RtcSession::getMedia(TrackType type){ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ enable = true; rtcp_mux = true; - rtcp_rsize = true; + rtcp_rsize = false; group_bundle = true; unified_plan = false; support_rtx = true; @@ -1176,7 +1176,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ break; } case TrackApplication: { - enable = true; + enable = false; break; } default: break; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 439b9c28..2cada4c4 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -103,53 +103,29 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){ } else { throw std::invalid_argument("支持group BUNDLE模式"); } + + SdpAttrCandidate candidate; + candidate.foundation = "udpcandidate"; + candidate.component = 1; + candidate.transport = "udp"; + candidate.priority = getSSRC(); + candidate.address = getIP(); + candidate.port = getPort(); + candidate.type = "host"; + for (auto &m :_answer_sdp->media) { + m.candidate.emplace_back(candidate); + m.port = getPort(); + m.rtcp_addr.address = getIP(); + m.rtcp_addr.port = getPort(); + m.addr.address = getIP(); + } + _answer_sdp->connection.address = getIP(); + auto str = _answer_sdp->toString(); InfoL << "\r\n" << str; return str; } -std::string WebRtcTransport::getOfferSdp() { - RTC::DtlsTransport::Fingerprint remote_fingerprint; - remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm("sha-256"); - remote_fingerprint.value = ""; - dtls_transport_->SetRemoteFingerprint(remote_fingerprint); - - char sdp[1024 * 10] = {0}; - auto ssrc = getSSRC(); - auto ip = getIP(); - auto pt = getPayloadType(); - auto port = getPort(); - sprintf(sdp, - "v=0\r\n" - "o=- 1495799811084970 1495799811084970 IN IP4 %s\r\n" - "s=Streaming Test\r\n" - "t=0 0\r\n" - "a=group:BUNDLE video\r\n" - "a=msid-semantic: WMS janus\r\n" - "m=video %u RTP/SAVPF %u\r\n" - "c=IN IP4 %s\r\n" - "a=mid:video\r\n" - "a=sendonly\r\n" - "a=rtcp-mux\r\n" - "a=ice-lite\r\n" - "a=ice-ufrag:%s\r\n" - "a=ice-pwd:%s\r\n" - "a=ice-options:trickle\r\n" - "a=fingerprint:sha-256 %s\r\n" - "a=setup:actpass\r\n" - "a=connection:new\r\n" - "a=rtpmap:%u H264/90000\r\n" - "a=ssrc:%u cname:janusvideo\r\n" - "a=ssrc:%u msid:janus janusv0\r\n" - "a=ssrc:%u mslabel:janus\r\n" - "a=ssrc:%u label:janusv0\r\n" - "a=candidate:%s 1 udp %u %s %u typ %s\r\n", - ip.c_str(), port, pt, ip.c_str(), - ice_server_->GetUsernameFragment().c_str(),ice_server_->GetPassword().c_str(), - "", pt, ssrc, ssrc, ssrc, ssrc, "4", ssrc, ip.c_str(), port, "host"); - return sdp; -} - bool is_dtls(char *buf) { return ((*buf > 19) && (*buf < 64)); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 67a2083a..3a26b07b 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -20,8 +20,6 @@ public: std::string getAnswerSdp(const string &offer); - std::string getOfferSdp(); - /// 收到udp数据 /// \param buf /// \param len From 9ffa5c2135f593912c25009f166c62b5eae76f9a Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 14:07:48 +0800 Subject: [PATCH 042/218] =?UTF-8?q?=E6=81=A2=E5=A4=8D=E6=A3=80=E6=9F=A5dtl?= =?UTF-8?q?s=E8=AF=81=E4=B9=A6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/DtlsTransport.cpp | 5 ----- 1 file changed, 5 deletions(-) diff --git a/webrtc/DtlsTransport.cpp b/webrtc/DtlsTransport.cpp index 950f9620..f627fd29 100644 --- a/webrtc/DtlsTransport.cpp +++ b/webrtc/DtlsTransport.cpp @@ -1180,13 +1180,8 @@ namespace RTC "fingerprint in the remote certificate (%s) does not match the announced one (%s)", hexFingerprint, this->remoteFingerprint.value.c_str()); - - //todo 先屏蔽检查客户端签名 -#if 0 X509_free(certificate); - return false; -#endif } MS_DEBUG_TAG(dtls, "valid remote fingerprint"); From 1978748f2ec633736b4af95fe45e5ffb5eb3bdef Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 1 Apr 2021 14:16:42 +0800 Subject: [PATCH 043/218] =?UTF-8?q?=E6=8E=A5=E6=94=B6=E5=88=B0rtp=E5=8C=85?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 2 +- webrtc/WebRtcTransport.cpp | 28 ++++++++++++++-------------- 2 files changed, 15 insertions(+), 15 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 173299d4..1fe3f987 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1294,11 +1294,11 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const answer_media.fingerprint = configure.fingerprint; answer_media.ice_lite = configure.ice_lite; switch (offer_media.role) { + case DtlsRole::actpass : case DtlsRole::active : { answer_media.role = DtlsRole::passive; break; } - case DtlsRole::actpass : case DtlsRole::passive : { answer_media.role = DtlsRole::active; break; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 2cada4c4..e82029f9 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -24,6 +24,10 @@ void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) { InfoL; +} + +void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) { + InfoL; if (_answer_sdp->media[0].role == DtlsRole::passive) { dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); } else { @@ -31,10 +35,6 @@ void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) { } } -void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) { - InfoL; -} - void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) { InfoL; } @@ -141,16 +141,6 @@ bool is_rtcp(char *buf) { } void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple) { - if (is_rtp(buf)) { - RtpHeader *header = (RtpHeader *) buf; - InfoL << "rtp:" << header->dumpString(len); - return; - } - if (is_rtcp(buf)) { - RtcpHeader *header = (RtcpHeader *) buf; -// InfoL << "rtcp:" << header->dumpString(); - return; - } if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len); if (packet == nullptr) { @@ -164,6 +154,16 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTup dtls_transport_->ProcessDtlsData((uint8_t *) buf, len); return; } + if (is_rtp(buf)) { + RtpHeader *header = (RtpHeader *) buf; + InfoL << "rtp:" << header->dumpString(len); + return; + } + if (is_rtcp(buf)) { + RtcpHeader *header = (RtcpHeader *) buf; +// InfoL << "rtcp:" << header->dumpString(); + return; + } } void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { From 7110dc75ae8237be1a714cb8dbf729273470e4a0 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Fri, 2 Apr 2021 17:08:11 +0800 Subject: [PATCH 044/218] =?UTF-8?q?=E6=95=B4=E7=90=86=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 7 ++ webrtc/Sdp.h | 2 + webrtc/WebRtcTransport.cpp | 132 +++++++++++++++++++------------------ webrtc/WebRtcTransport.h | 108 +++++++++++++++++------------- 4 files changed, 142 insertions(+), 107 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 010aa5af..e96c660c 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1237,6 +1237,13 @@ shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ if (ret->media.empty()) { throw std::invalid_argument("生成的answer sdp中媒体个数为0"); } + + //设置音视频端口复用 + if (!offer.group.mids.empty()) { + for (auto &m : ret->media) { + ret->group.mids.emplace_back(m.mid); + } + } return ret; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 4d81c8fb..fa3b193c 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -437,6 +437,7 @@ public: class SdpAttrCandidate : public SdpItem { public: + using Ptr = std::shared_ptr; //https://tools.ietf.org/html/rfc5245 //15.1. "candidate" Attribute //a=candidate:4 1 udp 2 192.168.1.7 58107 typ host @@ -656,6 +657,7 @@ public: class RtcConfigure { public: + using Ptr = std::shared_ptr; class RtcTrackConfigure { public: bool enable; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9468e3d1..f380603d 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -3,19 +3,19 @@ #include "Rtcp/Rtcp.h" WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { - dtls_transport_ = std::make_shared(poller, this); - ice_server_ = std::make_shared(this, makeRandStr(4), makeRandStr(24)); + _dtls_transport = std::make_shared(poller, this); + _ice_server = std::make_shared(this, makeRandStr(4), makeRandStr(24)); } void WebRtcTransport::onDestory(){ - dtls_transport_ = nullptr; - ice_server_ = nullptr; + _dtls_transport = nullptr; + _ice_server = nullptr; } ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) { - onWrite((char *)packet->GetData(), packet->GetSize(), (struct sockaddr_in *)tuple); + onSendSockData((char *) packet->GetData(), packet->GetSize(), (struct sockaddr_in *) tuple); } void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) { @@ -29,9 +29,9 @@ void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) { void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) { InfoL; if (_answer_sdp->media[0].role == DtlsRole::passive) { - dtls_transport_->Run(RTC::DtlsTransport::Role::SERVER); + _dtls_transport->Run(RTC::DtlsTransport::Role::SERVER); } else { - dtls_transport_->Run(RTC::DtlsTransport::Role::CLIENT); + _dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT); } } @@ -50,21 +50,21 @@ void WebRtcTransport::OnDtlsTransportConnected( size_t srtpRemoteKeyLen, std::string &remoteCert) { InfoL; - srtp_session_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); - srtp_session_recv_ = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen); - onDtlsConnected(); + _srtp_session_send = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); + _srtp_session_recv = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen); + onStartWebRTC(); } void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) { - onWrite((char *)data, len); + onSendSockData((char *)data, len); } ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -void WebRtcTransport::onWrite(const char *buf, size_t len){ - auto tuple = ice_server_->GetSelectedTuple(); +void WebRtcTransport::onSendSockData(const char *buf, size_t len){ + auto tuple = _ice_server->GetSelectedTuple(); assert(tuple); - onWrite(buf, len, (struct sockaddr_in *)tuple); + onSendSockData(buf, len, (struct sockaddr_in *) tuple); } string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ @@ -77,46 +77,47 @@ string getFingerprint(const string &algorithm_str, const std::shared_ptr(); - _offer_sdp->loadFrom(offer); - - SdpAttrFingerprint fingerprint; - fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; - fingerprint.hash = getFingerprint(fingerprint.algorithm, dtls_transport_); - - RtcConfigure configure; - configure.setDefaultSetting(ice_server_->GetUsernameFragment(), ice_server_->GetPassword(), RtpDirection::recvonly, fingerprint); - - SdpAttrCandidate candidate; - candidate.foundation = "udpcandidate"; - candidate.component = 1; - candidate.transport = "udp"; - candidate.priority = getSSRC(); - candidate.address = getIP(); - candidate.port = getPort(); - candidate.type = "host"; - configure.addCandidate(candidate); - - _answer_sdp = configure.createAnswer(*_offer_sdp); - +void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){ //设置远端dtls签名 RTC::DtlsTransport::Fingerprint remote_fingerprint; remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm); remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash; - dtls_transport_->SetRemoteFingerprint(remote_fingerprint); + _dtls_transport->SetRemoteFingerprint(remote_fingerprint); +} - if (!_offer_sdp->group.mids.empty()) { - for (auto &m : _answer_sdp->media) { - _answer_sdp->group.mids.emplace_back(m.mid); +void WebRtcTransport::onCheckSdp(SdpType type, const RtcSession &sdp) const{ + for (auto &m : sdp.media) { + if (m.type != TrackApplication && !m.rtcp_mux) { + throw std::invalid_argument("只支持rtcp-mux模式"); } - } else { - throw std::invalid_argument("支持group BUNDLE模式"); } + if (sdp.group.mids.empty()) { + throw std::invalid_argument("只支持group BUNDLE模式"); + } +} + +std::string WebRtcTransport::getAnswerSdp(const string &offer){ + //// 解析offer sdp //// + _offer_sdp = std::make_shared(); + _offer_sdp->loadFrom(offer); + onCheckSdp(SdpType::offer, *_offer_sdp); + setRemoteDtlsFingerprint(*_offer_sdp); + + //// sdp 配置 //// + SdpAttrFingerprint fingerprint; + fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; + fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport); + RtcConfigure configure; + configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::recvonly, fingerprint); + configure.addCandidate(*getIceCandidate()); + onRtcConfigure(configure); + + //// 生成answer sdp //// + _answer_sdp = configure.createAnswer(*_offer_sdp); + onCheckSdp(SdpType::answer, *_answer_sdp); auto str = _answer_sdp->toString(); - InfoL << "\r\n" << str; + TraceL << "\r\n" << str; return str; } @@ -134,18 +135,18 @@ bool is_rtcp(char *buf) { return ((header->pt >= 64) && (header->pt < 96)); } -void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple) { +void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple) { if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len); if (packet == nullptr) { WarnL << "parse stun error" << std::endl; return; } - ice_server_->ProcessStunPacket(packet, tuple); + _ice_server->ProcessStunPacket(packet, tuple); return; } if (is_dtls(buf)) { - dtls_transport_->ProcessDtlsData((uint8_t *) buf, len); + _dtls_transport->ProcessDtlsData((uint8_t *) buf, len); return; } if (is_rtp(buf)) { @@ -158,14 +159,14 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTup } } -void WebRtcTransport::WritRtpPacket(char *buf, size_t len) { +void WebRtcTransport::sendRtpPacket(char *buf, size_t len) { const uint8_t *p = (uint8_t *) buf; bool ret = false; - if (srtp_session_) { - ret = srtp_session_->EncryptRtp(&p, &len); + if (_srtp_session_send) { + ret = _srtp_session_send->EncryptRtp(&p, &len); } if (ret) { - onWrite((char *) p, len); + onSendSockData((char *) p, len); } } @@ -183,7 +184,7 @@ WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcT //随机端口，绑定全部网卡 _socket->bindUdpSock(0); _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable { - OnInputDataPacket(buf->data(), buf->size(), addr); + inputSockData(buf->data(), buf->size(), addr); }); } @@ -196,7 +197,7 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { _src = src; } -void WebRtcTransportImp::onDtlsConnected() { +void WebRtcTransportImp::onStartWebRTC() { _reader = _src->getRing()->attach(_socket->getPoller(), true); weak_ptr weak_self = shared_from_this(); _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt){ @@ -207,14 +208,14 @@ void WebRtcTransportImp::onDtlsConnected() { pkt->for_each([&](const RtpPacket::Ptr &rtp) { if(rtp->type == TrackVideo) { //目前只支持视频 - strongSelf->WritRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, + strongSelf->sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize); } }); }); } -void WebRtcTransportImp::onWrite(const char *buf, size_t len, struct sockaddr_in *dst) { +void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) { auto ptr = BufferRaw::create(); ptr->assign(buf, len); _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr)); @@ -224,13 +225,6 @@ uint32_t WebRtcTransportImp::getSSRC() const { return _src->getSsrc(TrackVideo); } -int WebRtcTransportImp::getPayloadType() const{ - auto sdp = SdpParser(_src->getSdp()); - auto track = sdp.getTrack(TrackVideo); - assert(track); - return track ? track->_pt : 0; -} - uint16_t WebRtcTransportImp::getPort() const { //todo udp端口号应该与外网映射端口相同 return _socket->get_local_port(); @@ -241,6 +235,18 @@ std::string WebRtcTransportImp::getIP() const { return SockUtil::get_local_ip(); } +SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ + auto candidate = std::make_shared(); + candidate->foundation = "udpcandidate"; + candidate->component = 1; + candidate->transport = "udp"; + candidate->priority = 100; + candidate->address = getIP(); + candidate->port = getPort(); + candidate->type = "host"; + return candidate; +} + /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 3a26b07b..f37bea0b 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -2,12 +2,16 @@ #include #include - #include "DtlsTransport.hpp" #include "IceServer.hpp" #include "SrtpSession.hpp" #include "StunPacket.hpp" #include "Sdp.h" +#include "Poller/EventPoller.h" +#include "Network/Socket.h" +#include "Rtsp/RtspMediaSource.h" +using namespace toolkit; +using namespace mediakit; class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: @@ -15,33 +19,43 @@ public: WebRtcTransport(const EventPoller::Ptr &poller); ~WebRtcTransport() override = default; - /// 销毁对象 + /** + * 消费对象 + */ virtual void onDestory(); + /** + * 创建webrtc answer sdp + * @param offer offer sdp + * @return answer sdp + */ std::string getAnswerSdp(const string &offer); - /// 收到udp数据 - /// \param buf - /// \param len - /// \param remote_address - void OnInputDataPacket(char *buf, size_t len, RTC::TransportTuple *tuple); + /** + * socket收到udp数据 + * @param buf 数据指针 + * @param len 数据长度 + * @param tuple 数据来源 + */ + void inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple); - /// 发送rtp - /// \param buf - /// \param len - void WritRtpPacket(char *buf, size_t len); + /** + * 发送rtp + * @param buf rtcp内容 + * @param len rtcp长度 + */ + void sendRtpPacket(char *buf, size_t len); protected: - // dtls相关的回调 + //// dtls相关的回调 //// void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override {}; - void OnDtlsTransportConnected( - const RTC::DtlsTransport *dtlsTransport, - RTC::SrtpSession::CryptoSuite srtpCryptoSuite, - uint8_t *srtpLocalKey, - size_t srtpLocalKeyLen, - uint8_t *srtpRemoteKey, - size_t srtpRemoteKeyLen, - std::string &remoteCert) override; + void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, + RTC::SrtpSession::CryptoSuite srtpCryptoSuite, + uint8_t *srtpLocalKey, + size_t srtpLocalKeyLen, + uint8_t *srtpRemoteKey, + size_t srtpRemoteKeyLen, + std::string &remoteCert) override; void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override {}; void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override {}; @@ -49,7 +63,7 @@ protected: void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override {}; protected: - //ice相关的回调 + //// ice相关的回调 /// void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override; void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override; void OnIceServerConnected(const RTC::IceServer *iceServer) override; @@ -57,51 +71,57 @@ protected: void OnIceServerDisconnected(const RTC::IceServer *iceServer) override; protected: - /// 输出udp数据 - /// \param buf - /// \param len - /// \param dst - virtual void onWrite(const char *buf, size_t len, struct sockaddr_in *dst) = 0; + virtual uint32_t getSSRC() const = 0; virtual uint16_t getPort() const = 0; virtual std::string getIP() const = 0; - virtual int getPayloadType() const = 0; - virtual void onDtlsConnected() = 0; + virtual void onStartWebRTC() = 0; + virtual void onRtcConfigure(RtcConfigure &configure) const {} + virtual void onCheckSdp(SdpType type, const RtcSession &sdp) const; + + virtual SdpAttrCandidate::Ptr getIceCandidate() const = 0; + virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) = 0; private: - void onWrite(const char *buf, size_t len); + void onSendSockData(const char *buf, size_t len); + void setRemoteDtlsFingerprint(const RtcSession &remote); private: - std::shared_ptr ice_server_; - std::shared_ptr dtls_transport_; - std::shared_ptr srtp_session_; - std::shared_ptr srtp_session_recv_; + std::shared_ptr _ice_server; + std::shared_ptr _dtls_transport; + std::shared_ptr _srtp_session_send; + std::shared_ptr _srtp_session_recv; RtcSession::Ptr _offer_sdp; RtcSession::Ptr _answer_sdp; }; -#include "Poller/EventPoller.h" -#include "Network/Socket.h" -#include "Rtsp/RtspMediaSource.h" -using namespace toolkit; -using namespace mediakit; - class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; - - static Ptr create(const EventPoller::Ptr &poller); ~WebRtcTransportImp() override = default; + /** + * 创建WebRTC对象 + * @param poller 改对象需要绑定的线程 + * @return 对象 + */ + static Ptr create(const EventPoller::Ptr &poller); + + /** + * 绑定rtsp媒体源 + * @param src 媒体源 + */ void attach(const RtspMediaSource::Ptr &src); protected: - void onWrite(const char *buf, size_t len, struct sockaddr_in *dst) override; - int getPayloadType() const override; + void onStartWebRTC() override; + void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) override; uint32_t getSSRC() const override; uint16_t getPort() const override; std::string getIP() const override; - void onDtlsConnected() override; + SdpAttrCandidate::Ptr getIceCandidate() const override; + +private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onDestory() override; From ee072191e0fa95889e8a4dae7342b3f3f0893153 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Fri, 2 Apr 2021 17:56:24 +0800 Subject: [PATCH 045/218] =?UTF-8?q?=E5=BC=80=E5=A7=8B=E5=AF=B9=E6=8E=A5rts?= =?UTF-8?q?p=E6=BA=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 78 +++++++++++++++++++++++--------------- webrtc/WebRtcTransport.h | 25 ++++++------ 2 files changed, 61 insertions(+), 42 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index f380603d..8cecdfaf 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -61,10 +61,10 @@ void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTran ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -void WebRtcTransport::onSendSockData(const char *buf, size_t len){ +void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){ auto tuple = _ice_server->GetSelectedTuple(); assert(tuple); - onSendSockData(buf, len, (struct sockaddr_in *) tuple); + onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush); } string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ @@ -85,7 +85,7 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){ _dtls_transport->SetRemoteFingerprint(remote_fingerprint); } -void WebRtcTransport::onCheckSdp(SdpType type, const RtcSession &sdp) const{ +void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp) const{ for (auto &m : sdp.media) { if (m.type != TrackApplication && !m.rtcp_mux) { throw std::invalid_argument("只支持rtcp-mux模式"); @@ -150,23 +150,27 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * return; } if (is_rtp(buf)) { - RtpHeader *header = (RtpHeader *) buf; + if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) { + onRtp(buf, len); + } return; } if (is_rtcp(buf)) { - RtcpHeader *header = (RtcpHeader *) buf; + if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) { + onRtcp(buf, len); + } return; } } -void WebRtcTransport::sendRtpPacket(char *buf, size_t len) { +void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush) { const uint8_t *p = (uint8_t *) buf; bool ret = false; if (_srtp_session_send) { ret = _srtp_session_send->EncryptRtp(&p, &len); } if (ret) { - onSendSockData((char *) p, len); + onSendSockData((char *) p, len, flush); } } @@ -205,34 +209,38 @@ void WebRtcTransportImp::onStartWebRTC() { if (!strongSelf) { return; } + size_t i = 0; pkt->for_each([&](const RtpPacket::Ptr &rtp) { - if(rtp->type == TrackVideo) { - //目前只支持视频 - strongSelf->sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, - rtp->size() - RtpPacket::kRtpTcpHeaderSize); - } + strongSelf->onSendRtp(rtp, ++i == pkt->size()); }); }); } -void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) { +void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ + InfoL << flush; +} + +void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ + WebRtcTransport::onCheckSdp(type, sdp); + if (type != SdpType::answer) { + return; + } + for (auto &m : sdp.media) { + if (m.type == TrackApplication) { + continue; + } + m.rtp_ssrc.ssrc = _src->getSsrc(m.type); + m.rtx_ssrc.ssrc = 2 + m.rtp_ssrc.ssrc; + + m.rtp_ssrc.cname = "zlmediakit rtc"; + m.rtx_ssrc.cname = "zlmediakit rtc"; + } +} + +void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) { auto ptr = BufferRaw::create(); ptr->assign(buf, len); - _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr)); -} - -uint32_t WebRtcTransportImp::getSSRC() const { - return _src->getSsrc(TrackVideo); -} - -uint16_t WebRtcTransportImp::getPort() const { - //todo udp端口号应该与外网映射端口相同 - return _socket->get_local_port(); -} - -std::string WebRtcTransportImp::getIP() const { - //todo 替换为外网ip - return SockUtil::get_local_ip(); + _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush); } SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ @@ -241,12 +249,22 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ candidate->component = 1; candidate->transport = "udp"; candidate->priority = 100; - candidate->address = getIP(); - candidate->port = getPort(); + candidate->address = SockUtil::get_local_ip(); + candidate->port = _socket->get_local_port(); candidate->type = "host"; return candidate; } +void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + RtpHeader *rtp = (RtpHeader *) buf; +// TraceL << (int)rtp->ssrc; +} + +void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { + RtcpHeader *rtcp = (RtcpHeader *) buf; +// TraceL << rtcpTypeToStr((RtcpType)rtcp->pt); +} + /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index f37bea0b..a337bf89 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -44,7 +44,7 @@ public: * @param buf rtcp内容 * @param len rtcp长度 */ - void sendRtpPacket(char *buf, size_t len); + void sendRtpPacket(char *buf, size_t len, bool flush); protected: //// dtls相关的回调 //// @@ -71,19 +71,18 @@ protected: void OnIceServerDisconnected(const RTC::IceServer *iceServer) override; protected: - - virtual uint32_t getSSRC() const = 0; - virtual uint16_t getPort() const = 0; - virtual std::string getIP() const = 0; virtual void onStartWebRTC() = 0; virtual void onRtcConfigure(RtcConfigure &configure) const {} - virtual void onCheckSdp(SdpType type, const RtcSession &sdp) const; + virtual void onCheckSdp(SdpType type, RtcSession &sdp) const; virtual SdpAttrCandidate::Ptr getIceCandidate() const = 0; - virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) = 0; + virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0; + + virtual void onRtp(const char *buf, size_t len) = 0; + virtual void onRtcp(const char *buf, size_t len) = 0; private: - void onSendSockData(const char *buf, size_t len); + void onSendSockData(const char *buf, size_t len, bool flush = true); void setRemoteDtlsFingerprint(const RtcSession &remote); private: @@ -115,15 +114,17 @@ public: protected: void onStartWebRTC() override; - void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst) override; - uint32_t getSSRC() const override; - uint16_t getPort() const override; - std::string getIP() const override; + void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override; + void onCheckSdp(SdpType type, RtcSession &sdp) const override; + SdpAttrCandidate::Ptr getIceCandidate() const override; + void onRtp(const char *buf, size_t len) override; + void onRtcp(const char *buf, size_t len) override; private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onDestory() override; + void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); private: Socket::Ptr _socket; From 9a2f2cbf2ed788019a49ed2591c3ad20aedf8e6b Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Fri, 2 Apr 2021 18:28:01 +0800 Subject: [PATCH 046/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtsp=20sdp=E5=8C=B9?= =?UTF-8?q?=E9=85=8D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 29 +++++++++++++++++++---------- webrtc/Sdp.h | 2 +- webrtc/WebRtcTransport.cpp | 32 +++++++++++++++++++++++++++++++- webrtc/WebRtcTransport.h | 5 ++--- 4 files changed, 53 insertions(+), 15 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index e96c660c..bc573e59 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -4,7 +4,9 @@ #include "Sdp.h" #include "Common/Parser.h" +#include "Rtsp/Rtsp.h" #include +using namespace mediakit; using onCreateSdpItem = function; static map sdpItemCreator; @@ -596,11 +598,11 @@ void SdpAttrFmtp::parse(const string &str) { auto vec = split(str.substr(pos + 1), ";"); for (auto &item : vec) { trim(item); - auto pr_vec = split(item, "="); - if (pr_vec.size() != 2) { + auto pos = item.find('='); + if(pos == string::npos){ SDP_THROW(); } - arr.emplace_back(std::make_pair(pr_vec[0], pr_vec[1])); + arr.emplace_back(std::make_pair(item.substr(0, pos), item.substr(pos + 1))); } if (arr.empty()) { SDP_THROW(); @@ -746,7 +748,7 @@ string SdpAttrCandidate::toString() const { return SdpItem::toString(); } -void RtcSession::loadFrom(const string &str) { +void RtcSession::loadFrom(const string &str, bool check) { RtcSessionSdp sdp; sdp.parse(str); @@ -887,12 +889,17 @@ void RtcSession::loadFrom(const string &str) { auto &plan = rtc_media.plan.back(); auto rtpmap_it = rtpmap_map.find(pt); if (rtpmap_it == rtpmap_map.end()) { - throw std::invalid_argument(StrPrinter << "该pt不存在相对于的a=rtpmap:" << pt); + plan.pt = pt; + plan.codec = RtpPayload::getCodecId(pt); + plan.sample_rate = RtpPayload::getClockRate(pt); + plan.channel = RtpPayload::getAudioChannel(pt); + } else { + plan.pt = rtpmap_it->second.pt; + plan.codec = rtpmap_it->second.codec; + plan.sample_rate = rtpmap_it->second.sample_rate; + plan.channel = rtpmap_it->second.channel; } - plan.pt = rtpmap_it->second.pt; - plan.codec = rtpmap_it->second.codec; - plan.sample_rate = rtpmap_it->second.sample_rate; - plan.channel = rtpmap_it->second.channel; + auto fmtp_it = fmtp_map.find(pt); if (fmtp_it != fmtp_map.end()) { plan.fmtp = fmtp_it->second.arr; @@ -905,7 +912,9 @@ void RtcSession::loadFrom(const string &str) { } group = sdp.getItemClass('a', "group"); - checkValid(); + if (check) { + checkValid(); + } } std::shared_ptr wrapSdpAttr(SdpItem::Ptr item){ diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index fa3b193c..c6720618 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -649,7 +649,7 @@ public: vector media; SdpAttrGroup group; - void loadFrom(const string &sdp); + void loadFrom(const string &sdp, bool check = true); void checkValid() const; string toString() const; RtcMedia *getMedia(TrackType type); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 8cecdfaf..43fbebd9 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -109,7 +109,6 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){ fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport); RtcConfigure configure; configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::recvonly, fingerprint); - configure.addCandidate(*getIceCandidate()); onRtcConfigure(configure); //// 生成answer sdp //// @@ -217,6 +216,7 @@ void WebRtcTransportImp::onStartWebRTC() { } void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ + //需要修改pt InfoL << flush; } @@ -237,6 +237,36 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ } } +void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { + WebRtcTransport::onRtcConfigure(configure); + + RtcSession sdp; + sdp.loadFrom(_src->getSdp(), false); + + configure.audio.enable = false; + configure.video.enable = false; + + for (auto &m : sdp.media) { + switch (m.type) { + case TrackVideo: { + configure.video.enable = true; + configure.video.preferred_codec = {getCodecId(m.plan[0].codec)}; + break; + } + case TrackAudio: { + configure.audio.enable = true; + configure.audio.preferred_codec = {getCodecId(m.plan[0].codec)}; + break; + } + default: + break; + } + } + + configure.addCandidate(*getIceCandidate()); +} + + void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) { auto ptr = BufferRaw::create(); ptr->assign(buf, len); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index a337bf89..271d1bbf 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -74,8 +74,6 @@ protected: virtual void onStartWebRTC() = 0; virtual void onRtcConfigure(RtcConfigure &configure) const {} virtual void onCheckSdp(SdpType type, RtcSession &sdp) const; - - virtual SdpAttrCandidate::Ptr getIceCandidate() const = 0; virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0; virtual void onRtp(const char *buf, size_t len) = 0; @@ -116,8 +114,8 @@ protected: void onStartWebRTC() override; void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override; void onCheckSdp(SdpType type, RtcSession &sdp) const override; + void onRtcConfigure(RtcConfigure &configure) const override; - SdpAttrCandidate::Ptr getIceCandidate() const override; void onRtp(const char *buf, size_t len) override; void onRtcp(const char *buf, size_t len) override; @@ -125,6 +123,7 @@ private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onDestory() override; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); + SdpAttrCandidate::Ptr getIceCandidate() const; private: Socket::Ptr _socket; From 4a62314be97966b462a9137e034bd030e8d87376 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Fri, 2 Apr 2021 18:36:33 +0800 Subject: [PATCH 047/218] =?UTF-8?q?=E5=AE=8C=E5=96=84?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 43fbebd9..827aa6a9 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -232,8 +232,8 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ m.rtp_ssrc.ssrc = _src->getSsrc(m.type); m.rtx_ssrc.ssrc = 2 + m.rtp_ssrc.ssrc; - m.rtp_ssrc.cname = "zlmediakit rtc"; - m.rtx_ssrc.cname = "zlmediakit rtc"; + m.rtp_ssrc.cname = "zlmediakit-rtc"; + m.rtx_ssrc.cname = "zlmediakit-rtc"; } } @@ -250,12 +250,12 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { switch (m.type) { case TrackVideo: { configure.video.enable = true; - configure.video.preferred_codec = {getCodecId(m.plan[0].codec)}; + configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec)); break; } case TrackAudio: { configure.audio.enable = true; - configure.audio.preferred_codec = {getCodecId(m.plan[0].codec)}; + configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec)); break; } default: From c42b678cab65a9e8d7cf6104c29abd379c57b2db Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 2 Apr 2021 20:35:43 +0800 Subject: [PATCH 048/218] =?UTF-8?q?=E5=AE=8C=E6=88=90rtc=E5=8F=91=E9=80=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 3 ++- webrtc/WebRtcTransport.cpp | 43 +++++++++++++++++++++++++++++++------- webrtc/WebRtcTransport.h | 6 ++++++ 3 files changed, 43 insertions(+), 9 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index bc573e59..5f487482 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1129,7 +1129,8 @@ void RtcMedia::checkValid() const{ auto rtx_plan = getPlan("rtx"); if (rtx_plan) { //开启rtx后必须指定rtx_ssrc - CHECK(!rtx_ssrc.empty() || !send_rtp); + //todo 此处不确定 +// CHECK(!rtx_ssrc.empty() || !send_rtp); } } diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 827aa6a9..e79e8516 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -67,6 +67,14 @@ void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){ onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush); } +const RtcSession& WebRtcTransport::getSdp(SdpType type) const{ + switch (type) { + case SdpType::offer: return *_offer_sdp; + case SdpType::answer: return *_answer_sdp; + default: throw std::invalid_argument("不识别的sdp类型"); + } +} + string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str); for (auto &finger_prints : transport->GetLocalFingerprints()) { @@ -108,7 +116,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){ fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport); RtcConfigure configure; - configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::recvonly, fingerprint); + configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint); onRtcConfigure(configure); //// 生成answer sdp //// @@ -201,6 +209,9 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { } void WebRtcTransportImp::onStartWebRTC() { + if (!canSendRtp()) { + return; + } _reader = _src->getRing()->attach(_socket->getPoller(), true); weak_ptr weak_self = shared_from_this(); _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt){ @@ -215,25 +226,43 @@ void WebRtcTransportImp::onStartWebRTC() { }); } +uint8_t WebRtcTransportImp::getSendPayloadType(TrackType type) { + for (auto &m : getSdp(SdpType::answer).media) { + if (m.type == type) { + return m.plan[0].pt; + } + } + return 0; +} + void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //需要修改pt - InfoL << flush; + if (rtp->type == TrackVideo) { + rtp->getHeader()->pt = getSendPayloadType(rtp->type); + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + } else { + + } } +bool WebRtcTransportImp::canSendRtp() const{ + auto &sdp = getSdp(SdpType::answer); + return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; +} + + void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ WebRtcTransport::onCheckSdp(type, sdp); - if (type != SdpType::answer) { + if (type != SdpType::answer || !canSendRtp()) { return; } + for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; } m.rtp_ssrc.ssrc = _src->getSsrc(m.type); - m.rtx_ssrc.ssrc = 2 + m.rtp_ssrc.ssrc; - m.rtp_ssrc.cname = "zlmediakit-rtc"; - m.rtx_ssrc.cname = "zlmediakit-rtc"; } } @@ -287,12 +316,10 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { RtpHeader *rtp = (RtpHeader *) buf; -// TraceL << (int)rtp->ssrc; } void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { RtcpHeader *rtcp = (RtcpHeader *) buf; -// TraceL << rtcpTypeToStr((RtcpType)rtcp->pt); } /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 271d1bbf..ed8d357a 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -79,6 +79,9 @@ protected: virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; +protected: + const RtcSession& getSdp(SdpType type) const; + private: void onSendSockData(const char *buf, size_t len, bool flush = true); void setRemoteDtlsFingerprint(const RtcSession &remote); @@ -124,11 +127,14 @@ private: void onDestory() override; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); SdpAttrCandidate::Ptr getIceCandidate() const; + uint8_t getSendPayloadType(TrackType type); + bool canSendRtp() const; private: Socket::Ptr _socket; RtspMediaSource::Ptr _src; RtspMediaSource::RingType::RingReader::Ptr _reader; + RtcSession _answer_sdp; }; From dc485c6211ef7fe20f69c855ffe66f98f354ac88 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 2 Apr 2021 21:08:40 +0800 Subject: [PATCH 049/218] =?UTF-8?q?=E6=94=AF=E6=8C=81=E6=A0=B9=E6=8D=AE?= =?UTF-8?q?=E6=83=85=E5=86=B5=E9=80=89=E6=8B=A9=E6=98=AF=E5=90=A6=E5=8F=91?= =?UTF-8?q?=E7=94=9Frtp?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/DtlsTransport.hpp | 2 +- webrtc/SrtpSession.hpp | 2 +- webrtc/WebRtcTransport.cpp | 35 +++++++++++++++-------------------- webrtc/WebRtcTransport.h | 3 ++- 4 files changed, 19 insertions(+), 23 deletions(-) diff --git a/webrtc/DtlsTransport.hpp b/webrtc/DtlsTransport.hpp index b3fa244e..41fa4c09 100644 --- a/webrtc/DtlsTransport.hpp +++ b/webrtc/DtlsTransport.hpp @@ -228,7 +228,7 @@ namespace RTC bool handshakeDoneNow{ false }; std::string remoteCert; //最大不超过mtu - static constexpr int SslReadBufferSize{ 1600 }; + static constexpr int SslReadBufferSize{ 2000 }; uint8_t sslReadBuffer[SslReadBufferSize]; }; } // namespace RTC diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index fb21ea9f..c601ca69 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -59,7 +59,7 @@ namespace RTC // Allocated by this. srtp_t session{ nullptr }; //rtp包最大1600 - static constexpr size_t EncryptBufferSize{ 1600 }; + static constexpr size_t EncryptBufferSize{ 2000 }; uint8_t EncryptBuffer[EncryptBufferSize]; DepLibSRTP::Ptr _env; }; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index e79e8516..e35381c6 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -226,23 +226,17 @@ void WebRtcTransportImp::onStartWebRTC() { }); } -uint8_t WebRtcTransportImp::getSendPayloadType(TrackType type) { - for (auto &m : getSdp(SdpType::answer).media) { - if (m.type == type) { - return m.plan[0].pt; - } - } - return 0; -} - void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - //需要修改pt - if (rtp->type == TrackVideo) { - rtp->getHeader()->pt = getSendPayloadType(rtp->type); - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); - } else { - + if (!_send_rtp_pt[rtp->type]) { + //忽略，对方不支持该编码类型 + return; } + auto tmp = rtp->getHeader()->pt; + //设置pt + rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + //还原pt + rtp->getHeader()->pt = tmp; } bool WebRtcTransportImp::canSendRtp() const{ @@ -250,7 +244,6 @@ bool WebRtcTransportImp::canSendRtp() const{ return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; } - void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer || !canSendRtp()) { @@ -263,19 +256,21 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ } m.rtp_ssrc.ssrc = _src->getSsrc(m.type); m.rtp_ssrc.cname = "zlmediakit-rtc"; + auto rtsp_media = _rtsp_send_sdp.getMedia(m.type); + if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { + _send_rtp_pt[m.type] = m.plan[0].pt; + } } } void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { WebRtcTransport::onRtcConfigure(configure); - - RtcSession sdp; - sdp.loadFrom(_src->getSdp(), false); + _rtsp_send_sdp.loadFrom(_src->getSdp(), false); configure.audio.enable = false; configure.video.enable = false; - for (auto &m : sdp.media) { + for (auto &m : _rtsp_send_sdp.media) { switch (m.type) { case TrackVideo: { configure.video.enable = true; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index ed8d357a..441be5db 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -127,7 +127,6 @@ private: void onDestory() override; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); SdpAttrCandidate::Ptr getIceCandidate() const; - uint8_t getSendPayloadType(TrackType type); bool canSendRtp() const; private: @@ -135,6 +134,8 @@ private: RtspMediaSource::Ptr _src; RtspMediaSource::RingType::RingReader::Ptr _reader; RtcSession _answer_sdp; + mutable RtcSession _rtsp_send_sdp; + mutable uint8_t _send_rtp_pt[2] = {0, 0}; }; From 8c460bfcffd62c4e5051f2faecfa1848ffa08361 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 2 Apr 2021 23:01:58 +0800 Subject: [PATCH 050/218] =?UTF-8?q?=E5=9F=BA=E6=9C=AC=E5=AE=8C=E6=88=90rtc?= =?UTF-8?q?=E8=BD=ACrtsp?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 78 +++++++++++++++++++++++++++++++-- webrtc/Sdp.h | 8 +++- webrtc/WebRtcTransport.cpp | 88 +++++++++++++++++++++++++++++++++++++- webrtc/WebRtcTransport.h | 18 +++++++- 4 files changed, 184 insertions(+), 8 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 5f487482..f83b2b7e 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -923,9 +923,74 @@ std::shared_ptr wrapSdpAttr(SdpItem::Ptr item){ return ret; } -string RtcSession::toString() const{ +static void toRtsp(vector &items) { + for (auto it = items.begin(); it != items.end();) { + switch ((*it)->getKey()[0]) { + case 'v': + case 'o': + case 's': + case 'i': + case 't': + case 'c': + case 'b':{ + ++it; + break; + } + + case 'm': { + auto m = dynamic_pointer_cast(*it); + CHECK(m); + m->proto = "RTP/AVP"; + ++it; + break; + } + case 'a': { + auto attr = dynamic_pointer_cast(*it); + CHECK(attr); + if (!strcasecmp(attr->detail->getKey(), "rtpmap") + || !strcasecmp(attr->detail->getKey(), "fmtp")) { + ++it; + break; + } + } + default: { + it = items.erase(it); + break; + } + } + } +} + +string RtcSession::toRtspSdp() const{ checkValid(); - RtcSessionSdp sdp; + RtcSession copy = *this; + copy.media.clear(); + for (auto &m : media) { + switch (m.type) { + case TrackAudio: + case TrackVideo: { + copy.media.emplace_back(m); + copy.media.back().plan.resize(1); + break; + } + default: + continue; + } + } + + auto sdp = copy.toRtcSessionSdp(); + toRtsp(sdp->items); + int i = 0; + for (auto &m : sdp->medias) { + toRtsp(m.items); + m.items.push_back(wrapSdpAttr(std::make_shared("control", string("trackID=") + to_string(i++)))); + } + return sdp->toString(); +} + +RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ + RtcSessionSdp::Ptr ret = std::make_shared(); + auto &sdp = *ret; sdp.items.emplace_back(std::make_shared >(to_string(version))); sdp.items.emplace_back(std::make_shared(origin)); sdp.items.emplace_back(std::make_shared >(session_name)); @@ -1076,7 +1141,12 @@ string RtcSession::toString() const{ sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(cand))); } } - return sdp.toString(); + return ret; +} + +string RtcSession::toString() const{ + checkValid(); + return toRtcSessionSdp()->toString(); } string RtcCodecPlan::getFmtp(const char *key) const{ @@ -1146,7 +1216,7 @@ void RtcSession::checkValid() const{ } } -RtcMedia *RtcSession::getMedia(TrackType type){ +const RtcMedia *RtcSession::getMedia(TrackType type) const{ for(auto &m : media){ if(m.type == type){ return &m; diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index c6720618..0d0f972a 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -550,6 +550,8 @@ public: class RtcSessionSdp : public RtcSdpBase{ public: + using Ptr = std::shared_ptr; + vector medias; void parse(const string &str); string toString() const override; @@ -652,7 +654,11 @@ public: void loadFrom(const string &sdp, bool check = true); void checkValid() const; string toString() const; - RtcMedia *getMedia(TrackType type); + string toRtspSdp() const; + const RtcMedia *getMedia(TrackType type) const; + +private: + RtcSessionSdp::Ptr toRtcSessionSdp() const; }; class RtcConfigure { diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index e35381c6..46ff8477 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -1,6 +1,11 @@ #include "WebRtcTransport.h" #include #include "Rtcp/Rtcp.h" +#include "Rtsp/RtpReceiver.h" +#define RTX_SSRC_OFFSET 2 +#define RTP_CNAME "zlmediakit-rtp" +#define RTX_CNAME "zlmediakit-rtx" + WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { _dtls_transport = std::make_shared(poller, this); @@ -51,7 +56,7 @@ void WebRtcTransport::OnDtlsTransportConnected( std::string &remoteCert) { InfoL; _srtp_session_send = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); - _srtp_session_recv = std::make_shared(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen); + _srtp_session_recv = std::make_shared(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen); onStartWebRTC(); } @@ -209,6 +214,29 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { } void WebRtcTransportImp::onStartWebRTC() { + if (canRecvRtp()) { + _push_src = std::make_shared(DEFAULT_VHOST, "live", "push"); + auto rtsp_sdp = getSdp(SdpType::answer).toRtspSdp(); + _push_src->setSdp(rtsp_sdp); + + for (auto &m : getSdp(SdpType::offer).media) { + for (auto &plan : m.plan) { + auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); + if (!hit_pan) { + continue; + } + auto &ref = _rtp_receiver[plan.pt]; + ref.plan = &plan; + ref.media = &m; + ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { + onSortedRtp(ref, std::move(rtp)); + }, [ref, this](const RtpPacket::Ptr &rtp) { + onBeforeSortedRtp(ref, rtp); + }); + } + } + } if (!canSendRtp()) { return; } @@ -244,6 +272,11 @@ bool WebRtcTransportImp::canSendRtp() const{ return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; } +bool WebRtcTransportImp::canRecvRtp() const{ + auto &sdp = getSdp(SdpType::answer); + return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly; +} + void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer || !canSendRtp()) { @@ -255,7 +288,12 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ continue; } m.rtp_ssrc.ssrc = _src->getSsrc(m.type); - m.rtp_ssrc.cname = "zlmediakit-rtc"; + m.rtp_ssrc.cname = RTP_CNAME; + //todo 先屏蔽rtx，因为chrome报错 + if (false && m.getRelatedRtxPlan(m.plan[0].pt)) { + m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc; + m.rtx_ssrc.cname = RTX_CNAME; + } auto rtsp_media = _rtsp_send_sdp.getMedia(m.type); if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { _send_rtp_pt[m.type] = m.plan[0].pt; @@ -309,14 +347,60 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ return candidate; } +class RtpReceiverImp : public RtpReceiver { +public: + RtpReceiverImp( function cb, function cb_before = nullptr){ + _on_sort = std::move(cb); + _on_before_sort = std::move(cb_before); + } + + ~RtpReceiverImp() override = default; + + bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){ + return handleOneRtp((int) type, type, samplerate, ptr, len); + } + +protected: + void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override { + _on_sort(std::move(rtp)); + } + + void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override { + if (_on_before_sort) { + _on_before_sort(rtp); + } + } + +private: + function _on_sort; + function _on_before_sort; +}; + void WebRtcTransportImp::onRtp(const char *buf, size_t len) { RtpHeader *rtp = (RtpHeader *) buf; + auto it = _rtp_receiver.find(rtp->pt); + if (it == _rtp_receiver.end()) { + WarnL; + return; + } + auto &info = it->second; + info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); } void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { RtcpHeader *rtcp = (RtcpHeader *) buf; } +void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { + if(!info.is_common_rtp){ + WarnL; + } + _push_src->onWrite(std::move(rtp), true); +} + +void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { + +} /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 441be5db..d754f2ca 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -9,7 +9,7 @@ #include "Sdp.h" #include "Poller/EventPoller.h" #include "Network/Socket.h" -#include "Rtsp/RtspMediaSource.h" +#include "Rtsp/RtspMediaSourceImp.h" using namespace toolkit; using namespace mediakit; @@ -95,6 +95,8 @@ private: RtcSession::Ptr _answer_sdp; }; +class RtpReceiverImp; + class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; @@ -128,6 +130,18 @@ private: void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; + bool canRecvRtp() const; + + class RtpPayloadInfo { + public: + bool is_common_rtp; + const RtcCodecPlan *plan; + const RtcMedia *media; + std::shared_ptr receiver; + }; + + void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp); + void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: Socket::Ptr _socket; @@ -136,6 +150,8 @@ private: RtcSession _answer_sdp; mutable RtcSession _rtsp_send_sdp; mutable uint8_t _send_rtp_pt[2] = {0, 0}; + RtspMediaSourceImp::Ptr _push_src; + unordered_map _rtp_receiver; }; From a176cb83a87607f0da1c5e1caecc301830119dc0 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 2 Apr 2021 23:12:37 +0800 Subject: [PATCH 051/218] =?UTF-8?q?=E9=A2=84=E7=95=99rtcp=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 20 ++++++++++++++++++-- webrtc/WebRtcTransport.h | 2 ++ 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 46ff8477..4eccfa2f 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -186,6 +186,17 @@ void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush) { } } +void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){ + const uint8_t *p = (uint8_t *) buf; + bool ret = false; + if (_srtp_session_send) { + ret = _srtp_session_send->EncryptRtcp(&p, &len); + } + if (ret) { + onSendSockData((char *) p, len, flush); + } +} + /////////////////////////////////////////////////////////////////////////////////// WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &poller){ WebRtcTransportImp::Ptr ret(new WebRtcTransportImp(poller), [](WebRtcTransportImp *ptr){ @@ -389,17 +400,22 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { RtcpHeader *rtcp = (RtcpHeader *) buf; + //todo rtcp相关 } void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if(!info.is_common_rtp){ WarnL; } - _push_src->onWrite(std::move(rtp), true); + if (_pli_ticker.elapsedTime() > 2000) { + //todo 发送pli + _pli_ticker.resetTime(); + } + _push_src->onWrite(std::move(rtp), false); } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { - + //todo rtcp相关 } /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index d754f2ca..11960872 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -45,6 +45,7 @@ public: * @param len rtcp长度 */ void sendRtpPacket(char *buf, size_t len, bool flush); + void sendRtcpPacket(char *buf, size_t len, bool flush); protected: //// dtls相关的回调 //// @@ -152,6 +153,7 @@ private: mutable uint8_t _send_rtp_pt[2] = {0, 0}; RtspMediaSourceImp::Ptr _push_src; unordered_map _rtp_receiver; + Ticker _pli_ticker; }; From e9c963dc828ab4e6b6b5d50ac944820dbe0ffe1d Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 3 Apr 2021 00:04:52 +0800 Subject: [PATCH 052/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E9=83=A8=E5=88=86rtc?= =?UTF-8?q?p=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 53 ++++++++++++++++++++++++++++++++++++-- webrtc/WebRtcTransport.h | 4 +++ 2 files changed, 55 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 4eccfa2f..5321d987 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -237,9 +237,12 @@ void WebRtcTransportImp::onStartWebRTC() { continue; } auto &ref = _rtp_receiver[plan.pt]; + _ssrc_info[m.rtp_ssrc.ssrc] = &ref; ref.plan = &plan; ref.media = &m; ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); + ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { onSortedRtp(ref, std::move(rtp)); }, [ref, this](const RtpPacket::Ptr &rtp) { @@ -274,6 +277,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //设置pt rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); //还原pt rtp->getHeader()->pt = tmp; } @@ -399,23 +403,68 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { } void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { - RtcpHeader *rtcp = (RtcpHeader *) buf; - //todo rtcp相关 + auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len); + for (auto rtcp : rtcps) { + switch ((RtcpType) rtcp->pt) { + case RtcpType::RTCP_SR : { + //对方汇报rtp发送情况 + RtcpSR *sr = (RtcpSR *) rtcp; + auto it = _ssrc_info.find(sr->items.ssrc); + if (it != _ssrc_info.end()) { + it->second->rtcp_context_recv->onRtcp(sr); + auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->ssrc, sr->items.ssrc); + sendRtcpPacket(rr->data(), rr->size(), true); + InfoL << "send rtcp rr"; + } + break; + } + case RtcpType::RTCP_RR : { + //对方汇报rtp接收情况 + RtcpRR *rr = (RtcpRR *) rtcp; + auto it = _ssrc_info.find(rr->items.ssrc); + if (it != _ssrc_info.end()) { + auto sr = it->second->rtcp_context_send->createRtcpSR(rr->ssrc); + sendRtcpPacket(sr->data(), sr->size(), true); + InfoL << "send rtcp sr"; + } + break; + } + default: break; + } + } +} + +int makeRtcpPli(char *packet, int len) { + if (packet == NULL || len != 12) + return -1; + memset(packet, 0, len); + RtcpHeader *rtcp = (RtcpHeader *) packet; + rtcp->version = 2; + rtcp->pt = (uint8_t) RtcpType::RTCP_PSFB; + rtcp->report_count = 1; + rtcp->length = htons((len / 4) - 1); + return 12; } void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if(!info.is_common_rtp){ WarnL; + return; } if (_pli_ticker.elapsedTime() > 2000) { //todo 发送pli _pli_ticker.resetTime(); + char rtcpbuf[12]; + makeRtcpPli(rtcpbuf, 12); + sendRtcpPacket(rtcpbuf, 12, true); + InfoL << "send pli"; } _push_src->onWrite(std::move(rtp), false); } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { //todo rtcp相关 + info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } /////////////////////////////////////////////////////////////////// diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 11960872..74a00b81 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -10,6 +10,7 @@ #include "Poller/EventPoller.h" #include "Network/Socket.h" #include "Rtsp/RtspMediaSourceImp.h" +#include "Rtcp/RtcpContext.h" using namespace toolkit; using namespace mediakit; @@ -139,6 +140,8 @@ private: const RtcCodecPlan *plan; const RtcMedia *media; std::shared_ptr receiver; + RtcpContext::Ptr rtcp_context_recv; + RtcpContext::Ptr rtcp_context_send; }; void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp); @@ -153,6 +156,7 @@ private: mutable uint8_t _send_rtp_pt[2] = {0, 0}; RtspMediaSourceImp::Ptr _push_src; unordered_map _rtp_receiver; + unordered_map _ssrc_info; Ticker _pli_ticker; }; From cfda7d8ba636e777122c18e22e031f5dd6121b9d Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 3 Apr 2021 08:32:20 +0800 Subject: [PATCH 053/218] =?UTF-8?q?=E6=95=B4=E7=90=86=E4=BB=A3=E7=A0=81?= =?UTF-8?q?=E5=B9=B6=E5=AE=8C=E5=96=84rtcp=E7=9B=B8=E5=85=B3=E5=8A=9F?= =?UTF-8?q?=E8=83=BD?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 96 +++++++++++++++++++------------------- 1 file changed, 47 insertions(+), 49 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 5321d987..eaf7b97c 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -224,6 +224,12 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { _src = src; } +void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) { + auto ptr = BufferRaw::create(); + ptr->assign(buf, len); + _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush); +} + void WebRtcTransportImp::onStartWebRTC() { if (canRecvRtp()) { _push_src = std::make_shared(DEFAULT_VHOST, "live", "push"); @@ -251,35 +257,20 @@ void WebRtcTransportImp::onStartWebRTC() { } } } - if (!canSendRtp()) { - return; - } - _reader = _src->getRing()->attach(_socket->getPoller(), true); - weak_ptr weak_self = shared_from_this(); - _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt){ - auto strongSelf = weak_self.lock(); - if (!strongSelf) { - return; - } - size_t i = 0; - pkt->for_each([&](const RtpPacket::Ptr &rtp) { - strongSelf->onSendRtp(rtp, ++i == pkt->size()); + if (canSendRtp()) { + _reader = _src->getRing()->attach(_socket->getPoller(), true); + weak_ptr weak_self = shared_from_this(); + _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) { + auto strongSelf = weak_self.lock(); + if (!strongSelf) { + return; + } + size_t i = 0; + pkt->for_each([&](const RtpPacket::Ptr &rtp) { + strongSelf->onSendRtp(rtp, ++i == pkt->size()); + }); }); - }); -} - -void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - if (!_send_rtp_pt[rtp->type]) { - //忽略，对方不支持该编码类型 - return; } - auto tmp = rtp->getHeader()->pt; - //设置pt - rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); - _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); - //还原pt - rtp->getHeader()->pt = tmp; } bool WebRtcTransportImp::canSendRtp() const{ @@ -343,13 +334,6 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { configure.addCandidate(*getIceCandidate()); } - -void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) { - auto ptr = BufferRaw::create(); - ptr->assign(buf, len); - _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush); -} - SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ auto candidate = std::make_shared(); candidate->foundation = "udpcandidate"; @@ -391,17 +375,6 @@ private: function _on_before_sort; }; -void WebRtcTransportImp::onRtp(const char *buf, size_t len) { - RtpHeader *rtp = (RtpHeader *) buf; - auto it = _rtp_receiver.find(rtp->pt); - if (it == _rtp_receiver.end()) { - WarnL; - return; - } - auto &info = it->second; - info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); -} - void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len); for (auto rtcp : rtcps) { @@ -409,10 +382,10 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { case RtcpType::RTCP_SR : { //对方汇报rtp发送情况 RtcpSR *sr = (RtcpSR *) rtcp; - auto it = _ssrc_info.find(sr->items.ssrc); + auto it = _ssrc_info.find(sr->ssrc); if (it != _ssrc_info.end()) { it->second->rtcp_context_recv->onRtcp(sr); - auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->ssrc, sr->items.ssrc); + auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); InfoL << "send rtcp rr"; } @@ -421,9 +394,9 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { case RtcpType::RTCP_RR : { //对方汇报rtp接收情况 RtcpRR *rr = (RtcpRR *) rtcp; - auto it = _ssrc_info.find(rr->items.ssrc); + auto it = _ssrc_info.find(rr->ssrc); if (it != _ssrc_info.end()) { - auto sr = it->second->rtcp_context_send->createRtcpSR(rr->ssrc); + auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); InfoL << "send rtcp sr"; } @@ -446,6 +419,17 @@ int makeRtcpPli(char *packet, int len) { return 12; } +void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + RtpHeader *rtp = (RtpHeader *) buf; + auto it = _rtp_receiver.find(rtp->pt); + if (it == _rtp_receiver.end()) { + WarnL; + return; + } + auto &info = it->second; + info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); +} + void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if(!info.is_common_rtp){ WarnL; @@ -466,6 +450,20 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp //todo rtcp相关 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } + +void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ + if (!_send_rtp_pt[rtp->type]) { + //忽略，对方不支持该编码类型 + return; + } + auto tmp = rtp->getHeader()->pt; + //设置pt + rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + //还原pt + rtp->getHeader()->pt = tmp; +} /////////////////////////////////////////////////////////////////// From 130a06897faf6856a945936847b7ee8a12ff17f9 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 3 Apr 2021 08:45:18 +0800 Subject: [PATCH 054/218] =?UTF-8?q?=E6=95=B4=E7=90=86=E5=B9=B6=E6=B3=A8?= =?UTF-8?q?=E9=87=8A=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 78 +++++++++++++++++++++++--------------- 1 file changed, 47 insertions(+), 31 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index eaf7b97c..69a4003a 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -164,12 +164,16 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * if (is_rtp(buf)) { if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) { onRtp(buf, len); + } else { + WarnL; } return; } if (is_rtcp(buf)) { if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) { onRtcp(buf, len); + } else { + WarnL; } return; } @@ -230,6 +234,18 @@ void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sock _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush); } +/////////////////////////////////////////////////////////////////// + +bool WebRtcTransportImp::canSendRtp() const{ + auto &sdp = getSdp(SdpType::answer); + return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; +} + +bool WebRtcTransportImp::canRecvRtp() const{ + auto &sdp = getSdp(SdpType::answer); + return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly; +} + void WebRtcTransportImp::onStartWebRTC() { if (canRecvRtp()) { _push_src = std::make_shared(DEFAULT_VHOST, "live", "push"); @@ -242,6 +258,7 @@ void WebRtcTransportImp::onStartWebRTC() { if (!hit_pan) { continue; } + //获取offer端rtp的ssrc和pt相关信息 auto &ref = _rtp_receiver[plan.pt]; _ssrc_info[m.rtp_ssrc.ssrc] = &ref; ref.plan = &plan; @@ -273,22 +290,13 @@ void WebRtcTransportImp::onStartWebRTC() { } } -bool WebRtcTransportImp::canSendRtp() const{ - auto &sdp = getSdp(SdpType::answer); - return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; -} - -bool WebRtcTransportImp::canRecvRtp() const{ - auto &sdp = getSdp(SdpType::answer); - return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly; -} - void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer || !canSendRtp()) { return; } + //添加answer sdp的ssrc信息，并且记录发送rtp的pt for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; @@ -311,18 +319,14 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { WebRtcTransport::onRtcConfigure(configure); _rtsp_send_sdp.loadFrom(_src->getSdp(), false); - configure.audio.enable = false; - configure.video.enable = false; - + //根据rtsp流的相关信息，设置rtc最佳编码 for (auto &m : _rtsp_send_sdp.media) { switch (m.type) { case TrackVideo: { - configure.video.enable = true; configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec)); break; } case TrackAudio: { - configure.audio.enable = true; configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec)); break; } @@ -331,21 +335,27 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { } } + //添加接收端口candidate信息 configure.addCandidate(*getIceCandidate()); } SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ auto candidate = std::make_shared(); candidate->foundation = "udpcandidate"; + //rtp端口 candidate->component = 1; candidate->transport = "udp"; + //优先级，单candidate时随便 candidate->priority = 100; + //todo 此处修改为配置文件 candidate->address = SockUtil::get_local_ip(); candidate->port = _socket->get_local_port(); candidate->type = "host"; return candidate; } +/////////////////////////////////////////////////////////////////// + class RtpReceiverImp : public RtpReceiver { public: RtpReceiverImp( function cb, function cb_before = nullptr){ @@ -402,11 +412,28 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } break; } + case RtcpType::RTCP_BYE : { + //todo 此处应该销毁对象 + break; + } default: break; } } } +void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + RtpHeader *rtp = (RtpHeader *) buf; + //根据接收到的rtp的pt信息，找到该流的信息 + auto it = _rtp_receiver.find(rtp->pt); + if (it == _rtp_receiver.end()) { + WarnL; + return; + } + auto &info = it->second; + //解析并排序rtp + info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); +} + int makeRtcpPli(char *packet, int len) { if (packet == NULL || len != 12) return -1; @@ -419,24 +446,16 @@ int makeRtcpPli(char *packet, int len) { return 12; } -void WebRtcTransportImp::onRtp(const char *buf, size_t len) { - RtpHeader *rtp = (RtpHeader *) buf; - auto it = _rtp_receiver.find(rtp->pt); - if (it == _rtp_receiver.end()) { - WarnL; - return; - } - auto &info = it->second; - info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); -} +/////////////////////////////////////////////////////////////////// void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if(!info.is_common_rtp){ + //todo rtx/red/ulpfec类型的rtp先未处理 WarnL; return; } if (_pli_ticker.elapsedTime() > 2000) { - //todo 发送pli + //todo 定期发送pli _pli_ticker.resetTime(); char rtcpbuf[12]; makeRtcpPli(rtcpbuf, 12); @@ -447,7 +466,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { - //todo rtcp相关 + //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } @@ -460,11 +479,8 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //设置pt rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + //统计rtp发送情况，好做sr汇报 _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); //还原pt rtp->getHeader()->pt = tmp; } -/////////////////////////////////////////////////////////////////// - - - From 2abb5078f93fbb88a8c360b2eaed757ad4499e55 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 3 Apr 2021 09:34:49 +0800 Subject: [PATCH 055/218] =?UTF-8?q?=E5=AE=9E=E7=8E=B0rtc=E8=BD=ACrtsp?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 40 +++++++++++++++++++++++++++++++++++++- src/Rtcp/Rtcp.h | 29 +++++++++++++++++++++++++++ webrtc/WebRtcTransport.cpp | 26 +++++++++++-------------- webrtc/WebRtcTransport.h | 1 + 4 files changed, 80 insertions(+), 16 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 2e0b21db..220a8439 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -82,6 +82,11 @@ string RtcpHeader::dumpString() const { RtcpSdes *rtcp = (RtcpSdes *)this; return rtcp->dumpString(); } + + case RtcpType::RTCP_PSFB: { + RtcpPli *rtcp = (RtcpPli *)this; + return rtcp->dumpString(); + } default: return StrPrinter << dumpHeader() << hexdump((char *)this + sizeof(*this), length << 2); } } @@ -105,6 +110,12 @@ void RtcpHeader::net2Host(size_t len){ sdes->net2Host(len); break; } + + case RtcpType::RTCP_PSFB: { + RtcpPli *pli = (RtcpPli *)this; + pli->net2Host(len); + break; + } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); } } @@ -121,7 +132,7 @@ vector RtcpHeader::loadFromBytes(char *data, size_t len){ ret.emplace_back(rtcp); } catch (std::exception &ex) { //不能处理的rtcp包，或者无法解析的rtcp包，忽略掉 - WarnL << ex.what(); + WarnL << ex.what() << ",长度为:" << rtcp_len; } ptr += rtcp_len; remain -= rtcp_len; @@ -402,4 +413,31 @@ vector RtcpSdes::getItemList() { return ret; } +//////////////////////////////////////////////////////////////////// + +std::shared_ptr RtcpPli::create() { + auto bytes = alignSize(sizeof(RtcpPli)); + auto ptr = (RtcpRR *) new char[bytes]; + setupHeader(ptr, RtcpType::RTCP_PSFB, 1, bytes); + return std::shared_ptr((RtcpPli *) ptr, [](RtcpPli *ptr) { + delete[] (char *) ptr; + }); +} + +string RtcpPli::dumpString() const { + _StrPrinter printer; + printer << RtcpHeader::dumpHeader(); + printer << "ssrc:" << ssrc << "\r\n"; + printer << "ssrc_media:" << ssrc_media; + return std::move(printer); +} + +void RtcpPli::net2Host(size_t size) { + static const size_t kMinSize = sizeof(RtcpPli); + CHECK_MIN_SIZE(size, kMinSize); + RtcpHeader::net2Host(); + ssrc = ntohl(ssrc); + ssrc_media = ntohl(ssrc_media); +} + }//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index dbf4c769..8fcc1a1a 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -456,6 +456,35 @@ private: void net2Host(size_t size); } PACKED; +//PLI +class RtcpPli : public RtcpHeader { +public: + friend class RtcpHeader; + uint32_t ssrc; + uint32_t ssrc_media; + +public: + /** + * 创建SDES包，只赋值了RtcpHeader以及SdesItem对象的length和text部分 + * @param item_text SdesItem列表，只赋值length和text部分 + * @return SDES包 + */ + static std::shared_ptr create(); + +private: + /** + * 打印字段详情 + * 使用net2Host转换成主机字节序后才可使用此函数 + */ + string dumpString() const; + + /** + * 网络字节序转换为主机字节序 + * @param size 字节长度，防止内存越界 + */ + void net2Host(size_t size); +} PACKED; + #if defined(_WIN32) #pragma pack(pop) #endif // defined(_WIN32) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 69a4003a..23dc13e0 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -253,6 +253,9 @@ void WebRtcTransportImp::onStartWebRTC() { _push_src->setSdp(rtsp_sdp); for (auto &m : getSdp(SdpType::offer).media) { + if (m.type == TrackVideo) { + _recv_video_ssrc = m.rtp_ssrc.ssrc; + } for (auto &plan : m.plan) { auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); if (!hit_pan) { @@ -416,6 +419,10 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { //todo 此处应该销毁对象 break; } + case RtcpType::RTCP_PSFB: { +// InfoL << rtcp->dumpString(); + break; + } default: break; } } @@ -434,18 +441,6 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); } -int makeRtcpPli(char *packet, int len) { - if (packet == NULL || len != 12) - return -1; - memset(packet, 0, len); - RtcpHeader *rtcp = (RtcpHeader *) packet; - rtcp->version = 2; - rtcp->pt = (uint8_t) RtcpType::RTCP_PSFB; - rtcp->report_count = 1; - rtcp->length = htons((len / 4) - 1); - return 12; -} - /////////////////////////////////////////////////////////////////// void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { @@ -457,9 +452,10 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr if (_pli_ticker.elapsedTime() > 2000) { //todo 定期发送pli _pli_ticker.resetTime(); - char rtcpbuf[12]; - makeRtcpPli(rtcpbuf, 12); - sendRtcpPacket(rtcpbuf, 12, true); + auto pli = RtcpPli::create(); + pli->ssrc = htonl(0); + pli->ssrc_media = htonl(_recv_video_ssrc); + sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true); InfoL << "send pli"; } _push_src->onWrite(std::move(rtp), false); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 74a00b81..f8d4f6b7 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -157,6 +157,7 @@ private: RtspMediaSourceImp::Ptr _push_src; unordered_map _rtp_receiver; unordered_map _ssrc_info; + uint32_t _recv_video_ssrc; Ticker _pli_ticker; }; From c70721a520d8948d420e1dd9ab40e9cbc2d6b3c8 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 21:42:11 +0800 Subject: [PATCH 056/218] =?UTF-8?q?=E4=BF=AE=E6=94=B9rtp=20pt=E6=97=B6?= =?UTF-8?q?=EF=BC=8C=E7=A1=AE=E4=BF=9D=E4=B8=8D=E5=BD=B1=E5=93=8D=E5=85=B6?= =?UTF-8?q?=E4=BB=96=E6=92=AD=E6=94=BE=E5=99=A8?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/SrtpSession.cpp | 3 ++- webrtc/SrtpSession.hpp | 2 +- webrtc/WebRtcTransport.cpp | 16 ++++++---------- webrtc/WebRtcTransport.h | 2 +- 4 files changed, 10 insertions(+), 13 deletions(-) diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index a970fd7c..64c1bca0 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -213,7 +213,7 @@ namespace RTC } } - bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len) + bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt) { MS_TRACE(); @@ -226,6 +226,7 @@ namespace RTC } std::memcpy(EncryptBuffer, *data, *len); + EncryptBuffer[1] |= (pt & 0x7F); srtp_err_status_t err = srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index c601ca69..5b3edefd 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -46,7 +46,7 @@ namespace RTC ~SrtpSession(); public: - bool EncryptRtp(const uint8_t** data, size_t* len); + bool EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt); bool DecryptSrtp(uint8_t* data, size_t* len); bool EncryptRtcp(const uint8_t** data, size_t* len); bool DecryptSrtcp(uint8_t* data, size_t* len); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 23dc13e0..9920157e 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -179,11 +179,11 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * } } -void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush) { +void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) { const uint8_t *p = (uint8_t *) buf; bool ret = false; if (_srtp_session_send) { - ret = _srtp_session_send->EncryptRtp(&p, &len); + ret = _srtp_session_send->EncryptRtp(&p, &len, pt); } if (ret) { onSendSockData((char *) p, len, flush); @@ -467,16 +467,12 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp } void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - if (!_send_rtp_pt[rtp->type]) { + auto &pt = _send_rtp_pt[rtp->type]; + if (!pt) { //忽略，对方不支持该编码类型 return; } - auto tmp = rtp->getHeader()->pt; - //设置pt - rtp->getHeader()->pt = _send_rtp_pt[rtp->type]; - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush); + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); //统计rtp发送情况，好做sr汇报 - _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); - //还原pt - rtp->getHeader()->pt = tmp; + _rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index f8d4f6b7..3495fe84 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -45,7 +45,7 @@ public: * @param buf rtcp内容 * @param len rtcp长度 */ - void sendRtpPacket(char *buf, size_t len, bool flush); + void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt); void sendRtcpPacket(char *buf, size_t len, bool flush); protected: From f33d3ba1750c5c35a4d96c3ccbfce7d0eda9ca81 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:12:35 +0800 Subject: [PATCH 057/218] =?UTF-8?q?=E8=BF=98=E5=8E=9F=E9=85=8D=E7=BD=AE?= =?UTF-8?q?=E6=96=87=E4=BB=B6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/conf/config.ini b/conf/config.ini index 46d57ce3..59a7f06b 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -136,18 +136,18 @@ charSet=utf-8 #http链接超时时间 keepAliveSecond=30 #http请求体最大字节数，如果post的body太大，则不适合缓存body在内存 -maxReqSize=40960 +maxReqSize=4096 #404网页内容，用户可以自定义404网页 notFound=404 Not Found

您访问的资源不存在！

ZLMediaKit-4.0
#http服务器监听端口 -port=0 +port=80 #http文件服务器根目录 #可以为相对(相对于本可执行程序目录)或绝对路径 rootPath=./www #http文件服务器读文件缓存大小，单位BYTE，调整该参数可以优化文件io性能 sendBufSize=65536 #https服务器监听端口 -sslport=20443 +sslport=443 #是否显示文件夹菜单，开启后可以浏览文件夹 dirMenu=1 @@ -187,9 +187,9 @@ keepAliveSecond=15 #在接收rtmp推流时，是否重新生成时间戳(很多推流器的时间戳着实很烂) modifyStamp=0 #rtmp服务器监听端口 -port=8935 +port=1935 #rtmps服务器监听地址 -sslport=0 +sslport=19350 [rtp] #音频mtu大小，该参数限制rtp最大字节数，推荐不要超过1400 @@ -202,7 +202,7 @@ videoMtuSize=1400 #导出调试数据(包括rtp/ps/h264)至该目录,置空则关闭数据导出 dumpDir= #udp和tcp代理服务器，支持rtp(必须是ts或ps类型)代理 -port=0 +port=10000 #rtp超时时间，单位秒 timeoutSec=15 @@ -221,13 +221,13 @@ handshakeSecond=15 #或者tcp发送缓存超过这个时间，则会断开连接，单位秒 keepAliveSecond=15 #rtsp服务器监听地址 -port=8554 +port=554 #rtsps服务器监听地址 -sslport=0 +sslport=322 [shell] #调试telnet服务器接受最大bufffer大小 maxReqSize=1024 #调试telnet服务器监听端口 -port=0 +port=9000 From 5f0aeaa32a6fa98198c0aff1cb6213db20788af9 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:13:01 +0800 Subject: [PATCH 058/218] =?UTF-8?q?=E5=8C=BA=E5=88=86=E6=8E=A8=E6=B5=81?= =?UTF-8?q?=E5=92=8C=E6=92=AD=E6=94=BE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- www/webrtc/{index.html => index_play.html} | 4 +- www/webrtc/index_push.html | 97 ++++++++++++++++++++++ 2 files changed, 99 insertions(+), 2 deletions(-) rename www/webrtc/{index.html => index_play.html} (92%) create mode 100644 www/webrtc/index_push.html diff --git a/www/webrtc/index.html b/www/webrtc/index_play.html similarity index 92% rename from www/webrtc/index.html rename to www/webrtc/index_play.html index 81fd23be..17b85b22 100644 --- a/www/webrtc/index.html +++ b/www/webrtc/index_play.html @@ -22,12 +22,12 @@

- +

- +

diff --git a/www/webrtc/index_push.html b/www/webrtc/index_push.html new file mode 100644 index 00000000..8821bf2a --- /dev/null +++ b/www/webrtc/index_push.html @@ -0,0 +1,97 @@ + + + + ZLM RTC demo + + + + + +
+
+ + + +
+ +
+ +

+ + +

+ +

+ + +

+ + + +
+
+ + + + + + + + \ No newline at end of file From 09add8d351f6c5698d208f7c7e63953762c51b78 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:14:42 +0800 Subject: [PATCH 059/218] =?UTF-8?q?=E4=BF=AE=E6=94=B9=E4=B8=BA=E6=8D=95?= =?UTF-8?q?=E6=8D=89=E6=91=84=E5=83=8F=E5=A4=B4?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- www/webrtc/ZLMRTCClient.js | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js index b1e049ab..969178b6 100644 --- a/www/webrtc/ZLMRTCClient.js +++ b/www/webrtc/ZLMRTCClient.js @@ -7307,8 +7307,9 @@ var ZLMRTCClient = (function (exports) { } start() { - let audioConstraints = new AudioTrackConstraints(AudioSourceInfo.SCREENCAST); - let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.SCREENCAST); + //todo 此处修改捕获桌面或摄像头 + let audioConstraints = new AudioTrackConstraints(AudioSourceInfo.MIC); + let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.CAMERA); MediaStreamFactory.createMediaStream(new StreamConstraints(audioConstraints, videoConstraints)).then(stream => { this._localStream = stream; this.dispatch(Events$1.WEBRTC_ON_LOCAL_STREAM, stream); From 49d8e2f825a85f24c8621b58cfcbddff2c031e67 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:17:11 +0800 Subject: [PATCH 060/218] =?UTF-8?q?http=E8=AF=B7=E6=B1=82=E6=9C=80?= =?UTF-8?q?=E5=A4=A7=E5=85=81=E8=AE=B840KB=E6=95=B0=E6=8D=AE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/conf/config.ini b/conf/config.ini index 59a7f06b..904cc57a 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -136,7 +136,7 @@ charSet=utf-8 #http链接超时时间 keepAliveSecond=30 #http请求体最大字节数，如果post的body太大，则不适合缓存body在内存 -maxReqSize=4096 +maxReqSize=40960 #404网页内容，用户可以自定义404网页 notFound=404 Not Found

您访问的资源不存在！

ZLMediaKit-4.0
#http服务器监听端口 From fe02f2cf1cf56dcce5f29514ea976492f6cd5883 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:20:10 +0800 Subject: [PATCH 061/218] =?UTF-8?q?rtc=E6=8E=A8=E6=B5=81=E5=92=8C=E6=92=AD?= =?UTF-8?q?=E6=94=BE=E6=B7=BB=E5=8A=A0=E4=BA=8B=E4=BB=B6=E8=A7=A6=E5=8F=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 90 ++++++++++++++++++++++++---- webrtc/WebRtcTransport.cpp | 118 ++++++++++++++++++++----------------- webrtc/WebRtcTransport.h | 22 ++++--- 3 files changed, 155 insertions(+), 75 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 22c2feea..1ae2b0cd 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1082,19 +1082,87 @@ void installWebApi() { #ifdef ENABLE_WEBRTC static list rtcs; - api_regist("/index/api/webrtc",[](API_ARGS_STRING){ + api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){ CHECK_ARGS("app", "stream"); - auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, DEFAULT_VHOST, allArgs.getUrlArgs()["app"], allArgs.getUrlArgs()["stream"])); - if (!src) { - throw ApiRetException("流不存在", API::NotFound); - } - headerOut["Content-Type"] = "text/plain"; + + auto offer_sdp = allArgs.Content(); + auto type = allArgs.getUrlArgs()["type"]; + MediaInfo info(StrPrinter << "rtc://" << headerIn["Host"] << "/" << allArgs.getUrlArgs()["app"] << "/" << allArgs.getUrlArgs()["stream"] << "?" << allArgs.Params()); + + //设置返回类型 + headerOut["Content-Type"] = HttpFileManager::getContentType(".json"); + //设置跨域 headerOut["Access-Control-Allow-Origin"] = "*"; - auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); - rtc->attach(src); - val["sdp"] = rtc->getAnswerSdp(allArgs.Content()); - val["type"] = "answer"; - rtcs.emplace_back(rtc); + + if (type.empty() || !strcasecmp(type.data(), "play")) { + Broadcast::AuthInvoker authInvoker = [invoker, offer_sdp, val, info, headerOut](const string &err) mutable { + try { + auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid)); + if (!src) { + throw runtime_error("流不存在"); + } + if (!err.empty()) { + throw runtime_error(StrPrinter << "播放鉴权失败:" << err); + } + auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); + rtc->attach(src); + val["sdp"] = rtc->getAnswerSdp(offer_sdp); + val["type"] = "answer"; + rtcs.emplace_back(rtc); + invoker(200, headerOut, val.toStyledString()); + } catch (std::exception &ex) { + val["code"] = API::Exception; + val["msg"] = ex.what(); + invoker(200, headerOut, val.toStyledString()); + } + }; + + //广播通用播放url鉴权事件 + auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, authInvoker, sender); + if (!flag) { + //该事件无人监听,默认不鉴权 + authInvoker(""); + } + return; + } + + if (!strcasecmp(type.data(), "push")) { + Broadcast::PublishAuthInvoker authInvoker = [invoker, offer_sdp, val, info, headerOut](const string &err, bool enableHls, bool enableMP4) mutable { + try { + auto src = dynamic_pointer_cast(MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid)); + if (src) { + throw std::runtime_error("已经在推流"); + } + if (!err.empty()) { + throw runtime_error(StrPrinter << "推流鉴权失败:" << err); + } + auto push_src = std::make_shared(info._vhost, info._app, info._streamid); + push_src->setProtocolTranslation(enableHls, enableMP4); + auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); + rtc->attach(push_src); + val["sdp"] = rtc->getAnswerSdp(offer_sdp); + val["type"] = "answer"; + rtcs.emplace_back(rtc); + invoker(200, headerOut, val.toStyledString()); + } catch (std::exception &ex) { + val["code"] = API::Exception; + val["msg"] = ex.what(); + invoker(200, headerOut, val.toStyledString()); + } + }; + + //rtsp推流需要鉴权 + auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, info, authInvoker, sender); + if (!flag) { + //该事件无人监听,默认不鉴权 + GET_CONFIG(bool, toHls, General::kPublishToHls); + GET_CONFIG(bool, toMP4, General::kPublishToMP4); + authInvoker("", toHls, toMP4); + } + return; + } + + throw ApiRetException("不支持该类型", API::InvalidArgs); }); #endif diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9920157e..0e32859b 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -6,10 +6,10 @@ #define RTP_CNAME "zlmediakit-rtp" #define RTX_CNAME "zlmediakit-rtx" - WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { + _poller = poller; _dtls_transport = std::make_shared(poller, this); - _ice_server = std::make_shared(this, makeRandStr(4), makeRandStr(24)); + _ice_server = std::make_shared(this, makeRandStr(4), makeRandStr(28).substr(4)); } void WebRtcTransport::onDestory(){ @@ -17,6 +17,10 @@ void WebRtcTransport::onDestory(){ _ice_server = nullptr; } +const EventPoller::Ptr& WebRtcTransport::getPoller() const{ + return _poller; +} + ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) { @@ -127,10 +131,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){ //// 生成answer sdp //// _answer_sdp = configure.createAnswer(*_offer_sdp); onCheckSdp(SdpType::answer, *_answer_sdp); - - auto str = _answer_sdp->toString(); - TraceL << "\r\n" << str; - return str; + return _answer_sdp->toString(); } bool is_dtls(char *buf) { @@ -247,35 +248,33 @@ bool WebRtcTransportImp::canRecvRtp() const{ } void WebRtcTransportImp::onStartWebRTC() { - if (canRecvRtp()) { - _push_src = std::make_shared(DEFAULT_VHOST, "live", "push"); - auto rtsp_sdp = getSdp(SdpType::answer).toRtspSdp(); - _push_src->setSdp(rtsp_sdp); - - for (auto &m : getSdp(SdpType::offer).media) { - if (m.type == TrackVideo) { - _recv_video_ssrc = m.rtp_ssrc.ssrc; - } - for (auto &plan : m.plan) { - auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); - if (!hit_pan) { - continue; - } - //获取offer端rtp的ssrc和pt相关信息 - auto &ref = _rtp_receiver[plan.pt]; - _ssrc_info[m.rtp_ssrc.ssrc] = &ref; - ref.plan = &plan; - ref.media = &m; - ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; - ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); - ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); - ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { - onSortedRtp(ref, std::move(rtp)); - }, [ref, this](const RtpPacket::Ptr &rtp) { - onBeforeSortedRtp(ref, rtp); - }); - } + for (auto &m : getSdp(SdpType::offer).media) { + if (m.type == TrackVideo) { + _recv_video_ssrc = m.rtp_ssrc.ssrc; } + for (auto &plan : m.plan) { + auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); + if (!hit_pan) { + continue; + } + //获取offer端rtp的ssrc和pt相关信息 + auto &ref = _rtp_info_pt[plan.pt]; + _rtp_info_ssrc[m.rtp_ssrc.ssrc] = &ref; + ref.plan = &plan; + ref.media = &m; + ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); + ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); + ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { + onSortedRtp(ref, std::move(rtp)); + }, [ref, this](const RtpPacket::Ptr &rtp) { + onBeforeSortedRtp(ref, rtp); + }); + } + } + + if (canRecvRtp()) { + _src->setSdp(getSdp(SdpType::answer).toRtspSdp()); } if (canSendRtp()) { _reader = _src->getRing()->attach(_socket->getPoller(), true); @@ -320,22 +319,31 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { WebRtcTransport::onRtcConfigure(configure); - _rtsp_send_sdp.loadFrom(_src->getSdp(), false); - //根据rtsp流的相关信息，设置rtc最佳编码 - for (auto &m : _rtsp_send_sdp.media) { - switch (m.type) { - case TrackVideo: { - configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec)); - break; + if (!_src->getSdp().empty()) { + //这是播放 + configure.video.direction = RtpDirection::sendonly; + configure.audio.direction = RtpDirection::sendonly; + _rtsp_send_sdp.loadFrom(_src->getSdp(), false); + //根据rtsp流的相关信息，设置rtc最佳编码 + for (auto &m : _rtsp_send_sdp.media) { + switch (m.type) { + case TrackVideo: { + configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec)); + break; + } + case TrackAudio: { + configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec)); + break; + } + default: + break; } - case TrackAudio: { - configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec)); - break; - } - default: - break; } + } else { + //这是推流 + configure.video.direction = RtpDirection::recvonly; + configure.audio.direction = RtpDirection::recvonly; } //添加接收端口candidate信息 @@ -395,8 +403,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { case RtcpType::RTCP_SR : { //对方汇报rtp发送情况 RtcpSR *sr = (RtcpSR *) rtcp; - auto it = _ssrc_info.find(sr->ssrc); - if (it != _ssrc_info.end()) { + auto it = _rtp_info_ssrc.find(sr->ssrc); + if (it != _rtp_info_ssrc.end()) { it->second->rtcp_context_recv->onRtcp(sr); auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); @@ -407,8 +415,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { case RtcpType::RTCP_RR : { //对方汇报rtp接收情况 RtcpRR *rr = (RtcpRR *) rtcp; - auto it = _ssrc_info.find(rr->ssrc); - if (it != _ssrc_info.end()) { + auto it = _rtp_info_ssrc.find(rr->ssrc); + if (it != _rtp_info_ssrc.end()) { auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); InfoL << "send rtcp sr"; @@ -431,8 +439,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { void WebRtcTransportImp::onRtp(const char *buf, size_t len) { RtpHeader *rtp = (RtpHeader *) buf; //根据接收到的rtp的pt信息，找到该流的信息 - auto it = _rtp_receiver.find(rtp->pt); - if (it == _rtp_receiver.end()) { + auto it = _rtp_info_pt.find(rtp->pt); + if (it == _rtp_info_pt.end()) { WarnL; return; } @@ -458,7 +466,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true); InfoL << "send pli"; } - _push_src->onWrite(std::move(rtp), false); + _src->onWrite(std::move(rtp), false); } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { @@ -474,5 +482,5 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ } sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); //统计rtp发送情况，好做sr汇报 - _rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 3495fe84..000c4cc7 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -44,10 +44,14 @@ public: * 发送rtp * @param buf rtcp内容 * @param len rtcp长度 + * @param flush 是否flush socket + * @param pt rtp payload type */ void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt); void sendRtcpPacket(char *buf, size_t len, bool flush); + const EventPoller::Ptr& getPoller() const; + protected: //// dtls相关的回调 //// void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override {}; @@ -89,6 +93,7 @@ private: void setRemoteDtlsFingerprint(const RtcSession &remote); private: + EventPoller::Ptr _poller; std::shared_ptr _ice_server; std::shared_ptr _dtls_transport; std::shared_ptr _srtp_session_send; @@ -148,17 +153,16 @@ private: void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: - Socket::Ptr _socket; - RtspMediaSource::Ptr _src; - RtspMediaSource::RingType::RingReader::Ptr _reader; - RtcSession _answer_sdp; - mutable RtcSession _rtsp_send_sdp; - mutable uint8_t _send_rtp_pt[2] = {0, 0}; - RtspMediaSourceImp::Ptr _push_src; - unordered_map _rtp_receiver; - unordered_map _ssrc_info; uint32_t _recv_video_ssrc; + mutable uint8_t _send_rtp_pt[2] = {0, 0}; Ticker _pli_ticker; + Socket::Ptr _socket; + RtcSession _answer_sdp; + RtspMediaSource::Ptr _src; + mutable RtcSession _rtsp_send_sdp; + RtspMediaSource::RingType::RingReader::Ptr _reader; + unordered_map _rtp_info_pt; + unordered_map _rtp_info_ssrc; }; From 6070654860a2ceaddaa348bb8895ddfe3f8927be Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 4 Apr 2021 23:58:59 +0800 Subject: [PATCH 062/218] =?UTF-8?q?rtc=E6=92=AD=E6=94=BE=E6=97=B6=EF=BC=8C?= =?UTF-8?q?=E6=AF=94=E5=AF=B9=E9=87=87=E6=A0=B7=E7=8E=87=E7=AD=89=E4=BF=A1?= =?UTF-8?q?=E6=81=AF?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 47 +++++++++++++++++++++++++++++++++----- webrtc/Sdp.h | 9 +++++++- webrtc/WebRtcTransport.cpp | 7 ++++-- 3 files changed, 54 insertions(+), 9 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index f83b2b7e..91d822bf 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1331,6 +1331,10 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const if (!configure.enable) { return; } + bool failed = false; + +RETRY: + for (auto &codec : configure.preferred_codec) { for (auto &offer_media : medias) { if (offer_media.type != type) { @@ -1342,12 +1346,15 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const } const RtcCodecPlan *offer_plan_ptr = nullptr; for (auto &plan : offer_media.plan) { - if (getCodecId(plan.codec) != codec) { - continue; - } - //命中偏好的编码格式 - if (!onMatchCodecPlan(plan, codec)) { - continue; + if (!failed) { + //如果匹配失败了，那么随便选择一个plan + if (getCodecId(plan.codec) != codec) { + continue; + } + //命中偏好的编码格式 + if (!onMatchCodecPlan(plan, codec)) { + continue; + } } //找到中意的codec offer_plan_ptr = &plan; @@ -1476,4 +1483,32 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const return; } } + + if (!failed) { + //只重试一次 + failed = true; + goto RETRY; + } +} + +void RtcConfigure::setPlayRtspInfo(const string &sdp){ + RtcSession session; + session.loadFrom(sdp, false); + for (auto &m : session.media) { + switch (m.type) { + case TrackVideo : _rtsp_video_plan = std::make_shared(m.plan[0]); break; + case TrackAudio : _rtsp_audio_plan = std::make_shared(m.plan[0]); break; + default: break; + } + } +} + +bool RtcConfigure::onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec){ + if (_rtsp_audio_plan && codec == getCodecId(_rtsp_audio_plan->codec)) { + if (plan.sample_rate != _rtsp_audio_plan->sample_rate || plan.channel != _rtsp_audio_plan->channel) { + //音频采样率和通道数必须相同 + return false; + } + } + return true; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 0d0f972a..e7ba518e 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -574,6 +574,7 @@ public: //rtc传输编码方案 class RtcCodecPlan{ public: + using Ptr = shared_ptr; uint8_t pt; string codec; uint32_t sample_rate; @@ -702,9 +703,15 @@ public: shared_ptr createAnswer(const RtcSession &offer); + void setPlayRtspInfo(const string &sdp); + private: void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); - bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec) { return true; } + bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec); + +private: + RtcCodecPlan::Ptr _rtsp_video_plan; + RtcCodecPlan::Ptr _rtsp_audio_plan; }; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 0e32859b..7aa6417b 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -324,16 +324,19 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { //这是播放 configure.video.direction = RtpDirection::sendonly; configure.audio.direction = RtpDirection::sendonly; + configure.setPlayRtspInfo(_src->getSdp()); _rtsp_send_sdp.loadFrom(_src->getSdp(), false); //根据rtsp流的相关信息，设置rtc最佳编码 for (auto &m : _rtsp_send_sdp.media) { switch (m.type) { case TrackVideo: { - configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec)); + configure.video.preferred_codec.clear(); + configure.video.preferred_codec.emplace_back(getCodecId(m.plan[0].codec)); break; } case TrackAudio: { - configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec)); + configure.audio.preferred_codec.clear(); + configure.audio.preferred_codec.emplace_back(getCodecId(m.plan[0].codec)); break; } default: From 3c935e7fddc951d4c27bf769adb35f6be1478242 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 5 Apr 2021 00:06:21 +0800 Subject: [PATCH 063/218] =?UTF-8?q?=E4=BC=98=E5=8C=96=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 14 ++++++++++++-- webrtc/WebRtcTransport.cpp | 26 ++++++-------------------- webrtc/WebRtcTransport.h | 1 - 3 files changed, 18 insertions(+), 23 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 91d822bf..42e85241 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1496,8 +1496,18 @@ void RtcConfigure::setPlayRtspInfo(const string &sdp){ session.loadFrom(sdp, false); for (auto &m : session.media) { switch (m.type) { - case TrackVideo : _rtsp_video_plan = std::make_shared(m.plan[0]); break; - case TrackAudio : _rtsp_audio_plan = std::make_shared(m.plan[0]); break; + case TrackVideo : { + _rtsp_video_plan = std::make_shared(m.plan[0]); + video.preferred_codec.clear(); + video.preferred_codec.emplace_back(getCodecId(_rtsp_video_plan->codec)); + break; + } + case TrackAudio : { + _rtsp_audio_plan = std::make_shared(m.plan[0]); + audio.preferred_codec.clear(); + audio.preferred_codec.emplace_back(getCodecId(_rtsp_audio_plan->codec)); + break; + } default: break; } } diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 7aa6417b..2dbbef3b 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -298,11 +298,14 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ return; } - //添加answer sdp的ssrc信息，并且记录发送rtp的pt + RtcSession rtsp_send_sdp; + rtsp_send_sdp.loadFrom(_src->getSdp(), false); + for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; } + //添加answer sdp的ssrc信息 m.rtp_ssrc.ssrc = _src->getSsrc(m.type); m.rtp_ssrc.cname = RTP_CNAME; //todo 先屏蔽rtx，因为chrome报错 @@ -310,8 +313,9 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc; m.rtx_ssrc.cname = RTX_CNAME; } - auto rtsp_media = _rtsp_send_sdp.getMedia(m.type); + auto rtsp_media = rtsp_send_sdp.getMedia(m.type); if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { + //记录发送rtp的pt _send_rtp_pt[m.type] = m.plan[0].pt; } } @@ -325,24 +329,6 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { configure.video.direction = RtpDirection::sendonly; configure.audio.direction = RtpDirection::sendonly; configure.setPlayRtspInfo(_src->getSdp()); - _rtsp_send_sdp.loadFrom(_src->getSdp(), false); - //根据rtsp流的相关信息，设置rtc最佳编码 - for (auto &m : _rtsp_send_sdp.media) { - switch (m.type) { - case TrackVideo: { - configure.video.preferred_codec.clear(); - configure.video.preferred_codec.emplace_back(getCodecId(m.plan[0].codec)); - break; - } - case TrackAudio: { - configure.audio.preferred_codec.clear(); - configure.audio.preferred_codec.emplace_back(getCodecId(m.plan[0].codec)); - break; - } - default: - break; - } - } } else { //这是推流 configure.video.direction = RtpDirection::recvonly; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 000c4cc7..28086a01 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -159,7 +159,6 @@ private: Socket::Ptr _socket; RtcSession _answer_sdp; RtspMediaSource::Ptr _src; - mutable RtcSession _rtsp_send_sdp; RtspMediaSource::RingType::RingReader::Ptr _reader; unordered_map _rtp_info_pt; unordered_map _rtp_info_ssrc; From ce04a7279c28eae9aebf67d67c2c9f6170d3db04 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 5 Apr 2021 00:12:46 +0800 Subject: [PATCH 064/218] =?UTF-8?q?=E4=BC=98=E5=8C=96=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 4 ++-- webrtc/WebRtcTransport.h | 16 +++++++++++----- 2 files changed, 13 insertions(+), 7 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 2dbbef3b..bf79e477 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -102,7 +102,7 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){ _dtls_transport->SetRemoteFingerprint(remote_fingerprint); } -void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp) const{ +void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ for (auto &m : sdp.media) { if (m.type != TrackApplication && !m.rtcp_mux) { throw std::invalid_argument("只支持rtcp-mux模式"); @@ -292,7 +292,7 @@ void WebRtcTransportImp::onStartWebRTC() { } } -void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{ +void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer || !canSendRtp()) { return; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 28086a01..6ba2cb4e 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -79,7 +79,7 @@ protected: protected: virtual void onStartWebRTC() = 0; virtual void onRtcConfigure(RtcConfigure &configure) const {} - virtual void onCheckSdp(SdpType type, RtcSession &sdp) const; + virtual void onCheckSdp(SdpType type, RtcSession &sdp); virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0; virtual void onRtp(const char *buf, size_t len) = 0; @@ -125,7 +125,7 @@ public: protected: void onStartWebRTC() override; void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override; - void onCheckSdp(SdpType type, RtcSession &sdp) const override; + void onCheckSdp(SdpType type, RtcSession &sdp) override; void onRtcConfigure(RtcConfigure &configure) const override; void onRtp(const char *buf, size_t len) override; @@ -153,14 +153,20 @@ private: void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: - uint32_t _recv_video_ssrc; - mutable uint8_t _send_rtp_pt[2] = {0, 0}; + //pli rtcp计时器 Ticker _pli_ticker; + //rtc rtp推流的视频ssrc + uint32_t _recv_video_ssrc; + //记录协商的rtp的pt类型 + uint8_t _send_rtp_pt[2] = {0, 0}; + //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; - RtcSession _answer_sdp; + //推流或播放的rtsp源 RtspMediaSource::Ptr _src; RtspMediaSource::RingType::RingReader::Ptr _reader; + //根据rtp的pt获取相关信息 unordered_map _rtp_info_pt; + //根据推流端rtp的ssrc获取相关信息 unordered_map _rtp_info_ssrc; }; From 606f25131130159bfa24662281b5b4358235a60e Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 5 Apr 2021 11:07:41 +0800 Subject: [PATCH 065/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E9=9F=B3=E9=A2=91?= =?UTF-8?q?=E7=9B=B8=E5=85=B3=E7=9A=84bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 42e85241..6c55c9a1 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -558,6 +558,10 @@ void SdpAttrRtpMap::parse(const string &str) { if (sscanf(str.data(), "%" SCNu8 " %31[^/]/%" SCNd32, &pt, buf, &sample_rate) != 3) { SDP_THROW(); } + if (getTrackType(getCodecId(buf)) == TrackAudio) { + //未指定通道数时，且为音频时，那么通道数默认为1 + channel = 1; + } } codec = buf; } @@ -890,7 +894,7 @@ void RtcSession::loadFrom(const string &str, bool check) { auto rtpmap_it = rtpmap_map.find(pt); if (rtpmap_it == rtpmap_map.end()) { plan.pt = pt; - plan.codec = RtpPayload::getCodecId(pt); + plan.codec = RtpPayload::getName(pt); plan.sample_rate = RtpPayload::getClockRate(pt); plan.channel = RtpPayload::getAudioChannel(pt); } else { From 50ca789c0c617e2b3593352e6b008e26ecdf9b9c Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 5 Apr 2021 11:32:38 +0800 Subject: [PATCH 066/218] =?UTF-8?q?=E9=A2=84=E7=95=99=E5=90=8C=E6=97=B6?= =?UTF-8?q?=E6=8E=A8=E6=B5=81=E6=8B=89=E6=B5=81=E7=9A=84=E6=8E=A5=E5=8F=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 4 ++-- webrtc/WebRtcTransport.cpp | 40 +++++++++++++++++++++++--------------- webrtc/WebRtcTransport.h | 10 +++++++--- 3 files changed, 33 insertions(+), 21 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 1ae2b0cd..f94d0d59 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1105,7 +1105,7 @@ void installWebApi() { throw runtime_error(StrPrinter << "播放鉴权失败:" << err); } auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); - rtc->attach(src); + rtc->attach(src, true); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; rtcs.emplace_back(rtc); @@ -1139,7 +1139,7 @@ void installWebApi() { auto push_src = std::make_shared(info._vhost, info._app, info._streamid); push_src->setProtocolTranslation(enableHls, enableMP4); auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); - rtc->attach(push_src); + rtc->attach(push_src, false); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; rtcs.emplace_back(rtc); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index bf79e477..ee0558f9 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -224,9 +224,13 @@ void WebRtcTransportImp::onDestory() { WebRtcTransport::onDestory(); } -void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) { +void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, bool is_play) { assert(src); - _src = src; + if (is_play) { + _play_src = src; + } else { + _push_src = src; + } } void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) { @@ -239,12 +243,12 @@ void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sock bool WebRtcTransportImp::canSendRtp() const{ auto &sdp = getSdp(SdpType::answer); - return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly; + return _play_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly); } bool WebRtcTransportImp::canRecvRtp() const{ auto &sdp = getSdp(SdpType::answer); - return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly; + return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly); } void WebRtcTransportImp::onStartWebRTC() { @@ -274,10 +278,10 @@ void WebRtcTransportImp::onStartWebRTC() { } if (canRecvRtp()) { - _src->setSdp(getSdp(SdpType::answer).toRtspSdp()); + _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp()); } if (canSendRtp()) { - _reader = _src->getRing()->attach(_socket->getPoller(), true); + _reader = _play_src->getRing()->attach(_socket->getPoller(), true); weak_ptr weak_self = shared_from_this(); _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) { auto strongSelf = weak_self.lock(); @@ -299,14 +303,14 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ } RtcSession rtsp_send_sdp; - rtsp_send_sdp.loadFrom(_src->getSdp(), false); + rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; } //添加answer sdp的ssrc信息 - m.rtp_ssrc.ssrc = _src->getSsrc(m.type); + m.rtp_ssrc.ssrc = _play_src->getSsrc(m.type); m.rtp_ssrc.cname = RTP_CNAME; //todo 先屏蔽rtx，因为chrome报错 if (false && m.getRelatedRtxPlan(m.plan[0].pt)) { @@ -324,15 +328,17 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const { WebRtcTransport::onRtcConfigure(configure); - if (!_src->getSdp().empty()) { - //这是播放 - configure.video.direction = RtpDirection::sendonly; - configure.audio.direction = RtpDirection::sendonly; - configure.setPlayRtspInfo(_src->getSdp()); - } else { - //这是推流 + if (_play_src) { + //这是播放,同时也可能有推流 + configure.video.direction = _push_src ? RtpDirection::sendrecv : RtpDirection::sendonly; + configure.audio.direction = configure.video.direction; + configure.setPlayRtspInfo(_play_src->getSdp()); + } else if (_push_src) { + //这只是推流 configure.video.direction = RtpDirection::recvonly; configure.audio.direction = RtpDirection::recvonly; + } else { + throw std::invalid_argument("未设置播放或推流的媒体源"); } //添加接收端口candidate信息 @@ -455,7 +461,9 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true); InfoL << "send pli"; } - _src->onWrite(std::move(rtp), false); + if (_push_src) { + _push_src->onWrite(std::move(rtp), false); + } } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 6ba2cb4e..125771d8 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -119,8 +119,9 @@ public: /** * 绑定rtsp媒体源 * @param src 媒体源 + * @param is_play 是播放还是推流 */ - void attach(const RtspMediaSource::Ptr &src); + void attach(const RtspMediaSource::Ptr &src, bool is_play = true); protected: void onStartWebRTC() override; @@ -161,8 +162,11 @@ private: uint8_t _send_rtp_pt[2] = {0, 0}; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; - //推流或播放的rtsp源 - RtspMediaSource::Ptr _src; + //推流的rtsp源 + RtspMediaSource::Ptr _push_src; + //播放的rtsp源 + RtspMediaSource::Ptr _play_src; + //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 unordered_map _rtp_info_pt; From 8edd093d68c050ab4f2e01404ee948f057c8c6a8 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 6 Apr 2021 17:23:13 +0800 Subject: [PATCH 067/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0bom=E5=A4=B4?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/DtlsTransport.cpp | 2 +- webrtc/DtlsTransport.hpp | 2 +- webrtc/IceServer.cpp | 2 +- webrtc/IceServer.hpp | 7 +++++-- webrtc/Sdp.cpp | 2 +- webrtc/Sdp.h | 2 +- webrtc/SrtpSession.cpp | 2 +- webrtc/SrtpSession.hpp | 2 +- webrtc/StunPacket.cpp | 2 +- webrtc/StunPacket.hpp | 2 +- webrtc/Utils.hpp | 2 +- webrtc/WebRtcTransport.cpp | 2 +- webrtc/WebRtcTransport.h | 2 +- webrtc/logger.h | 2 +- 14 files changed, 18 insertions(+), 15 deletions(-) diff --git a/webrtc/DtlsTransport.cpp b/webrtc/DtlsTransport.cpp index f627fd29..0dd9492c 100644 --- a/webrtc/DtlsTransport.cpp +++ b/webrtc/DtlsTransport.cpp @@ -1,4 +1,4 @@ -#define MS_CLASS "RTC::DtlsTransport" +#define MS_CLASS "RTC::DtlsTransport" // #define MS_LOG_DEV_LEVEL 3 #include "DtlsTransport.hpp" diff --git a/webrtc/DtlsTransport.hpp b/webrtc/DtlsTransport.hpp index 41fa4c09..2d324441 100644 --- a/webrtc/DtlsTransport.hpp +++ b/webrtc/DtlsTransport.hpp @@ -1,4 +1,4 @@ -#ifndef MS_RTC_DTLS_TRANSPORT_HPP +#ifndef MS_RTC_DTLS_TRANSPORT_HPP #define MS_RTC_DTLS_TRANSPORT_HPP #include "SrtpSession.hpp" diff --git a/webrtc/IceServer.cpp b/webrtc/IceServer.cpp index d80ef0f0..15aeb0bd 100644 --- a/webrtc/IceServer.cpp +++ b/webrtc/IceServer.cpp @@ -1,4 +1,4 @@ -#define MS_CLASS "RTC::IceServer" +#define MS_CLASS "RTC::IceServer" // #define MS_LOG_DEV_LEVEL 3 #include diff --git a/webrtc/IceServer.hpp b/webrtc/IceServer.hpp index 103727f0..49045f43 100644 --- a/webrtc/IceServer.hpp +++ b/webrtc/IceServer.hpp @@ -1,16 +1,19 @@ -#ifndef MS_RTC_ICE_SERVER_HPP +#ifndef MS_RTC_ICE_SERVER_HPP #define MS_RTC_ICE_SERVER_HPP #include "StunPacket.hpp" #include "logger.h" +#include "Utils.hpp" #include #include #include #include +using _TransportTuple = struct sockaddr; + namespace RTC { - using TransportTuple = struct sockaddr; + using TransportTuple = _TransportTuple; class IceServer { public: diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 6c55c9a1..b25eabb4 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1,4 +1,4 @@ -// +// // Created by xzl on 2021/3/27. // diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index e7ba518e..b8328577 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -1,4 +1,4 @@ -// +// // Created by xzl on 2021/3/27. // diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index 64c1bca0..a73c5aaa 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -1,4 +1,4 @@ -#define MS_CLASS "RTC::SrtpSession" +#define MS_CLASS "RTC::SrtpSession" // #define MS_LOG_DEV_LEVEL 3 #include "SrtpSession.hpp" diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index 5b3edefd..988d730b 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -1,4 +1,4 @@ -#ifndef MS_RTC_SRTP_SESSION_HPP +#ifndef MS_RTC_SRTP_SESSION_HPP #define MS_RTC_SRTP_SESSION_HPP #include "Utils.hpp" diff --git a/webrtc/StunPacket.cpp b/webrtc/StunPacket.cpp index 019fadd9..223d3f57 100644 --- a/webrtc/StunPacket.cpp +++ b/webrtc/StunPacket.cpp @@ -1,4 +1,4 @@ -#define MS_CLASS "RTC::StunPacket" +#define MS_CLASS "RTC::StunPacket" // #define MS_LOG_DEV_LEVEL 3 #include "StunPacket.hpp" diff --git a/webrtc/StunPacket.hpp b/webrtc/StunPacket.hpp index 8f65aab6..ebd1b1d0 100644 --- a/webrtc/StunPacket.hpp +++ b/webrtc/StunPacket.hpp @@ -1,4 +1,4 @@ -#ifndef MS_RTC_STUN_PACKET_HPP +#ifndef MS_RTC_STUN_PACKET_HPP #define MS_RTC_STUN_PACKET_HPP diff --git a/webrtc/Utils.hpp b/webrtc/Utils.hpp index 6b33b9f5..0b1b66ad 100644 --- a/webrtc/Utils.hpp +++ b/webrtc/Utils.hpp @@ -1,4 +1,4 @@ -#ifndef MS_UTILS_HPP +#ifndef MS_UTILS_HPP #define MS_UTILS_HPP #if defined(_WIN32) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index ee0558f9..bad1648e 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -1,4 +1,4 @@ -#include "WebRtcTransport.h" +#include "WebRtcTransport.h" #include #include "Rtcp/Rtcp.h" #include "Rtsp/RtpReceiver.h" diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 125771d8..db84b186 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -1,4 +1,4 @@ -#pragma once +#pragma once #include #include diff --git a/webrtc/logger.h b/webrtc/logger.h index 6b345e98..67da6219 100644 --- a/webrtc/logger.h +++ b/webrtc/logger.h @@ -1,4 +1,4 @@ -#pragma once +#pragma once #include #include From 9c552128a7e7c95bf4bcb25418193dcd1736eed3 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 6 Apr 2021 18:29:14 +0800 Subject: [PATCH 068/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dsdp=E4=B8=AD=E6=B2=A1?= =?UTF-8?q?ssrc=E6=97=B6=E8=A7=A6=E5=8F=91=E7=9A=84bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index b25eabb4..8951e0ee 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -841,7 +841,9 @@ void RtcSession::loadFrom(const string &str, bool check) { if (rtc_ssrc_map.size() > 1) { throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却有多个"); } - ssrc_rtp = rtc_ssrc_map.begin()->second.ssrc; + if (rtc_ssrc_map.size() == 1) { + ssrc_rtp = rtc_ssrc_map.begin()->second.ssrc; + } } for (auto &pr : rtc_ssrc_map) { auto &rtc_ssrc = pr.second; @@ -982,6 +984,7 @@ string RtcSession::toRtspSdp() const{ } } + copy.session_name = "zlmediakit rtsp stream from webrtc"; auto sdp = copy.toRtcSessionSdp(); toRtsp(sdp->items); int i = 0; From b5bf930467cc14edd22577dd90662f64f6c52120 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 6 Apr 2021 18:30:36 +0800 Subject: [PATCH 069/218] =?UTF-8?q?=E8=B0=83=E6=95=B4cmake?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- CMakeLists.txt | 21 +++++++++++++++++---- cmake/FindSRTP.cmake | 4 ++-- webrtc/CMakeLists.txt | 16 ---------------- 3 files changed, 19 insertions(+), 22 deletions(-) delete mode 100644 webrtc/CMakeLists.txt diff --git a/CMakeLists.txt b/CMakeLists.txt index 36a93a12..54bfed8a 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -225,10 +225,23 @@ if(ENABLE_API) add_subdirectory(api) endif() -if(ENABLE_WEBRTC) - add_definitions(-DENABLE_WEBRTC) - add_subdirectory(webrtc) -endif() +if (ENABLE_WEBRTC) + #查找srtp是否安装 + find_package(SRTP QUIET) + if (SRTP_FOUND) + message(STATUS "found library:${SRTP_LIBRARIES}") + include_directories(${SRTP_INCLUDE_DIRS}) + list(APPEND LINK_LIB_LIST ${SRTP_LIBRARIES}) + + add_definitions(-DENABLE_WEBRTC) + include_directories(./webrtc) + file(GLOB SRC_WEBRTC_LIST ./webrtc/*.cpp ./webrtc/*.h ./webrtc/*.hpp) + add_library(webrtc ${SRC_WEBRTC_LIST}) + list(APPEND LINK_LIB_LIST webrtc) + else () + message(WARNING "srtp未找到, webrtc相关功能打开失败") + endif () +endif () if (NOT IOS) #测试程序 diff --git a/cmake/FindSRTP.cmake b/cmake/FindSRTP.cmake index 3046020e..8d88f405 100644 --- a/cmake/FindSRTP.cmake +++ b/cmake/FindSRTP.cmake @@ -32,7 +32,7 @@ set(_SRTP_ROOT_PATHS find_path(SRTP_INCLUDE_DIRS NAMES srtp2/srtp.h - HINTS _SRTP_ROOT_PATHS + HINTS _SRTP_ROOT_PATHS ${SRTP_PREFIX} PATH_SUFFIXES include ) @@ -42,7 +42,7 @@ endif() find_library(SRTP_LIBRARIES NAMES srtp2 - HINTS ${_SRTP_ROOT_PATHS} + HINTS ${_SRTP_ROOT_PATHS} ${SRTP_PREFIX} PATH_SUFFIXES bin lib ) diff --git a/webrtc/CMakeLists.txt b/webrtc/CMakeLists.txt deleted file mode 100644 index db7dd3a8..00000000 --- a/webrtc/CMakeLists.txt +++ /dev/null @@ -1,16 +0,0 @@ -list(APPEND LINK_LIB_LIST webrtc) -#查找srtp是否安装 -find_package(SRTP QUIET) -if (SRTP_FOUND) - message(STATUS "found library:${SRTP_LIBRARIES}") - include_directories(${SRTP_INCLUDE_DIRS}) - list(APPEND LINK_LIB_LIST ${SRTP_LIBRARIES}) -else () - message(FATAL_ERROR "srtp未找到!") -endif () - -include_directories(./) -file(GLOB SRC_LIST ./*.cpp ./*.h ./*.hpp) -add_library(webrtc ${SRC_LIST}) -set(LINK_LIB_LIST ${LINK_LIB_LIST} PARENT_SCOPE) - From a42b56ba94256aaeb319c9a199edc4ccd1e709b4 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 6 Apr 2021 22:51:16 +0800 Subject: [PATCH 070/218] =?UTF-8?q?=E5=85=BC=E5=AE=B9Firefox?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 8 ++++++-- webrtc/Sdp.h | 1 + 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 8951e0ee..fc7f2154 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -604,9 +604,10 @@ void SdpAttrFmtp::parse(const string &str) { trim(item); auto pos = item.find('='); if(pos == string::npos){ - SDP_THROW(); + arr.emplace_back(std::make_pair(item, "")); + } else { + arr.emplace_back(std::make_pair(item.substr(0, pos), item.substr(pos + 1))); } - arr.emplace_back(std::make_pair(item.substr(0, pos), item.substr(pos + 1))); } if (arr.empty()) { SDP_THROW(); @@ -784,6 +785,9 @@ void RtcSession::loadFrom(const string &str, bool check) { rtc_media.ice_pwd = media.getStringItem('a', "ice-pwd"); rtc_media.role = media.getItemClass('a', "setup").role; rtc_media.fingerprint = media.getItemClass('a', "fingerprint"); + if (rtc_media.fingerprint.empty()) { + rtc_media.fingerprint = sdp.getItemClass('a', "fingerprint"); + } rtc_media.ice_lite = media.getItem('a', "ice-lite").operator bool(); auto ice_options = media.getItemClass('a', "ice-options"); rtc_media.ice_trickle = ice_options.trickle; diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index b8328577..7fc0b0b9 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -301,6 +301,7 @@ public: void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "fingerprint";} + bool empty() const { return algorithm.empty() || hash.empty(); } }; class SdpAttrSetup : public SdpItem { From 204ef6e204b8f8ef630934893c5616cc321d8ce7 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 6 Apr 2021 23:21:46 +0800 Subject: [PATCH 071/218] =?UTF-8?q?=E5=8C=B9=E9=85=8DH264=20profile?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index fc7f2154..81355d0b 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1524,12 +1524,41 @@ void RtcConfigure::setPlayRtspInfo(const string &sdp){ } } +static map toMap(const vector > &fmt) { + map ret; + for (auto &pr : fmt) { + ret.emplace(pr); + } + return ret; +} + +static const string kProfile{"profile-level-id"}; +static const string kMode{"packetization-mode"}; + bool RtcConfigure::onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec){ if (_rtsp_audio_plan && codec == getCodecId(_rtsp_audio_plan->codec)) { if (plan.sample_rate != _rtsp_audio_plan->sample_rate || plan.channel != _rtsp_audio_plan->channel) { //音频采样率和通道数必须相同 return false; } + return true; } + if (_rtsp_video_plan && codec == CodecH264 && getCodecId(_rtsp_video_plan->codec) == CodecH264) { + //h264时，匹配packetization-mode和profile-level-id + auto rtc_fmt_map = toMap(plan.fmtp); + auto rtsp_fmt_map = toMap(_rtsp_video_plan->fmtp); + auto &profile = rtsp_fmt_map[kProfile]; + if (!profile.empty() && profile != rtc_fmt_map[kProfile]) { + //profile-level-id 不匹配 + return false; + } + auto &mode = rtsp_fmt_map[kMode]; + if (!mode.empty() && mode != rtc_fmt_map[kMode]) { + //packetization-mode不匹配 + return false; + } + return true; + } + return true; } From c9a20eda16b6ffdebfa036cb49f16d55991191dd Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 6 Apr 2021 23:22:22 +0800 Subject: [PATCH 072/218] =?UTF-8?q?profile=E5=8C=B9=E9=85=8D=E5=A4=B1?= =?UTF-8?q?=E8=B4=A5=E6=97=B6=EF=BC=8C=E4=BC=98=E5=85=88=E9=80=89=E6=8B=A9?= =?UTF-8?q?=E7=9B=B8=E5=90=8C=E7=BC=96=E7=A0=81=E6=A0=BC=E5=BC=8F=E7=9A=84?= =?UTF-8?q?=E6=96=B9=E6=A1=88?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 81355d0b..8a5f6f02 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1357,11 +1357,11 @@ RETRY: } const RtcCodecPlan *offer_plan_ptr = nullptr; for (auto &plan : offer_media.plan) { + if (getCodecId(plan.codec) != codec) { + continue; + } + //如果匹配失败了，那么随便选择一个plan if (!failed) { - //如果匹配失败了，那么随便选择一个plan - if (getCodecId(plan.codec) != codec) { - continue; - } //命中偏好的编码格式 if (!onMatchCodecPlan(plan, codec)) { continue; From b5e6c12ee8f073ef46abf7a130d0bd61954b9760 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 6 Apr 2021 23:46:45 +0800 Subject: [PATCH 073/218] =?UTF-8?q?H264=20profile=E6=AF=94=E5=AF=B9?= =?UTF-8?q?=E4=B8=8D=E5=8C=BA=E5=88=86=E5=A4=A7=E5=B0=8F=E5=86=99?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 6 +++--- webrtc/Sdp.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 8a5f6f02..38506b5e 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1535,7 +1535,7 @@ static map toMap(const vectorcodec)) { if (plan.sample_rate != _rtsp_audio_plan->sample_rate || plan.channel != _rtsp_audio_plan->channel) { //音频采样率和通道数必须相同 @@ -1548,12 +1548,12 @@ bool RtcConfigure::onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec){ auto rtc_fmt_map = toMap(plan.fmtp); auto rtsp_fmt_map = toMap(_rtsp_video_plan->fmtp); auto &profile = rtsp_fmt_map[kProfile]; - if (!profile.empty() && profile != rtc_fmt_map[kProfile]) { + if (!profile.empty() && strcasecmp(profile.data(), rtc_fmt_map[kProfile].data())) { //profile-level-id 不匹配 return false; } auto &mode = rtsp_fmt_map[kMode]; - if (!mode.empty() && mode != rtc_fmt_map[kMode]) { + if (!mode.empty() && atoi(mode.data()) != atoi(rtc_fmt_map[kMode].data())) { //packetization-mode不匹配 return false; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 7fc0b0b9..2a9eb6ee 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -708,7 +708,7 @@ public: private: void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); - bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec); + bool onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec); private: RtcCodecPlan::Ptr _rtsp_video_plan; From 79267ae5512ba6f1ab5a58428caa06f2d72b4e6a Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 6 Apr 2021 23:47:54 +0800 Subject: [PATCH 074/218] =?UTF-8?q?=E6=AF=94=E5=AF=B9codec=E5=A4=B1?= =?UTF-8?q?=E8=B4=A5=E6=97=B6=EF=BC=8C=E5=85=88=E5=85=B3=E9=97=ADprofile?= =?UTF-8?q?=E6=AF=94=E5=AF=B9=EF=BC=8C=E5=86=8D=E5=85=B3=E9=97=ADcodec?= =?UTF-8?q?=E6=AF=94=E5=AF=B9?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 27 ++++++++++++++++----------- 1 file changed, 16 insertions(+), 11 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 38506b5e..490890b0 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1342,7 +1342,8 @@ void RtcConfigure::matchMedia(shared_ptr &ret, TrackType type, const if (!configure.enable) { return; } - bool failed = false; + bool check_profile = true; + bool check_codec = true; RETRY: @@ -1357,15 +1358,13 @@ RETRY: } const RtcCodecPlan *offer_plan_ptr = nullptr; for (auto &plan : offer_media.plan) { - if (getCodecId(plan.codec) != codec) { + //先检查编码格式是否为偏好 + if (check_codec && getCodecId(plan.codec) != codec) { continue; } - //如果匹配失败了，那么随便选择一个plan - if (!failed) { - //命中偏好的编码格式 - if (!onMatchCodecPlan(plan, codec)) { - continue; - } + //命中偏好的编码格式,然后检查规格 + if (check_profile && !onCheckCodecProfile(plan, codec)) { + continue; } //找到中意的codec offer_plan_ptr = &plan; @@ -1495,9 +1494,15 @@ RETRY: } } - if (!failed) { - //只重试一次 - failed = true; + if (check_profile) { + //如果是由于检查profile导致匹配失败，那么重试一次，且不检查profile + check_profile = false; + goto RETRY; + } + + if (check_codec) { + //如果是由于检查codec导致匹配失败，那么重试一次，且不检查codec + check_codec = false; goto RETRY; } } From f374f723122d3d75427999b9a79d0c13ffa1a95a Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 16:45:50 +0800 Subject: [PATCH 075/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtcp=20bye=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 116 +++++++++++++++++++++++++++++++++++++++++++++- src/Rtcp/Rtcp.h | 70 ++++++++++++++++++++++++++-- 2 files changed, 182 insertions(+), 4 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 220a8439..86af2821 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -9,6 +9,7 @@ */ #include +#include #include "Rtcp.h" #include "Util/logger.h" @@ -87,10 +88,20 @@ string RtcpHeader::dumpString() const { RtcpPli *rtcp = (RtcpPli *)this; return rtcp->dumpString(); } + + case RtcpType::RTCP_BYE: { + RtcpBye *rtcp = (RtcpBye *)this; + return rtcp->dumpString(); + } + default: return StrPrinter << dumpHeader() << hexdump((char *)this + sizeof(*this), length << 2); } } +size_t RtcpHeader::getSize() const { + return (1 + length) << 2; +} + void RtcpHeader::net2Host(size_t len){ switch ((RtcpType)pt) { case RtcpType::RTCP_SR: { @@ -116,6 +127,12 @@ void RtcpHeader::net2Host(size_t len){ pli->net2Host(len); break; } + + case RtcpType::RTCP_BYE: { + RtcpBye *bye = (RtcpBye *)this; + bye->net2Host(len); + break; + } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); } } @@ -127,6 +144,10 @@ vector RtcpHeader::loadFromBytes(char *data, size_t len){ while (remain > (ssize_t) sizeof(RtcpHeader)) { RtcpHeader *rtcp = (RtcpHeader *) ptr; auto rtcp_len = (1 + ntohs(rtcp->length)) << 2; + if (remain < rtcp_len) { + WarnL << "非法的rtcp包,声明的长度超过实际数据长度"; + break; + } try { rtcp->net2Host(rtcp_len); ret.emplace_back(rtcp); @@ -144,7 +165,7 @@ class BufferRtcp : public Buffer { public: BufferRtcp(std::shared_ptr rtcp) { _rtcp = std::move(rtcp); - _size = (htons(_rtcp->length) + 1) << 2; + _size = (ntohs(_rtcp->length) + 1) << 2; } ~BufferRtcp() override {} @@ -440,4 +461,97 @@ void RtcpPli::net2Host(size_t size) { ssrc_media = ntohl(ssrc_media); } +//////////////////////////////////////////////////////////////////// + +std::shared_ptr RtcpBye::create(const std::initializer_list &ssrcs, const string &reason) { + assert(reason.size() <= 0xFF); + auto bytes = alignSize(sizeof(RtcpHeader) + sizeof(uint32_t) * ssrcs.size() + 1 + reason.size()); + auto ptr = (RtcpBye *) new char[bytes]; + setupHeader(ptr, RtcpType::RTCP_BYE, ssrcs.size(), bytes); + + auto *ssrc_ptr = &(((RtcpBye *) ptr)->ssrc); + for (auto ssrc : ssrcs) { + *ssrc_ptr = htonl(ssrc); + ++ssrc_ptr; + } + + if (!reason.empty()) { + uint8_t *reason_len_ptr = (uint8_t *) ptr + sizeof(RtcpHeader) + sizeof(uint32_t) * ssrcs.size(); + *reason_len_ptr = reason.size() & 0xFF; + memcpy(reason_len_ptr + 1, reason.data(), *reason_len_ptr); + } + + return std::shared_ptr(ptr, [](RtcpBye *ptr) { + delete[] (char *) ptr; + }); +} + +vector RtcpBye::getSSRC() { + vector ret; + uint32_t *ssrc_ptr = &ssrc; + for (size_t i = 0; i < report_count; ++i) { + ret.emplace_back(ssrc_ptr); + ssrc_ptr += 1; + } + return ret; +} + +string RtcpBye::getReason() const { + auto *reason_len_ptr = &reason_len + sizeof(ssrc) * (report_count - 1); + if (reason_len_ptr + 1 >= (uint8_t *) this + getSize()) { + return ""; + } + return string((char *) reason_len_ptr + 1, *reason_len_ptr); +} + +string RtcpBye::dumpString() const { + _StrPrinter printer; + printer << RtcpHeader::dumpHeader(); + for(auto ssrc : ((RtcpBye *)this)->getSSRC()) { + printer << "ssrc:" << *ssrc << "\r\n"; + } + printer << "reason:" << getReason(); + return std::move(printer); +} + +void RtcpBye::net2Host(size_t size) { + static const size_t kMinSize = sizeof(RtcpHeader); + CHECK_MIN_SIZE(size, kMinSize); + RtcpHeader::net2Host(); + + uint32_t *ssrc_ptr = &ssrc; + size_t offset = kMinSize; + size_t i = 0; + for (; i < report_count && offset + sizeof(ssrc) <= size; ++i) { + *ssrc_ptr = ntohl(*ssrc_ptr); + ssrc_ptr += 1; + offset += sizeof(ssrc); + } + //修正ssrc个数 + CHECK_LENGTH(size, i); + + if (offset < size) { + uint8_t *reason_len_ptr = &reason_len + sizeof(ssrc) * (report_count - 1); + if (reason_len_ptr + 1 + *reason_len_ptr > (uint8_t *) this + size) { + WarnL << "invalid rtcp bye reason length"; + //修正reason_len长度 + *reason_len_ptr = ((uint8_t *) this + size - reason_len_ptr - 1) & 0xFF; + } + } +} + +#if 0 +#include "Util/onceToken.h" + +static toolkit::onceToken token([](){ + auto bye = RtcpBye::create({1,2,3,4,5,6}, "this is a bye reason"); + auto buffer = RtcpHeader::toBuffer(bye); + + auto rtcps = RtcpHeader::loadFromBytes(buffer->data(), buffer->size()); + for(auto rtcp : rtcps){ + std::cout << rtcp->dumpString() << std::endl; + } +}); +#endif + }//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index 8fcc1a1a..c7a1556f 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -124,6 +124,12 @@ public: */ string dumpString() const; + /** + * 根据length字段获取rtcp总长度 + * 使用net2Host转换成主机字节序后才可使用此函数 + */ + size_t getSize() const; + protected: /** * 网络字节序转换为主机字节序 @@ -465,9 +471,7 @@ public: public: /** - * 创建SDES包，只赋值了RtcpHeader以及SdesItem对象的length和text部分 - * @param item_text SdesItem列表，只赋值length和text部分 - * @return SDES包 + * 创建pli包 */ static std::shared_ptr create(); @@ -485,6 +489,66 @@ private: void net2Host(size_t size); } PACKED; +//BYE +/* + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + |V=2|P| SC | PT=BYE=203 | length | + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | SSRC/CSRC | + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + : ... : + +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +(opt) | length | reason for leaving ... + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +*/ + +class RtcpBye : public RtcpHeader { +public: + friend class RtcpHeader; + /* 变长，根据count决定有多少个ssrc */ + uint32_t ssrc; + + /** 中间可能有若干个 ssrc **/ + + /* 可选 */ + uint8_t reason_len; + char reason; + +public: + /** + * 创建bye包 + * @param ssrc ssrc列表 + * @param reason 原因 + * @return rtcp bye包 + */ + static std::shared_ptr create(const std::initializer_list &ssrc, const string &reason); + + /** + * 获取ssrc列表 + */ + vector getSSRC(); + + /** + * 获取原因 + */ + string getReason() const; + +private: + /** + * 打印字段详情 + * 使用net2Host转换成主机字节序后才可使用此函数 + */ + string dumpString() const; + + /** + * 网络字节序转换为主机字节序 + * @param size 字节长度，防止内存越界 + */ + void net2Host(size_t size); +} PACKED; + #if defined(_WIN32) #pragma pack(pop) #endif // defined(_WIN32) From d354c9466c86efe6701c6212bcb495d3ffaa7d57 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 17:21:59 +0800 Subject: [PATCH 076/218] =?UTF-8?q?rtc=E6=B7=BB=E5=8A=A0=E5=AF=B9rtcp=20by?= =?UTF-8?q?e=E6=94=AF=E6=8C=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 44 ++++++++++++++++++++++++++++++-------- webrtc/WebRtcTransport.h | 10 ++++++++- 2 files changed, 44 insertions(+), 10 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index bad1648e..67d7dea8 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -12,6 +12,10 @@ WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { _ice_server = std::make_shared(this, makeRandStr(4), makeRandStr(28).substr(4)); } +void WebRtcTransport::onCreate(){ + +} + void WebRtcTransport::onDestory(){ _dtls_transport = nullptr; _ice_server = nullptr; @@ -208,18 +212,28 @@ WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &polle ptr->onDestory(); delete ptr; }); + ret->onCreate(); return ret; } -WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) { - _socket = Socket::createSocket(poller, false); +void WebRtcTransportImp::onCreate(){ + WebRtcTransport::onCreate(); + _socket = Socket::createSocket(getPoller(), false); //随机端口，绑定全部网卡 _socket->bindUdpSock(0); - _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable { - inputSockData(buf->data(), buf->size(), addr); + weak_ptr weak_self = shared_from_this(); + _socket->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable { + auto strong_self = weak_self.lock(); + if (strong_self) { + strong_self->inputSockData(buf->data(), buf->size(), addr); + } }); } +WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) { + +} + void WebRtcTransportImp::onDestory() { WebRtcTransport::onDestory(); } @@ -281,7 +295,7 @@ void WebRtcTransportImp::onStartWebRTC() { _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp()); } if (canSendRtp()) { - _reader = _play_src->getRing()->attach(_socket->getPoller(), true); + _reader = _play_src->getRing()->attach(getPoller(), true); weak_ptr weak_self = shared_from_this(); _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) { auto strongSelf = weak_self.lock(); @@ -403,7 +417,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { it->second->rtcp_context_recv->onRtcp(sr); auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); - InfoL << "send rtcp rr"; } break; } @@ -414,12 +427,21 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { if (it != _rtp_info_ssrc.end()) { auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); - InfoL << "send rtcp sr"; } break; } case RtcpType::RTCP_BYE : { - //todo 此处应该销毁对象 + //对方汇报停止发送rtp + RtcpBye *bye = (RtcpBye *) rtcp; + for (auto ssrc : bye->getSSRC()) { + auto it = _rtp_info_ssrc.find(*ssrc); + if (it == _rtp_info_ssrc.end()) { + continue; + } + _rtp_info_pt.erase(it->second->plan->pt); + _rtp_info_ssrc.erase(it); + } + onShutdown(SockException(Err_eof, "rtcp bye message received")); break; } case RtcpType::RTCP_PSFB: { @@ -459,7 +481,6 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr pli->ssrc = htonl(0); pli->ssrc_media = htonl(_recv_video_ssrc); sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true); - InfoL << "send pli"; } if (_push_src) { _push_src->onWrite(std::move(rtp), false); @@ -481,3 +502,8 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //统计rtp发送情况，好做sr汇报 _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } + +void WebRtcTransportImp::onShutdown(const SockException &ex){ + InfoL << ex.what(); +} + diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index db84b186..29fe7cb5 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -21,7 +21,12 @@ public: ~WebRtcTransport() override = default; /** - * 消费对象 + * 创建对象 + */ + virtual void onCreate(); + + /** + * 销毁对象 */ virtual void onDestory(); @@ -84,6 +89,7 @@ protected: virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; + virtual void onShutdown(const SockException &ex) = 0; protected: const RtcSession& getSdp(SdpType type) const; @@ -131,9 +137,11 @@ protected: void onRtp(const char *buf, size_t len) override; void onRtcp(const char *buf, size_t len) override; + void onShutdown(const SockException &ex) override; private: WebRtcTransportImp(const EventPoller::Ptr &poller); + void onCreate() override; void onDestory() override; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); SdpAttrCandidate::Ptr getIceCandidate() const; From b96052d387dc1c35030ca8bdc18eb17cd6fadd30 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 17:51:06 +0800 Subject: [PATCH 077/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtc=E5=BC=82=E5=B8=B8?= =?UTF-8?q?=E7=AE=A1=E7=90=86?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 3 -- webrtc/WebRtcTransport.cpp | 56 ++++++++++++++++++++++++++++++++++++-- webrtc/WebRtcTransport.h | 16 +++++++---- 3 files changed, 65 insertions(+), 10 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index f94d0d59..b1cc6935 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1081,7 +1081,6 @@ void installWebApi() { }); #ifdef ENABLE_WEBRTC - static list rtcs; api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){ CHECK_ARGS("app", "stream"); @@ -1108,7 +1107,6 @@ void installWebApi() { rtc->attach(src, true); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; - rtcs.emplace_back(rtc); invoker(200, headerOut, val.toStyledString()); } catch (std::exception &ex) { val["code"] = API::Exception; @@ -1142,7 +1140,6 @@ void installWebApi() { rtc->attach(push_src, false); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; - rtcs.emplace_back(rtc); invoker(200, headerOut, val.toStyledString()); } catch (std::exception &ex) { val["code"] = API::Exception; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 67d7dea8..6a9a925a 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -6,6 +6,21 @@ #define RTP_CNAME "zlmediakit-rtp" #define RTX_CNAME "zlmediakit-rtx" +//RTC配置项目 +namespace RTC { +#define RTC_FIELD "rtc." +//rtp和rtcp接受超时时间 +const string kTimeOutSec = RTC_FIELD"timeoutSec"; +//服务器外网ip +const string kExternIP = RTC_FIELD"externIP"; + +static onceToken token([]() { + mINI::Instance()[kTimeOutSec] = 15; + mINI::Instance()[kExternIP] = ""; +}); + +}//namespace RTC + WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) { _poller = poller; _dtls_transport = std::make_shared(poller, this); @@ -72,6 +87,23 @@ void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTran onSendSockData((char *)data, len); } +void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) { + InfoL; +} + +void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) { + InfoL; + onShutdown(SockException(Err_shutdown, "dtls transport failed")); +} + +void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) { + InfoL; + onShutdown(SockException(Err_shutdown, "dtls close notify received")); +} + +void WebRtcTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) { + InfoL << hexdump(data, len); +} ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){ @@ -228,10 +260,27 @@ void WebRtcTransportImp::onCreate(){ strong_self->inputSockData(buf->data(), buf->size(), addr); } }); + _self = shared_from_this(); + + GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec); + _timer = std::make_shared(timeoutSec / 2, [weak_self]() { + auto strong_self = weak_self.lock(); + if (!strong_self) { + return false; + } + if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) { + strong_self->onShutdown(SockException(Err_timeout, "接受rtp和rtcp超时")); + } + return true; + }, getPoller()); } WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) { + InfoL << this; +} +WebRtcTransportImp::~WebRtcTransportImp() { + InfoL << this; } void WebRtcTransportImp::onDestory() { @@ -367,8 +416,8 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ candidate->transport = "udp"; //优先级，单candidate时随便 candidate->priority = 100; - //todo 此处修改为配置文件 - candidate->address = SockUtil::get_local_ip(); + GET_CONFIG(string, extern_ip, RTC::kExternIP); + candidate->address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip; candidate->port = _socket->get_local_port(); candidate->type = "host"; return candidate; @@ -421,6 +470,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { break; } case RtcpType::RTCP_RR : { + _alive_ticker.resetTime(); //对方汇报rtp接收情况 RtcpRR *rr = (RtcpRR *) rtcp; auto it = _rtp_info_ssrc.find(rr->ssrc); @@ -454,6 +504,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + _alive_ticker.resetTime(); RtpHeader *rtp = (RtpHeader *) buf; //根据接收到的rtp的pt信息，找到该流的信息 auto it = _rtp_info_pt.find(rtp->pt); @@ -505,5 +556,6 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ void WebRtcTransportImp::onShutdown(const SockException &ex){ InfoL << ex.what(); + _self = nullptr; } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 29fe7cb5..c42317ab 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -59,7 +59,7 @@ public: protected: //// dtls相关的回调 //// - void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override {}; + void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override; void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, RTC::SrtpSession::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey, @@ -68,10 +68,10 @@ protected: size_t srtpRemoteKeyLen, std::string &remoteCert) override; - void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override {}; - void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override {}; + void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override; + void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override; void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; - void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override {}; + void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; protected: //// ice相关的回调 /// @@ -113,7 +113,7 @@ class RtpReceiverImp; class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; - ~WebRtcTransportImp() override = default; + ~WebRtcTransportImp() override; /** * 创建WebRTC对象 @@ -162,6 +162,12 @@ private: void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: + //保持自我强引用 + Ptr _self; + //检测超时的定时器 + Timer::Ptr _timer; + //刷新计时器 + Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; //rtc rtp推流的视频ssrc From 155c59e17edb31edf2b102d9501de472657c599f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 18:17:49 +0800 Subject: [PATCH 078/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtc=E7=B1=BB=E5=9E=8B?= =?UTF-8?q?=E7=9A=84=E6=8E=A8=E6=B5=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Common/MediaSource.cpp | 1 + src/Common/MediaSource.h | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/src/Common/MediaSource.cpp b/src/Common/MediaSource.cpp index ae4a47c8..67ff2994 100644 --- a/src/Common/MediaSource.cpp +++ b/src/Common/MediaSource.cpp @@ -35,6 +35,7 @@ string getOriginTypeString(MediaOriginType type){ SWITCH_CASE(ffmpeg_pull); SWITCH_CASE(mp4_vod); SWITCH_CASE(device_chn); + SWITCH_CASE(rtc_push); default : return "unknown"; } } diff --git a/src/Common/MediaSource.h b/src/Common/MediaSource.h index b85415fc..b945bc21 100644 --- a/src/Common/MediaSource.h +++ b/src/Common/MediaSource.h @@ -45,7 +45,8 @@ enum class MediaOriginType : uint8_t { pull, ffmpeg_pull, mp4_vod, - device_chn + device_chn, + rtc_push, }; string getOriginTypeString(MediaOriginType type); From 0c5fa36e4df836662dc807854d0dc9a967cbf9e1 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 18:18:27 +0800 Subject: [PATCH 079/218] =?UTF-8?q?rtc=E6=8E=A8=E6=B5=81=E6=B7=BB=E5=8A=A0?= =?UTF-8?q?=E5=AA=92=E4=BD=93=E4=BA=8B=E4=BB=B6=E7=9B=B8=E5=85=B3=E6=8E=A5?= =?UTF-8?q?=E5=8F=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 1 + webrtc/WebRtcTransport.cpp | 32 ++++++++++++++++++++++++++++++++ webrtc/WebRtcTransport.h | 16 +++++++++++++++- 3 files changed, 48 insertions(+), 1 deletion(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index b1cc6935..5fadf3bb 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1137,6 +1137,7 @@ void installWebApi() { auto push_src = std::make_shared(info._vhost, info._app, info._streamid); push_src->setProtocolTranslation(enableHls, enableMP4); auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); + push_src->setListener(rtc); rtc->attach(push_src, false); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 6a9a925a..9930d8f4 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -559,3 +559,35 @@ void WebRtcTransportImp::onShutdown(const SockException &ex){ _self = nullptr; } +///////////////////////////////////////////////////////////////////////////////////////////// + +bool WebRtcTransportImp::close(MediaSource &sender, bool force) { + //此回调在其他线程触发 + if(!_push_src || (!force && _push_src->totalReaderCount())){ + return false; + } + string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/" << sender.getApp() << "/" << sender.getId() << " " << force; + onShutdown(SockException(Err_shutdown,err)); + return true; +} + +int WebRtcTransportImp::totalReaderCount(MediaSource &sender) { + return _push_src ? _push_src->totalReaderCount() : sender.readerCount(); +} + +MediaOriginType WebRtcTransportImp::getOriginType(MediaSource &sender) const { + return MediaOriginType::rtc_push; +} + +string WebRtcTransportImp::getOriginUrl(MediaSource &sender) const { + return ""; +} + +std::shared_ptr WebRtcTransportImp::getOriginSock(MediaSource &sender) const { + return const_cast(this)->_socket; +} + +void WebRtcTransportImp::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) { + WebRtcTransport::OnIceServerSelectedTuple(iceServer, tuple); + _socket->setSendPeerAddr(tuple); +} diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index c42317ab..1cd63583 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -110,7 +110,7 @@ private: class RtpReceiverImp; -class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ +class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; ~WebRtcTransportImp() override; @@ -139,6 +139,20 @@ protected: void onRtcp(const char *buf, size_t len) override; void onShutdown(const SockException &ex) override; + void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override; + + ///////MediaSourceEvent override/////// + // 关闭 + bool close(MediaSource &sender, bool force) override; + // 播放总人数 + int totalReaderCount(MediaSource &sender) override; + // 获取媒体源类型 + MediaOriginType getOriginType(MediaSource &sender) const override; + // 获取媒体源url或者文件路径 + string getOriginUrl(MediaSource &sender) const override; + // 获取媒体源客户端相关信息 + std::shared_ptr getOriginSock(MediaSource &sender) const override; + private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onCreate() override; From a22a6bafb76a48774d31c062f04644549134c8d6 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 7 Apr 2021 18:35:38 +0800 Subject: [PATCH 080/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E6=B5=81=E9=87=8F?= =?UTF-8?q?=E7=BB=9F=E8=AE=A1=E4=BA=8B=E4=BB=B6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 4 ++-- webrtc/WebRtcTransport.cpp | 32 +++++++++++++++++++++++++++++++- webrtc/WebRtcTransport.h | 6 +++++- 3 files changed, 38 insertions(+), 4 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 5fadf3bb..7776958f 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1104,7 +1104,7 @@ void installWebApi() { throw runtime_error(StrPrinter << "播放鉴权失败:" << err); } auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); - rtc->attach(src, true); + rtc->attach(src, info, true); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; invoker(200, headerOut, val.toStyledString()); @@ -1138,7 +1138,7 @@ void installWebApi() { push_src->setProtocolTranslation(enableHls, enableMP4); auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller()); push_src->setListener(rtc); - rtc->attach(push_src, false); + rtc->attach(push_src, info, false); val["sdp"] = rtc->getAnswerSdp(offer_sdp); val["type"] = "answer"; invoker(200, headerOut, val.toStyledString()); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9930d8f4..a229171d 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -285,10 +285,37 @@ WebRtcTransportImp::~WebRtcTransportImp() { void WebRtcTransportImp::onDestory() { WebRtcTransport::onDestory(); + uint64_t duration = _alive_ticker.createdTime() / 1000; + + //流量统计事件广播 + GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold); + + if (_play_src) { + WarnP(_socket) << "RTC播放器(" + << _media_info._vhost << "/" + << _media_info._app << "/" + << _media_info._streamid + << ")结束播放,耗时(s):" << duration; + if (_bytes_usage >= iFlowThreshold * 1024) { + NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast(*_socket)); + } + } + + if (_push_src) { + WarnP(_socket) << "RTC推流器(" + << _media_info._vhost << "/" + << _media_info._app << "/" + << _media_info._streamid + << ")结束推流,耗时(s):" << duration; + if (_bytes_usage >= iFlowThreshold * 1024) { + NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast(*_socket)); + } + } } -void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, bool is_play) { +void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, const MediaInfo &info, bool is_play) { assert(src); + _media_info = info; if (is_play) { _play_src = src; } else { @@ -455,6 +482,7 @@ private: }; void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { + _bytes_usage += len; auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len); for (auto rtcp : rtcps) { switch ((RtcpType) rtcp->pt) { @@ -504,6 +532,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + _bytes_usage += len; _alive_ticker.resetTime(); RtpHeader *rtp = (RtpHeader *) buf; //根据接收到的rtp的pt信息，找到该流的信息 @@ -549,6 +578,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //忽略，对方不支持该编码类型 return; } + _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); //统计rtp发送情况，好做sr汇报 _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 1cd63583..6c2ea591 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -127,7 +127,7 @@ public: * @param src 媒体源 * @param is_play 是播放还是推流 */ - void attach(const RtspMediaSource::Ptr &src, bool is_play = true); + void attach(const RtspMediaSource::Ptr &src, const MediaInfo &info, bool is_play = true); protected: void onStartWebRTC() override; @@ -176,6 +176,10 @@ private: void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: + //用掉的总流量 + uint64_t _bytes_usage = 0; + //媒体相关元数据 + MediaInfo _media_info; //保持自我强引用 Ptr _self; //检测超时的定时器 From 87e1209d0aa767ef658507a5e06a04280a2f6fd4 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Wed, 7 Apr 2021 20:33:53 +0800 Subject: [PATCH 081/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E6=9F=90=E4=BA=9B?= =?UTF-8?q?=E5=B9=B3=E5=8F=B0=E4=B8=8B=EF=BC=8Cudp=20connect=E5=90=8Esendt?= =?UTF-8?q?o=E6=97=B6=E6=8C=87=E5=AE=9A=E5=AF=B9=E7=AB=AF=E5=9C=B0?= =?UTF-8?q?=E5=9D=80=E5=8F=91=E9=80=81=E5=A4=B1=E8=B4=A5=E7=9A=84=E9=97=AE?= =?UTF-8?q?=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 29 +++++++++++++++++++++++++---- webrtc/WebRtcTransport.h | 17 ++++++++++++++--- 2 files changed, 39 insertions(+), 7 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index a229171d..9f1a615f 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -120,6 +120,10 @@ const RtcSession& WebRtcTransport::getSdp(SdpType type) const{ } } +RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{ + return _ice_server->GetSelectedTuple(); +} + string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str); for (auto &finger_prints : transport->GetLocalFingerprints()) { @@ -614,10 +618,27 @@ string WebRtcTransportImp::getOriginUrl(MediaSource &sender) const { } std::shared_ptr WebRtcTransportImp::getOriginSock(MediaSource &sender) const { - return const_cast(this)->_socket; + return const_cast(this)->shared_from_this(); } -void WebRtcTransportImp::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) { - WebRtcTransport::OnIceServerSelectedTuple(iceServer, tuple); - _socket->setSendPeerAddr(tuple); +///////////////////////////////////////////////////////////////////////////////////////////// + +string WebRtcTransportImp::get_local_ip() { + return getSdp(SdpType::answer).media[0].candidate[0].address; } + +uint16_t WebRtcTransportImp::get_local_port() { + return _socket->get_local_port(); +} + +string WebRtcTransportImp::get_peer_ip() { + return SockUtil::inet_ntoa(((struct sockaddr_in *) getSelectedTuple())->sin_addr); +} + +uint16_t WebRtcTransportImp::get_peer_port() { + return ntohs(((struct sockaddr_in *) getSelectedTuple())->sin_port); +} + +string WebRtcTransportImp::getIdentifier() const { + return StrPrinter << this; +} \ No newline at end of file diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 6c2ea591..e418cdf8 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -93,6 +93,7 @@ protected: protected: const RtcSession& getSdp(SdpType type) const; + RTC::TransportTuple* getSelectedTuple() const; private: void onSendSockData(const char *buf, size_t len, bool flush = true); @@ -110,7 +111,7 @@ private: class RtpReceiverImp; -class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public std::enable_shared_from_this{ +class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; ~WebRtcTransportImp() override; @@ -139,8 +140,6 @@ protected: void onRtcp(const char *buf, size_t len) override; void onShutdown(const SockException &ex) override; - void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override; - ///////MediaSourceEvent override/////// // 关闭 bool close(MediaSource &sender, bool force) override; @@ -153,6 +152,18 @@ protected: // 获取媒体源客户端相关信息 std::shared_ptr getOriginSock(MediaSource &sender) const override; + ///////SockInfo override/////// + //获取本机ip + string get_local_ip() override; + //获取本机端口号 + uint16_t get_local_port() override; + //获取对方ip + string get_peer_ip() override; + //获取对方端口号 + uint16_t get_peer_port() override; + //获取标识符 + string getIdentifier() const override; + private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onCreate() override; From fb9eefca6968168c81574aeb0de900a7148b0f60 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Wed, 7 Apr 2021 20:40:42 +0800 Subject: [PATCH 082/218] =?UTF-8?q?=E5=AE=8C=E5=96=84=E6=97=A5=E5=BF=97?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9f1a615f..0126590c 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -295,22 +295,22 @@ void WebRtcTransportImp::onDestory() { GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold); if (_play_src) { - WarnP(_socket) << "RTC播放器(" - << _media_info._vhost << "/" - << _media_info._app << "/" - << _media_info._streamid - << ")结束播放,耗时(s):" << duration; + WarnL << "RTC播放器(" + << _media_info._vhost << "/" + << _media_info._app << "/" + << _media_info._streamid + << ")结束播放,耗时(s):" << duration; if (_bytes_usage >= iFlowThreshold * 1024) { NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast(*_socket)); } } if (_push_src) { - WarnP(_socket) << "RTC推流器(" - << _media_info._vhost << "/" - << _media_info._app << "/" - << _media_info._streamid - << ")结束推流,耗时(s):" << duration; + WarnL << "RTC推流器(" + << _media_info._vhost << "/" + << _media_info._app << "/" + << _media_info._streamid + << ")结束推流,耗时(s):" << duration; if (_bytes_usage >= iFlowThreshold * 1024) { NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast(*_socket)); } From 1650f63af533a92a74f5d5fcd13655a284d1bb95 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Wed, 7 Apr 2021 20:41:17 +0800 Subject: [PATCH 083/218] =?UTF-8?q?=E7=94=9F=E6=88=90answer=20sdp=E5=A4=B1?= =?UTF-8?q?=E8=B4=A5=E6=97=B6=EF=BC=8C=E7=AB=8B=E5=8D=B3=E9=94=80=E6=AF=81?= =?UTF-8?q?=E5=AF=B9=E8=B1=A1?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 38 ++++++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 16 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 0126590c..7d6548b6 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -154,24 +154,30 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ } std::string WebRtcTransport::getAnswerSdp(const string &offer){ - //// 解析offer sdp //// - _offer_sdp = std::make_shared(); - _offer_sdp->loadFrom(offer); - onCheckSdp(SdpType::offer, *_offer_sdp); - setRemoteDtlsFingerprint(*_offer_sdp); + try { + //// 解析offer sdp //// + _offer_sdp = std::make_shared(); + _offer_sdp->loadFrom(offer); + onCheckSdp(SdpType::offer, *_offer_sdp); + setRemoteDtlsFingerprint(*_offer_sdp); - //// sdp 配置 //// - SdpAttrFingerprint fingerprint; - fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; - fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport); - RtcConfigure configure; - configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint); - onRtcConfigure(configure); + //// sdp 配置 //// + SdpAttrFingerprint fingerprint; + fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm; + fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport); + RtcConfigure configure; + configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), + RtpDirection::sendrecv, fingerprint); + onRtcConfigure(configure); - //// 生成answer sdp //// - _answer_sdp = configure.createAnswer(*_offer_sdp); - onCheckSdp(SdpType::answer, *_answer_sdp); - return _answer_sdp->toString(); + //// 生成answer sdp //// + _answer_sdp = configure.createAnswer(*_offer_sdp); + onCheckSdp(SdpType::answer, *_answer_sdp); + return _answer_sdp->toString(); + } catch (exception &ex) { + onShutdown(SockException(Err_shutdown, ex.what())); + throw; + } } bool is_dtls(char *buf) { From b2afae0e8a41fa91ff452c11885621656a374b11 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Wed, 7 Apr 2021 21:02:28 +0800 Subject: [PATCH 084/218] =?UTF-8?q?=E6=9A=82=E6=97=B6=E5=B1=8F=E8=94=BD?= =?UTF-8?q?=E4=B8=80=E4=BA=9B=E6=97=A5=E5=BF=97?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 5 +++++ webrtc/WebRtcTransport.cpp | 5 ++--- 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 86af2821..912999b9 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -133,6 +133,11 @@ void RtcpHeader::net2Host(size_t len){ bye->net2Host(len); break; } + + case RtcpType::RTCP_RTPFB: { + //todo 支持rtcp-fb相关功能 + break; + } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); } } diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 7d6548b6..4084c610 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -533,7 +533,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { break; } case RtcpType::RTCP_PSFB: { -// InfoL << rtcp->dumpString(); + //todo 支持pli等更多类型的rtcp break; } default: break; @@ -561,11 +561,10 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if(!info.is_common_rtp){ //todo rtx/red/ulpfec类型的rtp先未处理 - WarnL; return; } if (_pli_ticker.elapsedTime() > 2000) { - //todo 定期发送pli + //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); auto pli = RtcpPli::create(); pli->ssrc = htonl(0); From 7378650620112bf2cd3d44f2693fdc46297cb57e Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 9 Apr 2021 12:15:32 +0800 Subject: [PATCH 085/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dwindows=E4=B8=8B?= =?UTF-8?q?=E8=8E=B7=E5=8F=96ssrc=E5=A4=B1=E8=B4=A5=E7=9A=84bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 490890b0..410f6c53 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -659,16 +659,16 @@ void SdpAttrSSRCGroup::parse(const string &str) { SDP_THROW(); } type = std::move(vec[0]); - u.fid.rtp_ssrc = atoi(vec[1].data()); - u.fid.rtx_ssrc = atoi(vec[2].data()); + u.fid.rtp_ssrc = atoll(vec[1].data()) & 0xFFFFFFFF; + u.fid.rtx_ssrc = atoll(vec[2].data()) & 0xFFFFFFFF; } else if (vec.size() == 4) { if (vec[0] != "SIM") { SDP_THROW(); } type = std::move(vec[0]); - u.sim.rtp_ssrc_low = atoi(vec[1].data()); - u.sim.rtp_ssrc_mid = atoi(vec[2].data()); - u.sim.rtp_ssrc_high = atoi(vec[3].data()); + u.sim.rtp_ssrc_low = atoll(vec[1].data()) & 0xFFFFFFFF; + u.sim.rtp_ssrc_mid = atoll(vec[2].data()) & 0xFFFFFFFF; + u.sim.rtp_ssrc_high = atoll(vec[3].data()) & 0xFFFFFFFF; } else { SDP_THROW(); } From b180ea060900f463c23077cba55d2605b9e3c526 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 9 Apr 2021 20:42:05 +0800 Subject: [PATCH 086/218] =?UTF-8?q?=E5=88=A0=E9=99=A4=E6=B5=8B=E8=AF=95?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/main.cpp | 148 ------------------------------------------------ 1 file changed, 148 deletions(-) diff --git a/server/main.cpp b/server/main.cpp index fdba471d..4bbf189f 100644 --- a/server/main.cpp +++ b/server/main.cpp @@ -210,153 +210,6 @@ static void inline listen_shell_input(){ //全局变量，在WebApi中用于保存配置文件用 string g_ini_file; - -void test_sdp(){ - char str1[] = "v=0\n" - "o=- 380154348540553537 2 IN IP4 127.0.0.1\n" - "s=-\n" - "b=CT:1900\n" - "t=0 0\n" - "a=group:BUNDLE video\n" - "a=msid-semantic: WMS\n" - "m=video 9 RTP/SAVPF 96\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:1ZFN\n" - "a=ice-pwd:70P3H0jPlGz1fiJl5XZfXMZH\n" - "a=ice-options:trickle\n" - "a=fingerprint:sha-256 3E:10:35:6B:9A:9E:B0:55:AC:2A:88:F5:74:C1:70:32:B5:8D:88:1D:37:B0:9C:69:A6:DD:07:10:73:27:1A:16\n" - "a=setup:active\n" - "a=mid:video\n" - "a=recvonly\n" - "a=rtcp-mux\n" - "a=rtpmap:96 H264/90000\n" - "a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f"; - char str2[] = "v=0\n" - "o=- 2584450093346841581 2 IN IP4 127.0.0.1\n" - "s=-\n" - "t=0 0\n" - "a=group:BUNDLE audio video data\n" - "a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:audio\n" - "a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level\n" - "a=sendrecv\n" - "a=rtcp-mux\n" - "a=rtpmap:111 opus/48000/2\n" - "a=rtcp-fb:111 transport-cc\n" - "a=fmtp:111 minptime=10;useinbandfec=1\n" - "a=rtpmap:103 ISAC/16000\n" - "a=rtpmap:104 ISAC/32000\n" - "a=rtpmap:9 G722/8000\n" - "a=rtpmap:0 PCMU/8000\n" - "a=rtpmap:8 PCMA/8000\n" - "a=rtpmap:106 CN/32000\n" - "a=rtpmap:105 CN/16000\n" - "a=rtpmap:13 CN/8000\n" - "a=rtpmap:110 telephone-event/48000\n" - "a=rtpmap:112 telephone-event/32000\n" - "a=rtpmap:113 telephone-event/16000\n" - "a=rtpmap:126 telephone-event/8000\n" - "a=ssrc:120276603 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363\n" - "a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363\n" - "m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125\n" - "c=IN IP4 0.0.0.0\n" - "a=rtcp:9 IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:video\n" - "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n" - "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\n" - "a=extmap:4 urn:3gpp:video-orientation\n" - "a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\n" - "a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\n" - "a=sendrecv\n" - "a=rtcp-mux\n" - "a=rtcp-rsize\n" - "a=rtpmap:96 VP8/90000\n" - "a=rtcp-fb:96 ccm fir\n" - "a=rtcp-fb:96 nack\n" - "a=rtcp-fb:96 nack pli\n" - "a=rtcp-fb:96 goog-remb\n" - "a=rtcp-fb:96 transport-cc\n" - "a=rtpmap:98 VP9/90000\n" - "a=rtcp-fb:98 ccm fir\n" - "a=rtcp-fb:98 nack\n" - "a=rtcp-fb:98 nack pli\n" - "a=rtcp-fb:98 goog-remb\n" - "a=rtcp-fb:98 transport-cc\n" - "a=rtpmap:100 H264/90000\n" - "a=rtcp-fb:100 ccm fir\n" - "a=rtcp-fb:100 nack\n" - "a=rtcp-fb:100 nack pli\n" - "a=rtcp-fb:100 goog-remb\n" - "a=rtcp-fb:100 transport-cc\n" - "a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\n" - "a=rtpmap:102 red/90000\n" - "a=rtpmap:127 ulpfec/90000\n" - "a=rtpmap:97 rtx/90000\n" - "a=fmtp:97 apt=96\n" - "a=rtpmap:99 rtx/90000\n" - "a=fmtp:99 apt=98\n" - "a=rtpmap:101 rtx/90000\n" - "a=fmtp:101 apt=100\n" - "a=rtpmap:125 rtx/90000\n" - "a=fmtp:125 apt=102\n" - "a=ssrc-group:FID 2580761338 611523443\n" - "a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:611523443 cname:iSkJ2vn5cYYubTve\n" - "a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n" - "a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n" - "a=candidate:3575467457 1 udp 2113937151 10.15.83.23 57857 typ host generation 0 ufrag 6R0z network-cost 999\n" - "m=application 9 DTLS/SCTP 5000\n" - "c=IN IP4 0.0.0.0\n" - "a=ice-ufrag:sXJ3\n" - "a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n" - "a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n" - "a=setup:actpass\n" - "a=mid:data\n" - "a=sctpmap:5000 webrtc-datachannel 1024\n" - "a=sctp-port:5000"; - - RtcSessionSdp sdp1; - sdp1.parse(str1); - - RtcSessionSdp sdp2; - sdp2.parse(str2); - - for (auto media : sdp1.medias) { - InfoL << getRtpDirectionString(media.getDirection()); - } - for (auto media : sdp2.medias) { - InfoL << getRtpDirectionString(media.getDirection()); - } - InfoL << sdp1.toString(); - InfoL << sdp2.toString(); - - RtcSession session1; - session1.loadFrom(str1); - - RtcSession session2; - session2.loadFrom(str2); - DebugL << session1.toString(); - DebugL << session2.toString(); -} - int start_main(int argc,char *argv[]) { { CMD_main cmd_main; @@ -414,7 +267,6 @@ int start_main(int argc,char *argv[]) { }); } -// test_sdp(); uint16_t shellPort = mINI::Instance()[Shell::kPort]; uint16_t rtspPort = mINI::Instance()[Rtsp::kPort]; uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort]; From 5b7974a522abceba6bc7760ea33de3b94ef04434 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Fri, 9 Apr 2021 20:42:36 +0800 Subject: [PATCH 087/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E7=89=88=E6=9D=83?= =?UTF-8?q?=E4=BF=A1=E6=81=AF?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- AUTHORS | 3 ++- webrtc/DtlsTransport.cpp | 20 +++++++++++++++++++- webrtc/DtlsTransport.hpp | 20 +++++++++++++++++++- webrtc/IceServer.cpp | 20 +++++++++++++++++++- webrtc/IceServer.hpp | 20 +++++++++++++++++++- webrtc/Sdp.cpp | 12 +++++++++--- webrtc/Sdp.h | 12 +++++++++--- webrtc/SrtpSession.cpp | 20 +++++++++++++++++++- webrtc/SrtpSession.hpp | 20 +++++++++++++++++++- webrtc/StunPacket.cpp | 20 +++++++++++++++++++- webrtc/StunPacket.hpp | 20 +++++++++++++++++++- webrtc/Utils.hpp | 20 +++++++++++++++++++- webrtc/WebRtcTransport.cpp | 12 +++++++++++- webrtc/WebRtcTransport.h | 12 +++++++++++- webrtc/readme.md | 37 +++++++++++++++++++++++++++++++++++++ www/webrtc/readme.txt | 2 ++ 16 files changed, 252 insertions(+), 18 deletions(-) create mode 100644 webrtc/readme.md create mode 100644 www/webrtc/readme.txt diff --git a/AUTHORS b/AUTHORS index 2fbea1c9..3456f103 100644 --- a/AUTHORS +++ b/AUTHORS @@ -31,4 +31,5 @@ Luke 大裤衩 <3503207480@qq.com> droid.chow [陈晓林](https://github.com/musicwood) -[老衲不出家](https://github.com/monktan89) \ No newline at end of file +[老衲不出家](https://github.com/monktan89) +[big panda](<2381267071@qq.com>) \ No newline at end of file diff --git a/webrtc/DtlsTransport.cpp b/webrtc/DtlsTransport.cpp index 0dd9492c..8abb2a21 100644 --- a/webrtc/DtlsTransport.cpp +++ b/webrtc/DtlsTransport.cpp @@ -1,4 +1,22 @@ -#define MS_CLASS "RTC::DtlsTransport" +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#define MS_CLASS "RTC::DtlsTransport" // #define MS_LOG_DEV_LEVEL 3 #include "DtlsTransport.hpp" diff --git a/webrtc/DtlsTransport.hpp b/webrtc/DtlsTransport.hpp index 2d324441..fb28a6a4 100644 --- a/webrtc/DtlsTransport.hpp +++ b/webrtc/DtlsTransport.hpp @@ -1,4 +1,22 @@ -#ifndef MS_RTC_DTLS_TRANSPORT_HPP +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#ifndef MS_RTC_DTLS_TRANSPORT_HPP #define MS_RTC_DTLS_TRANSPORT_HPP #include "SrtpSession.hpp" diff --git a/webrtc/IceServer.cpp b/webrtc/IceServer.cpp index 15aeb0bd..c976bcf0 100644 --- a/webrtc/IceServer.cpp +++ b/webrtc/IceServer.cpp @@ -1,4 +1,22 @@ -#define MS_CLASS "RTC::IceServer" +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#define MS_CLASS "RTC::IceServer" // #define MS_LOG_DEV_LEVEL 3 #include diff --git a/webrtc/IceServer.hpp b/webrtc/IceServer.hpp index 49045f43..8b9742ad 100644 --- a/webrtc/IceServer.hpp +++ b/webrtc/IceServer.hpp @@ -1,4 +1,22 @@ -#ifndef MS_RTC_ICE_SERVER_HPP +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#ifndef MS_RTC_ICE_SERVER_HPP #define MS_RTC_ICE_SERVER_HPP #include "StunPacket.hpp" diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 410f6c53..89065ecc 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1,6 +1,12 @@ -// -// Created by xzl on 2021/3/27. -// +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ #include "Sdp.h" #include "Common/Parser.h" diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 2a9eb6ee..9247018a 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -1,6 +1,12 @@ -// -// Created by xzl on 2021/3/27. -// +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ #ifndef ZLMEDIAKIT_SDP_H #define ZLMEDIAKIT_SDP_H diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index a73c5aaa..cf359b7e 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -1,4 +1,22 @@ -#define MS_CLASS "RTC::SrtpSession" +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#define MS_CLASS "RTC::SrtpSession" // #define MS_LOG_DEV_LEVEL 3 #include "SrtpSession.hpp" diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index 988d730b..d9d73fbe 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -1,4 +1,22 @@ -#ifndef MS_RTC_SRTP_SESSION_HPP +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#ifndef MS_RTC_SRTP_SESSION_HPP #define MS_RTC_SRTP_SESSION_HPP #include "Utils.hpp" diff --git a/webrtc/StunPacket.cpp b/webrtc/StunPacket.cpp index 223d3f57..50120a70 100644 --- a/webrtc/StunPacket.cpp +++ b/webrtc/StunPacket.cpp @@ -1,4 +1,22 @@ -#define MS_CLASS "RTC::StunPacket" +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#define MS_CLASS "RTC::StunPacket" // #define MS_LOG_DEV_LEVEL 3 #include "StunPacket.hpp" diff --git a/webrtc/StunPacket.hpp b/webrtc/StunPacket.hpp index ebd1b1d0..a6b2c940 100644 --- a/webrtc/StunPacket.hpp +++ b/webrtc/StunPacket.hpp @@ -1,4 +1,22 @@ -#ifndef MS_RTC_STUN_PACKET_HPP +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#ifndef MS_RTC_STUN_PACKET_HPP #define MS_RTC_STUN_PACKET_HPP diff --git a/webrtc/Utils.hpp b/webrtc/Utils.hpp index 0b1b66ad..1692ff50 100644 --- a/webrtc/Utils.hpp +++ b/webrtc/Utils.hpp @@ -1,4 +1,22 @@ -#ifndef MS_UTILS_HPP +/** +ISC License + +Copyright © 2015, Iñaki Baz Castillo + +Permission to use, copy, modify, and/or distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. + */ + +#ifndef MS_UTILS_HPP #define MS_UTILS_HPP #if defined(_WIN32) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 4084c610..342dabc0 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -1,4 +1,14 @@ -#include "WebRtcTransport.h" +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#include "WebRtcTransport.h" #include #include "Rtcp/Rtcp.h" #include "Rtsp/RtpReceiver.h" diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index e418cdf8..2b8f8a2c 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -1,4 +1,14 @@ -#pragma once +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#pragma once #include #include diff --git a/webrtc/readme.md b/webrtc/readme.md new file mode 100644 index 00000000..ff832278 --- /dev/null +++ b/webrtc/readme.md @@ -0,0 +1,37 @@ +# 致谢与声明 +本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为： + +- ice相关功能： + - IceServer.cpp + - IceServer.hpp + - StunPacket.cpp + - StunPacket.hpp + - Utils.hpp + +- dtls相关功能： + - DtlsTransport.cpp + - DtlsTransport.hpp + +- srtp相关功能： + - SrtpSession.cpp + - SrtpSession.hpp + + +以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者， +用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。 + +同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者， +在集成MediaSoup相关代码前期，主要参考这个项目。 + +另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)， +其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js + +# 现状与规划 +ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有： + +- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。 +- 2、实现rtp重传等相关功能。 +- 3、实现simulecast相关功能。 +- 4、fec、rtp扩展等其他功能。 +- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。 + diff --git a/www/webrtc/readme.txt b/www/webrtc/readme.txt new file mode 100644 index 00000000..ba651b40 --- /dev/null +++ b/www/webrtc/readme.txt @@ -0,0 +1,2 @@ +感谢[big panda](<2381267071@qq.com>) 开发并贡献此webrtc js测试客户端, +其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js \ No newline at end of file From e495dfca64027e25dbb81746dc1ff7d1867453fe Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 10 Apr 2021 21:58:27 +0800 Subject: [PATCH 088/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E5=BF=BD=E7=95=A5g71?= =?UTF-8?q?1a(pt=3D0)=E7=9A=84bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 2 +- webrtc/WebRtcTransport.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 342dabc0..9a58fb21 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -593,7 +593,7 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ auto &pt = _send_rtp_pt[rtp->type]; - if (!pt) { + if (pt == 0xFF) { //忽略，对方不支持该编码类型 return; } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 2b8f8a2c..50f45604 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -212,7 +212,7 @@ private: //rtc rtp推流的视频ssrc uint32_t _recv_video_ssrc; //记录协商的rtp的pt类型 - uint8_t _send_rtp_pt[2] = {0, 0}; + uint8_t _send_rtp_pt[2] = {0xFF, 0xFF}; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 From ebf650cf4fc018948fd1619159ced5932b827ba9 Mon Sep 17 00:00:00 2001 From: xgj Date: Sat, 10 Apr 2021 22:24:26 +0800 Subject: [PATCH 089/218] fix static library link order --- server/CMakeLists.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/server/CMakeLists.txt b/server/CMakeLists.txt index 6e0af5e1..bf59de98 100644 --- a/server/CMakeLists.txt +++ b/server/CMakeLists.txt @@ -48,4 +48,4 @@ else() install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE}) endif() -target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) +target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group) From 33da2e28ec3a6e4421602cf6c4c326037a57a975 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 01:14:41 +0800 Subject: [PATCH 090/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dpt=E8=A6=86=E7=9B=96?= =?UTF-8?q?=E5=A4=B1=E8=B4=A5=E7=9A=84bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/SrtpSession.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index cf359b7e..4591e466 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -244,7 +244,7 @@ namespace RTC } std::memcpy(EncryptBuffer, *data, *len); - EncryptBuffer[1] |= (pt & 0x7F); + EncryptBuffer[1] = (pt & 0x7F) | (EncryptBuffer[1] & 0x80); srtp_err_status_t err = srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); From 012f58d27c07446d6f6464d32e98cbeea62730ba Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 01:15:02 +0800 Subject: [PATCH 091/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtp=20=E8=B0=83?= =?UTF-8?q?=E8=AF=95=E6=8E=A5=E5=8F=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtsp/Rtsp.cpp | 4 ++++ src/Rtsp/Rtsp.h | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/src/Rtsp/Rtsp.cpp b/src/Rtsp/Rtsp.cpp index 492eb644..73f84434 100644 --- a/src/Rtsp/Rtsp.cpp +++ b/src/Rtsp/Rtsp.cpp @@ -521,6 +521,10 @@ RtpHeader* RtpPacket::getHeader(){ return (RtpHeader*)(data() + RtpPacket::kRtpTcpHeaderSize); } +string RtpPacket::dumpString() const{ + return ((RtpPacket *) this)->getHeader()->dumpString(size() - RtpPacket::kRtpTcpHeaderSize); +} + uint16_t RtpPacket::getSeq(){ return ntohs(getHeader()->seq); } diff --git a/src/Rtsp/Rtsp.h b/src/Rtsp/Rtsp.h index 884176e1..20307f45 100644 --- a/src/Rtsp/Rtsp.h +++ b/src/Rtsp/Rtsp.h @@ -150,7 +150,11 @@ public: kRtpTcpHeaderSize = 4 }; + //获取rtp头 RtpHeader* getHeader(); + //打印调试信息 + string dumpString() const; + //主机字节序的seq uint16_t getSeq(); //主机字节序的时间戳，已经转换为毫秒 From 5fadd2665df9921a1921b5af8a2f1508a8a89f11 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 11:44:16 +0800 Subject: [PATCH 092/218] =?UTF-8?q?=E9=9D=9Ewindows=E5=B9=B3=E5=8F=B0?= =?UTF-8?q?=EF=BC=8C=E6=89=8D=E5=BF=BD=E7=95=A5=E9=9D=99=E6=80=81=E5=BA=93?= =?UTF-8?q?=E9=93=BE=E6=8E=A5=E9=A1=BA=E5=BA=8F?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/CMakeLists.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/server/CMakeLists.txt b/server/CMakeLists.txt index bf59de98..2e11813c 100644 --- a/server/CMakeLists.txt +++ b/server/CMakeLists.txt @@ -48,4 +48,8 @@ else() install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE}) endif() -target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group) +if (WIN32) + target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) +else () + target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group) +endif () From 9e93561c7741032d3156f1af59ef0c0fd9dc7ebb Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 11:53:05 +0800 Subject: [PATCH 093/218] =?UTF-8?q?=E5=8F=AA=E6=9C=89linux=E5=B9=B3?= =?UTF-8?q?=E5=8F=B0=E6=89=93=E5=BC=80=E9=93=BE=E6=8E=A5=E9=A1=BA=E5=BA=8F?= =?UTF-8?q?=E6=97=A0=E5=85=B3=E9=80=89=E9=A1=B9?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/CMakeLists.txt | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/server/CMakeLists.txt b/server/CMakeLists.txt index 2e11813c..8ae6ff28 100644 --- a/server/CMakeLists.txt +++ b/server/CMakeLists.txt @@ -48,8 +48,8 @@ else() install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE}) endif() -if (WIN32) - target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) -else () +if (CMAKE_SYSTEM_NAME MATCHES "Linux") target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group) +else () + target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) endif () From 8b7289a689b530a0000b0dd0b9c949a2ed5cb8f9 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 12:06:40 +0800 Subject: [PATCH 094/218] =?UTF-8?q?rtsp=E6=8E=A8=E6=B5=81=E6=94=AF?= =?UTF-8?q?=E6=8C=81=E5=85=B3=E9=97=AD=E7=9B=B4=E6=8E=A5=E4=BB=A3=E7=90=86?= =?UTF-8?q?=E6=A8=A1=E5=BC=8F?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtsp/RtspMediaSourceImp.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/src/Rtsp/RtspMediaSourceImp.h b/src/Rtsp/RtspMediaSourceImp.h index 4ecc93b7..8401d4b8 100644 --- a/src/Rtsp/RtspMediaSourceImp.h +++ b/src/Rtsp/RtspMediaSourceImp.h @@ -56,7 +56,11 @@ public: //需要解复用rtp key_pos = _demuxer->inputRtp(rtp); } - RtspMediaSource::onWrite(std::move(rtp), key_pos); + GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy); + if (directProxy) { + //直接代理模式才直接使用原始rtp + RtspMediaSource::onWrite(std::move(rtp), key_pos); + } } /** @@ -72,8 +76,10 @@ public: * @param enableMP4 是否mp4录制 */ void setProtocolTranslation(bool enableHls,bool enableMP4){ - //不重复生成rtsp - _muxer = std::make_shared(getVhost(), getApp(), getId(), _demuxer->getDuration(), false, true, enableHls, enableMP4); + GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy); + //开启直接代理模式时，rtsp直接代理，不重复产生；但是有些rtsp推流端，由于sdp中已有sps pps，rtp中就不再包括sps pps, + //导致rtc无法播放，所以在rtsp推流rtc播放时，建议关闭直接代理模式 + _muxer = std::make_shared(getVhost(), getApp(), getId(), _demuxer->getDuration(), !directProxy, true, enableHls, enableMP4); _muxer->setMediaListener(getListener()); _muxer->setTrackListener(static_pointer_cast(shared_from_this())); //让_muxer对象拦截一部分事件(比如说录像相关事件) From fbe8beb3648b4cb1f91d0837c5534ab4fc7a3863 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 12:06:55 +0800 Subject: [PATCH 095/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtc=E7=9B=B8=E5=85=B3?= =?UTF-8?q?=E9=85=8D=E7=BD=AE=E4=B8=8E=E6=8F=8F=E8=BF=B0?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/conf/config.ini b/conf/config.ini index 904cc57a..c7330ed2 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -206,13 +206,21 @@ port=10000 #rtp超时时间，单位秒 timeoutSec=15 +[rtc] +#rtc播放推流、播放超时时间 +timeoutSec=15 +#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip +externIP= + [rtsp] #rtsp专有鉴权方式是采用base64还是md5方式 authBasic=0 -#rtsp拉流代理是否是直接代理模式 +#rtsp拉流、推流代理是否是直接代理模式 #直接代理后支持任意编码格式，但是会导致GOP缓存无法定位到I帧，可能会导致开播花屏 #并且如果是tcp方式拉流，如果rtp大于mtu会导致无法使用udp方式代理 #假定您的拉流源地址不是264或265或AAC，那么你可以使用直接代理的方式来支持rtsp代理 +#如果你是rtsp推拉流，但是rtc播放，也建议关闭直接代理模式， +#因为直接代理时，rtp中可能没有sps pps,会导致rtc无法播放 #默认开启rtsp直接代理，rtmp由于没有这些问题，是强制开启直接代理的 directProxy=1 #rtsp必须在此时间内完成握手，否则服务器会断开链接，单位秒 From 753fe05813b114ac99b46cbb614a1e9121ca001a Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 11 Apr 2021 12:07:04 +0800 Subject: [PATCH 096/218] =?UTF-8?q?=E9=87=87=E7=94=A8pts?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/CommonRtp.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/Extension/CommonRtp.cpp b/src/Extension/CommonRtp.cpp index 039bf93d..e009816a 100644 --- a/src/Extension/CommonRtp.cpp +++ b/src/Extension/CommonRtp.cpp @@ -69,7 +69,7 @@ CommonRtpEncoder::CommonRtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_si } void CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){ - auto stamp = frame->dts(); + auto stamp = frame->pts(); auto ptr = frame->data() + frame->prefixSize(); auto len = frame->size() - frame->prefixSize(); auto remain_size = len; From f939e0a89405d395599e1d788c9e73ac213ef6f0 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 13 Apr 2021 18:50:20 +0800 Subject: [PATCH 097/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E4=B8=80=E4=BA=9Brtc?= =?UTF-8?q?p=20fci=E7=9B=B8=E5=85=B3=E5=AE=9A=E4=B9=89?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 10 +++ src/Rtcp/Rtcp.h | 57 ++++++++++++++++- src/Rtcp/RtcpFCI.cpp | 5 ++ src/Rtcp/RtcpFCI.h | 148 +++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 219 insertions(+), 1 deletion(-) create mode 100644 src/Rtcp/RtcpFCI.cpp create mode 100644 src/Rtcp/RtcpFCI.h diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 912999b9..aa958665 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -33,6 +33,15 @@ const char *sdesTypeToStr(SdesType type){ } } +const char *psfbTypeToStr(PSFBType type) { + switch (type){ +#define SWITCH_CASE(key, value) case PSFBType::key : return #value "(" #key ")"; + PSFB_TYPE_MAP(SWITCH_CASE) +#undef SWITCH_CASE + default: return "unknown payload-specific fb message (rfc4585) type"; + } +} + static size_t alignSize(size_t bytes) { return (size_t)((bytes + 3) / 4) << 2; } @@ -136,6 +145,7 @@ void RtcpHeader::net2Host(size_t len){ case RtcpType::RTCP_RTPFB: { //todo 支持rtcp-fb相关功能 + net2Host(); break; } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index c7a1556f..c257420f 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -55,6 +55,33 @@ namespace mediakit { XX(RTCP_SDES_NOTE, 7) \ XX(RTCP_SDES_PRIVATE, 8) +//https://datatracker.ietf.org/doc/rfc4585/?include_text=1 +//6.3. Payload-Specific Feedback Messages +// +// Payload-Specific FB messages are identified by the value PT=PSFB as +// RTCP message type. +// +// Three payload-specific FB messages are defined so far plus an +// application layer FB message. They are identified by means of the +// FMT parameter as follows: +// +// 0: unassigned +// 1: Picture Loss Indication (PLI) +// 2: Slice Loss Indication (SLI) +// 3: Reference Picture Selection Indication (RPSI) +// 4-14: unassigned +// 15: Application layer FB (AFB) message +// 16-30: unassigned +// 31: reserved for future expansion of the sequence number space + +#define PSFB_TYPE_MAP(XX) \ + XX(RTCP_PSFB_PLI, 1) \ + XX(RTCP_PSFB_SLI, 2) \ + XX(RTCP_PSFB_RPSI, 3) \ + XX(RTCP_PSFB_FIR, 4) \ + XX(RTCP_PSFB_TSTR, 5) \ + XX(RTCP_PSFB_AFB, 15) + //rtcp类型枚举 enum class RtcpType : uint8_t { #define XX(key, value) key = value, @@ -69,6 +96,13 @@ enum class SdesType : uint8_t { #undef XX }; +//psfb类型枚举 +enum class PSFBType : uint8_t { +#define XX(key, value) key = value, + PSFB_TYPE_MAP(XX) +#undef XX +}; + /** * RtcpType转描述字符串 */ @@ -79,6 +113,11 @@ const char *rtcpTypeToStr(RtcpType type); */ const char *sdesTypeToStr(SdesType type); +/** + * psfb枚举转描述字符串 + */ +const char *psfbTypeToStr(PSFBType type); + class RtcpHeader { public: #if __BYTE_ORDER == __BIG_ENDIAN @@ -462,7 +501,23 @@ private: void net2Host(size_t size); } PACKED; -//PLI +//6.1. Common Packet Format for Feedback Messages +// +// All FB messages MUST use a common packet format that is depicted in +// Figure 3: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| FMT | PT | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of media source | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : Feedback Control Information (FCI) : +// : : + class RtcpPli : public RtcpHeader { public: friend class RtcpHeader; diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp new file mode 100644 index 00000000..724f03b2 --- /dev/null +++ b/src/Rtcp/RtcpFCI.cpp @@ -0,0 +1,5 @@ +// +// Created by xzl on 2021/4/13. +// + +#include "RtcpFCI.h" diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h new file mode 100644 index 00000000..d55ef1ea --- /dev/null +++ b/src/Rtcp/RtcpFCI.h @@ -0,0 +1,148 @@ +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef ZLMEDIAKIT_RTCPFCI_H +#define ZLMEDIAKIT_RTCPFCI_H + +#include "Rtcp.h" + +//https://tools.ietf.org/html/rfc4585 + +namespace mediakit { + +//https://tools.ietf.org/html/rfc4585#section-6.2.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PID | BLP | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Figure 4: Syntax for the Generic NACK message + +class FCI_NACK { +public: + // The PID field is used to specify a lost packet. The PID field + // refers to the RTP sequence number of the lost packet. + uint16_t pid; + // bitmask of following lost packets (BLP): 16 bits + uint16_t blp; +} PACKED; + +//https://tools.ietf.org/html/rfc4585#section-6.3.2.2 +// The Slice Loss Indication uses one additional FCI field, the content +// of which is depicted in Figure 6. The length of the FB message MUST +// be set to 2+n, with n being the number of SLIs contained in the FCI +// field. +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | First | Number | PictureID | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Figure 6: Syntax of the Slice Loss Indication (SLI) +// +class FCI_SLI { +public: + uint32_t first : 13; + uint32_t number : 13; + uint32_t pic_id : 6; +} PACKED; + +//https://tools.ietf.org/html/rfc4585#section-6.3.3.2 +//6.3.3.2. Format +// +// The FCI for the RPSI message follows the format depicted in Figure 7: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PB |0| Payload Type| Native RPSI bit string | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | defined per codec ... | Padding (0) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Figure 7: Syntax of the Reference Picture Selection Indication (RPSI) +class FCI_RPSI { +public: + //The number of unused bits required to pad the length of the RPSI + // message to a multiple of 32 bits. + uint8_t pb; + +#if __BYTE_ORDER == __BIG_ENDIAN + //0: 1 bit + // MUST be set to zero upon transmission and ignored upon reception. + uint8_t zero : 1; + //Payload Type: 7 bits + // Indicates the RTP payload type in the context of which the native + // RPSI bit string MUST be interpreted. + uint8_t pt : 7; +#else + uint8_t pt: 7; + uint8_t zero: 1; +#endif + + // Native RPSI bit string: variable length + // The RPSI information as natively defined by the video codec. + char str[1]; + + //Padding: #PB bits + // A number of bits set to zero to fill up the contents of the RPSI + // message to the next 32-bit boundary. The number of padding bits + // MUST be indicated by the PB field. +} PACKED; + +//tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| FMT=15 | PT=206 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of media source | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Unique identifier 'R' 'E' 'M' 'B' | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Num SSRC | BR Exp | BR Mantissa | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC feedback | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ... | +class FCI_REMB { +public: + //Unique identifier 'R' 'E' 'M' 'B' + char magic[4]; + + //Number of SSRCs in this message. + uint32_t num_ssrc : 8; + + // BR Exp (6 bits): The exponential scaling of the mantissa for the + // maximum total media bit rate value, ignoring all packet + // overhead. The value is an unsigned integer [0..63], as + // in RFC 5104 section 4.2.2.1. + uint32_t br_exp : 6; + + //BR Mantissa (18 bits): The mantissa of the maximum total media bit + // rate (ignoring all packet overhead) that the sender of + // the REMB estimates. The BR is the estimate of the + // traveled path for the SSRCs reported in this message. + // The value is an unsigned integer in number of bits per + // second. + uint32_t br_mantissa : 18; + + // SSRC feedback (32 bits) Consists of one or more SSRC entries which + // this feedback message applies to. + uint32_t ssrc_feedback; + +} PACKED; + +} //namespace mediakit +#endif //ZLMEDIAKIT_RTCPFCI_H From 37096f8ed073091fb452f72168678fa42c9de6b0 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 13 Apr 2021 22:27:57 +0800 Subject: [PATCH 098/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtcp=E7=9A=84?= =?UTF-8?q?=E5=AE=9A=E4=B9=89?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 8 +- src/Rtcp/Rtcp.h | 51 +++++++++--- src/Rtcp/RtcpFCI.cpp | 18 ++++- src/Rtcp/RtcpFCI.h | 185 +++++++++++++++++++++++++++++++++++-------- 4 files changed, 210 insertions(+), 52 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index aa958665..afee33be 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -38,7 +38,7 @@ const char *psfbTypeToStr(PSFBType type) { #define SWITCH_CASE(key, value) case PSFBType::key : return #value "(" #key ")"; PSFB_TYPE_MAP(SWITCH_CASE) #undef SWITCH_CASE - default: return "unknown payload-specific fb message (rfc4585) type"; + default: return "unknown payload-specific fb message fmt type"; } } @@ -385,7 +385,7 @@ string SdesItem::dumpString() const{ printer << "ssrc:" << ssrc << "\r\n"; printer << "type:" << sdesTypeToStr((SdesType) type) << "\r\n"; printer << "length:" << (int) length << "\r\n"; - printer << "text:" << (length ? string(&text, length) : "") << "\r\n"; + printer << "text:" << (length ? string(text, length) : "") << "\r\n"; return std::move(printer); } @@ -403,7 +403,7 @@ std::shared_ptr RtcpSdes::create(const std::initializer_list & for (auto &text : item_text) { item_ptr->length = (0xFF & text.size()); //确保赋值\0为RTCP_SDES_END - memcpy(&(item_ptr->text), text.data(), item_ptr->length + 1); + memcpy(item_ptr->text, text.data(), item_ptr->length + 1); item_ptr = (SdesItem *) ((char *) item_ptr + item_ptr->totalBytes()); } @@ -555,7 +555,7 @@ void RtcpBye::net2Host(size_t size) { } } -#if 0 +#if 1 #include "Util/onceToken.h" static toolkit::onceToken token([](){ diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index c257420f..88fa5475 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -25,8 +25,7 @@ namespace mediakit { #pragma pack(push, 1) #endif // defined(_WIN32) -//https://datatracker.ietf.org/doc/rfc3550 - +//http://www.networksorcery.com/enp/protocol/rtcp.htm #define RTCP_PT_MAP(XX) \ XX(RTCP_FIR, 192) \ XX(RTCP_NACK, 193) \ @@ -43,7 +42,8 @@ namespace mediakit { XX(RTCP_AVB, 208) \ XX(RTCP_RSI, 209) \ XX(RTCP_TOKEN, 210) - + +//https://tools.ietf.org/html/rfc3550#section-6.5 #define SDES_TYPE_MAP(XX) \ XX(RTCP_SDES_END, 0) \ XX(RTCP_SDES_CNAME, 1) \ @@ -69,19 +69,49 @@ namespace mediakit { // 1: Picture Loss Indication (PLI) // 2: Slice Loss Indication (SLI) // 3: Reference Picture Selection Indication (RPSI) -// 4-14: unassigned -// 15: Application layer FB (AFB) message +// 4: FIR https://tools.ietf.org/html/rfc5104#section-4.3.1.1 +// 5: TSTR https://tools.ietf.org/html/rfc5104#section-4.3.2.1 +// 6: TSTN https://tools.ietf.org/html/rfc5104#section-4.3.2.1 +// 7: VBCM https://tools.ietf.org/html/rfc5104#section-4.3.4.1 +// 8-14: unassigned +// 15: REMB / Application layer FB (AFB) message, https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 // 16-30: unassigned // 31: reserved for future expansion of the sequence number space - #define PSFB_TYPE_MAP(XX) \ XX(RTCP_PSFB_PLI, 1) \ XX(RTCP_PSFB_SLI, 2) \ XX(RTCP_PSFB_RPSI, 3) \ XX(RTCP_PSFB_FIR, 4) \ - XX(RTCP_PSFB_TSTR, 5) \ + XX(RTCP_PSFB_TSTR, 5)\ + XX(RTCP_PSFB_TSTN, 6)\ + XX(RTCP_PSFB_VBCM, 7) \ XX(RTCP_PSFB_AFB, 15) +//https://tools.ietf.org/html/rfc4585#section-6.2 +//6.2. Transport Layer Feedback Messages +// +// Transport layer FB messages are identified by the value RTPFB as RTCP +// message type. +// +// A single general purpose transport layer FB message is defined in +// this document: Generic NACK. It is identified by means of the FMT +// parameter as follows: +// +// 0: unassigned +// 1: Generic NACK +// 2: reserved https://tools.ietf.org/html/rfc5104#section-4.2 +// 3: TMMBR https://tools.ietf.org/html/rfc5104#section-4.2.1.1 +// 4: TMMBN https://tools.ietf.org/html/rfc5104#section-4.2.2.1 +// 5-14: unassigned +// 15 transport-cc https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 +// 16-30: unassigned +// 31: reserved for future expansion of the identifier number space +#define RTPFB_TYPE_MAP(XX) \ + XX(RTCP_RTPFB_NACK, 1) \ + XX(RTCP_RTPFB_TMMBR, 3) \ + XX(RTCP_RTPFB_TMMBN, 4) \ + XX(RTCP_RTPFB_TWCC, 15) + //rtcp类型枚举 enum class RtcpType : uint8_t { #define XX(key, value) key = value, @@ -436,7 +466,7 @@ public: //text长度股，可以为0 uint8_t length; //不定长 - char text; + char text[1]; //最后以RTCP_SDES_END结尾 //只字段为占位字段，不代表真实位置 uint8_t end; @@ -501,7 +531,8 @@ private: void net2Host(size_t size); } PACKED; -//6.1. Common Packet Format for Feedback Messages +// https://tools.ietf.org/html/rfc4585#section-6.1 +// 6.1. Common Packet Format for Feedback Messages // // All FB messages MUST use a common packet format that is depicted in // Figure 3: @@ -569,7 +600,7 @@ public: /* 可选 */ uint8_t reason_len; - char reason; + char reason[1]; public: /** diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 724f03b2..60faf0ab 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -1,5 +1,17 @@ -// -// Created by xzl on 2021/4/13. -// +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + #include "RtcpFCI.h" + +namespace mediakit { + + +}//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index d55ef1ea..adb4abf3 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -1,4 +1,4 @@ -/* +/* * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). @@ -13,42 +13,17 @@ #include "Rtcp.h" -//https://tools.ietf.org/html/rfc4585 - namespace mediakit { -//https://tools.ietf.org/html/rfc4585#section-6.2.1 -// 0 1 2 3 -// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | PID | BLP | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// -// Figure 4: Syntax for the Generic NACK message - -class FCI_NACK { -public: - // The PID field is used to specify a lost packet. The PID field - // refers to the RTP sequence number of the lost packet. - uint16_t pid; - // bitmask of following lost packets (BLP): 16 bits - uint16_t blp; -} PACKED; +/////////////////////////////////////////// PSFB //////////////////////////////////////////////////// +//PSFB fmt = 2 //https://tools.ietf.org/html/rfc4585#section-6.3.2.2 -// The Slice Loss Indication uses one additional FCI field, the content -// of which is depicted in Figure 6. The length of the FB message MUST -// be set to 2+n, with n being the number of SLIs contained in the FCI -// field. -// // 0 1 2 3 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | First | Number | PictureID | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// -// Figure 6: Syntax of the Slice Loss Indication (SLI) -// class FCI_SLI { public: uint32_t first : 13; @@ -56,11 +31,8 @@ public: uint32_t pic_id : 6; } PACKED; +//PSFB fmt = 3 //https://tools.ietf.org/html/rfc4585#section-6.3.3.2 -//6.3.3.2. Format -// -// The FCI for the RPSI message follows the format depicted in Figure 7: -// // 0 1 2 3 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ @@ -68,8 +40,6 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | defined per codec ... | Padding (0) | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// -// Figure 7: Syntax of the Reference Picture Selection Indication (RPSI) class FCI_RPSI { public: //The number of unused bits required to pad the length of the RPSI @@ -99,7 +69,62 @@ public: // MUST be indicated by the PB field. } PACKED; -//tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 +//PSFB fmt = 4 +//https://tools.ietf.org/html/rfc5104#section-4.3.1.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. | Reserved | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_FIR { + +} PACKED; + +//PSFB fmt = 5 +//https://tools.ietf.org/html/rfc5104#section-4.3.2.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. | Reserved | Index | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_TSTR { + +} PACKED; + +//PSFB fmt = 6 +//https://tools.ietf.org/html/rfc5104#section-4.3.2.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. | Reserved | Index | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_TSTN { + +} PACKED; + +//PSFB fmt = 7 +//https://tools.ietf.org/html/rfc5104#section-4.3.4.1 +//0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. |0| Payload Type| Length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | VBCM Octet String.... | Padding | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_VBCM { + +} PACKED; + +//PSFB fmt = 15 +//https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 // 0 1 2 3 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ @@ -144,5 +169,95 @@ public: } PACKED; +/////////////////////////////////////////// RTPFB //////////////////////////////////////////////////// + +//RTPFB fmt = 1 +//https://tools.ietf.org/html/rfc4585#section-6.2.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PID | BLP | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_NACK { +public: + // The PID field is used to specify a lost packet. The PID field + // refers to the RTP sequence number of the lost packet. + uint16_t pid; + // bitmask of following lost packets (BLP): 16 bits + uint16_t blp; +} PACKED; + +//RTPFB fmt = 3 +//https://tools.ietf.org/html/rfc5104#section-4.2.1.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | MxTBR Exp | MxTBR Mantissa |Measured Overhead| +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_TMMBR { +public: + //SSRC (32 bits): The SSRC value of the media sender that is + // requested to obey the new maximum bit rate. + uint32_t ssrc; + + // MxTBR Exp (6 bits): The exponential scaling of the mantissa for the + // maximum total media bit rate value. The value is an + // unsigned integer [0..63]. + uint32_t max_tbr_exp: 6; + + // MxTBR Mantissa (17 bits): The mantissa of the maximum total media + // bit rate value as an unsigned integer. + uint32_t max_mantissa: 17; + + // Measured Overhead (9 bits): The measured average packet overhead + // value in bytes. The measurement SHALL be done according + // to the description in section 4.2.1.2. The value is an + // unsigned integer [0..511]. + uint32_t measured_overhead: 9; +} PACKED; + +//RTPFB fmt = 4 +// https://tools.ietf.org/html/rfc5104#section-4.2.2.1 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | MxTBR Exp | MxTBR Mantissa |Measured Overhead| +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_TMMBN : public FCI_TMMBR{ +public: + +} PACKED; + +//RTPFB fmt = 15 +//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-3.1 +//https://zhuanlan.zhihu.com/p/206656654 +//0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | base sequence number | packet status count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | reference time | fb pkt. count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | packet chunk | packet chunk | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | packet chunk | recv delta | recv delta | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | recv delta | recv delta | zero padding | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class FCI_TWCC : public FCI_TMMBR{ +public: + +} PACKED; + } //namespace mediakit #endif //ZLMEDIAKIT_RTCPFCI_H From 765db1eb8b1cbfbd7fc7426e86a92129727a6cbc Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 14 Apr 2021 16:19:46 +0800 Subject: [PATCH 099/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtcp=20fci=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 128 ++++++++++++++++++++++++++++++++++++++++ src/Rtcp/RtcpFCI.h | 107 ++++++++++++++++++++++----------- tests/test_rtcp_fci.cpp | 41 +++++++++++++ 3 files changed, 243 insertions(+), 33 deletions(-) create mode 100644 tests/test_rtcp_fci.cpp diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 60faf0ab..d03d1a60 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -9,9 +9,137 @@ */ +#include "assert.h" #include "RtcpFCI.h" +#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); namespace mediakit { +FCI_SLI::FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id) { + //13 bits + first &= 0x1FFF; + //13 bits + number &= 0x1FFF; + //6 bits + pic_id &= 0x3F; + data = (first << 19) | (number << 6) | pic_id; + data = htonl(data); +} + +uint16_t FCI_SLI::getFirst() const { + return data >> 19; +} + +uint16_t FCI_SLI::getNumber() const { + return (data >> 6) & 0x1FFF; +} + +uint8_t FCI_SLI::getPicID() const { + return data & 0x3F; +} + +void FCI_SLI::net2Host() { + data = ntohl(data); +} + +string FCI_SLI::dumpString() const { + return StrPrinter << "First:" << getFirst() << ", Number:" << getNumber() << ", PictureID:" << (int)getPicID(); +} + +/////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void FCI_FIR::net2Host() { + ssrc = ntohl(ssrc); + reserved = ntohl(reserved) >> 8; +} + +string FCI_FIR::dumpString() const { + return StrPrinter << "ssrc:" << ssrc << ", seq_number:" << seq_number << ", reserved:" << reserved; +} + +FCI_FIR::FCI_FIR(uint32_t ssrc, uint8_t seq_number, uint32_t reserved) { + this->ssrc = htonl(ssrc); + this->seq_number = seq_number; + this->reserved = htonl(reserved) >> 8; +} + +/////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +string FCI_REMB::create(const initializer_list &ssrcs, uint32_t bitrate) { + CHECK(ssrcs.size() > 0 && ssrcs.size() <= 0xFF); + string ret; + ret.resize(kSize + ssrcs.size() * 4); + FCI_REMB *thiz = (FCI_REMB *) ret.data(); + memcpy(thiz->magic, "REMB", 4); + + /* bitrate --> BR Exp/BR Mantissa */ + uint8_t b = 0; + uint8_t exp = 0; + uint32_t mantissa = 0; + for (b = 0; b < 32; b++) { + if (bitrate <= ((uint32_t) 0x3FFFF << b)) { + exp = b; + break; + } + } + if (b > 31) { + b = 31; + } + mantissa = bitrate >> b; + //Num SSRC (8 bits) + thiz->bitrate[0] = ssrcs.size() & 0xFF; + //BR Exp (6 bits)/BR Mantissa (18 bits) + thiz->bitrate[1] = (uint8_t) ((exp << 2) + ((mantissa >> 16) & 0x03)); + //BR Mantissa (18 bits) + thiz->bitrate[2] = (uint8_t) (mantissa >> 8); + //BR Mantissa (18 bits) + thiz->bitrate[3] = (uint8_t) (mantissa); + + //设置ssrc列表 + auto ptr = thiz->ssrc_feedback; + for (auto &ssrc : ssrcs) { + *(ptr++) = htonl(ssrc); + } + return ret; +} + +void FCI_REMB::net2Host(size_t total_size) { + CHECK(total_size >= kSize); + CHECK(memcmp(magic, "REMB", 4) == 0); + auto num_ssrc = bitrate[0]; + auto expect_size = kSize + 4 * num_ssrc; + CHECK(total_size == expect_size); + auto ptr = ssrc_feedback; + while (num_ssrc--) { + *(ptr++) = ntohl(*ptr); + } +} + +uint32_t FCI_REMB::getBitRate() const { + uint8_t exp = (bitrate[1] >> 2) & 0x3F; + uint32_t mantissa = (bitrate[1] & 0x03) << 16; + mantissa += (bitrate[2] << 8); + mantissa += (bitrate[3]); + return mantissa << exp; +} + +vector FCI_REMB::getSSRC() { + vector ret; + auto num_ssrc = bitrate[0]; + auto ptr = ssrc_feedback; + while (num_ssrc--) { + ret.emplace_back(ptr++); + } + return ret; +} + +string FCI_REMB::dumpString() const { + _StrPrinter printer; + printer << "bitrate:" << getBitRate() << ", ssrc:"; + for (auto &ssrc : ((FCI_REMB *) this)->getSSRC()) { + printer << *ssrc << " "; + } + return printer; +} }//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index adb4abf3..f36056d9 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -24,11 +24,36 @@ namespace mediakit { // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | First | Number | PictureID | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +//First: 13 bits +// The macroblock (MB) address of the first lost macroblock. The MB +// numbering is done such that the macroblock in the upper left +// corner of the picture is considered macroblock number 1 and the +// number for each macroblock increases from left to right and then +// from top to bottom in raster-scan order (such that if there is a +// total of N macroblocks in a picture, the bottom right macroblock +// is considered macroblock number N). +// +// Number: 13 bits +// The number of lost macroblocks, in scan order as discussed above. +// +// PictureID: 6 bits +// The six least significant bits of the codec-specific identifier +// that is used to reference the picture in which the loss of the +// macroblock(s) has occurred. For many video codecs, the PictureID +// is identical to the Temporal Reference. class FCI_SLI { public: - uint32_t first : 13; - uint32_t number : 13; - uint32_t pic_id : 6; + static size_t constexpr kSize = 4; + + FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id); + uint16_t getFirst() const; + uint16_t getNumber() const; + uint8_t getPicID() const; + void net2Host(); + string dumpString() const; + +private: + uint32_t data; } PACKED; //PSFB fmt = 3 @@ -61,12 +86,15 @@ public: // Native RPSI bit string: variable length // The RPSI information as natively defined by the video codec. - char str[1]; + char bit_string[5]; //Padding: #PB bits // A number of bits set to zero to fill up the contents of the RPSI // message to the next 32-bit boundary. The number of padding bits // MUST be indicated by the PB field. + uint8_t padding; + + static size_t constexpr kSize = 8; } PACKED; //PSFB fmt = 4 @@ -79,6 +107,15 @@ public: // | Seq nr. | Reserved | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_FIR { +public: + uint32_t ssrc; + uint32_t seq_number: 8; + uint32_t reserved: 24; + + static size_t constexpr kSize = 8; + FCI_FIR(uint32_t ssrc, uint8_t seq_number, uint32_t reserved = 0); + void net2Host(); + string dumpString() const; } PACKED; @@ -92,7 +129,8 @@ class FCI_FIR { // | Seq nr. | Reserved | Index | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_TSTR { - +public: + static size_t constexpr kSize = 8; } PACKED; //PSFB fmt = 6 @@ -104,7 +142,7 @@ class FCI_TSTR { // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | Seq nr. | Reserved | Index | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -class FCI_TSTN { +class FCI_TSTN : public FCI_TSTR{ } PACKED; @@ -120,7 +158,8 @@ class FCI_TSTN { // | VBCM Octet String.... | Padding | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_VBCM { - +public: + static size_t constexpr kSize = 12; } PACKED; //PSFB fmt = 15 @@ -128,12 +167,6 @@ class FCI_VBCM { // 0 1 2 3 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// |V=2|P| FMT=15 | PT=206 | length | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | SSRC of packet sender | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | SSRC of media source | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | Unique identifier 'R' 'E' 'M' 'B' | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | Num SSRC | BR Exp | BR Mantissa | @@ -141,31 +174,39 @@ class FCI_VBCM { // | SSRC feedback | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | ... | +// Num SSRC (8 bits): Number of SSRCs in this message. +// +// BR Exp (6 bits): The exponential scaling of the mantissa for the +// maximum total media bit rate value, ignoring all packet +// overhead. The value is an unsigned integer [0..63], as +// in RFC 5104 section 4.2.2.1. +// +// BR Mantissa (18 bits): The mantissa of the maximum total media bit +// rate (ignoring all packet overhead) that the sender of +// the REMB estimates. The BR is the estimate of the +// traveled path for the SSRCs reported in this message. +// The value is an unsigned integer in number of bits per +// second. +// +// SSRC feedback (32 bits) Consists of one or more SSRC entries which +// this feedback message applies to. class FCI_REMB { public: + static size_t constexpr kSize = 8; + + static string create(const std::initializer_list &ssrcs, uint32_t bitrate); + void net2Host(size_t total_size); + string dumpString() const; + uint32_t getBitRate() const; + vector getSSRC(); + //Unique identifier 'R' 'E' 'M' 'B' char magic[4]; - - //Number of SSRCs in this message. - uint32_t num_ssrc : 8; - - // BR Exp (6 bits): The exponential scaling of the mantissa for the - // maximum total media bit rate value, ignoring all packet - // overhead. The value is an unsigned integer [0..63], as - // in RFC 5104 section 4.2.2.1. - uint32_t br_exp : 6; - - //BR Mantissa (18 bits): The mantissa of the maximum total media bit - // rate (ignoring all packet overhead) that the sender of - // the REMB estimates. The BR is the estimate of the - // traveled path for the SSRCs reported in this message. - // The value is an unsigned integer in number of bits per - // second. - uint32_t br_mantissa : 18; - + //Num SSRC (8 bits)/BR Exp (6 bits)/ BR Mantissa (18 bits) + uint8_t bitrate[4]; // SSRC feedback (32 bits) Consists of one or more SSRC entries which // this feedback message applies to. - uint32_t ssrc_feedback; + uint32_t ssrc_feedback[1]; } PACKED; @@ -254,7 +295,7 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | recv delta | recv delta | zero padding | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -class FCI_TWCC : public FCI_TMMBR{ +class FCI_TWCC{ public: } PACKED; diff --git a/tests/test_rtcp_fci.cpp b/tests/test_rtcp_fci.cpp new file mode 100644 index 00000000..c974d4b0 --- /dev/null +++ b/tests/test_rtcp_fci.cpp @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2016 The ZLToolKit project authors. All Rights Reserved. + * + * This file is part of ZLToolKit(https://github.com/xia-chu/ZLToolKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#include +#include "Util/logger.h" +#include "Rtcp/RtcpFCI.h" +using namespace std; +using namespace toolkit; +using namespace mediakit; + +int main() { + //初始化日志系统 + Logger::Instance().add(std::make_shared ()); + + { + FCI_SLI fci(0xFFFF, 0, 0xFF); + InfoL << 0b10101010101 << " " << 0b01010101010 << " " << (int) 0b101010 << " " << hexdump(&fci, FCI_SLI::kSize); + fci.net2Host(); + InfoL << fci.dumpString(); + } + { + FCI_FIR fci(123456, 139, 456789); + InfoL << hexdump(&fci, FCI_FIR::kSize); + fci.net2Host(); + InfoL << fci.dumpString(); + } + { + auto str = FCI_REMB::create({1234,2345,5678}, 4 * 1024 * 1024); + FCI_REMB *ptr = (FCI_REMB *)str.data(); + ptr->net2Host(str.size()); + InfoL << ptr->dumpString(); + } + return 0; +} From 6e4dfaf53f133e24ec6ed82e4f1f4884d129873b Mon Sep 17 00:00:00 2001 From: xgj Date: Thu, 15 Apr 2021 00:15:41 +0800 Subject: [PATCH 100/218] fix webrtc h264 rtp Packetizer error --- src/Extension/H264Rtp.cpp | 47 ++++++++++++++++++++++++++++++++++++--- src/Extension/H264Rtp.h | 2 ++ 2 files changed, 46 insertions(+), 3 deletions(-) diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp index 54a7fc5c..ddf81717 100644 --- a/src/Extension/H264Rtp.cpp +++ b/src/Extension/H264Rtp.cpp @@ -189,8 +189,41 @@ void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) { auto len = frame->size() - frame->prefixSize(); auto pts = frame->pts(); auto nal_type = H264_TYPE(ptr[0]); - auto packet_size = getMaxSize() - 2; + if(nal_type == H264Frame::NAL_SEI || nal_type == H264Frame::NAL_AUD){ + return; + } + if(nal_type == H264Frame::NAL_SPS){ + _sps = std::string(ptr,len); + return; + } + + if(nal_type == H264Frame::NAL_PPS){ + _pps = std::string(ptr,len); + return; + } + + if(nal_type == H264Frame::NAL_IDR) + {// 保证每一个I帧前都有SPS与PPS ,为了兼容webrtc 需要在一个rtp包中，并且只能是 STAP-A + // https://blog.csdn.net/momo0853/article/details/88872873 + auto rtp = makeRtp(getTrackType(), nullptr,_sps.size()+_pps.size()+2*2+1,true,pts); + uint8_t *payload = rtp->getPayload(); + payload[0] = 24; + payload[1] = _sps.size() >> 8; + payload[2] = _sps.size() & 0xff; + memcpy(payload+3,(uint8_t *) _sps.data(),_sps.size()); + + payload[_sps.size()+3] = _pps.size() >> 8; + payload[_sps.size()+4] = _pps.size() & 0xff; + + memcpy(payload+3+_sps.size()+2,(uint8_t *) _pps.data(),_pps.size()); + RtpCodec::inputRtp(rtp,true); + } + + + + auto packet_size = getMaxSize() - 2; + //InfoL<<"nal type = "< using namespace mediakit; @@ -610,12 +609,12 @@ void SdpAttrFmtp::parse(const string &str) { trim(item); auto pos = item.find('='); if(pos == string::npos){ - arr.emplace_back(std::make_pair(item, "")); + fmtp.emplace(std::make_pair(item, "")); } else { - arr.emplace_back(std::make_pair(item.substr(0, pos), item.substr(pos + 1))); + fmtp.emplace(std::make_pair(item.substr(0, pos), item.substr(pos + 1))); } } - if (arr.empty()) { + if (fmtp.empty()) { SDP_THROW(); } } @@ -624,7 +623,7 @@ string SdpAttrFmtp::toString() const { if (value.empty()) { value = to_string(pt); int i = 0; - for (auto &pr : arr) { + for (auto &pr : fmtp) { value += (i++ ? ';' : ' '); value += pr.first + "=" + pr.second; } @@ -918,7 +917,7 @@ void RtcSession::loadFrom(const string &str, bool check) { auto fmtp_it = fmtp_map.find(pt); if (fmtp_it != fmtp_map.end()) { - plan.fmtp = fmtp_it->second.arr; + plan.fmtp = fmtp_it->second.fmtp; } for (auto rtpfb_it = rtcpfb_map.find(pt); rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) { @@ -1088,7 +1087,7 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ if (!p.fmtp.empty()) { auto fmtp = std::make_shared(); fmtp->pt = p.pt; - fmtp->arr = p.fmtp; + fmtp->fmtp = p.fmtp; sdp_media.items.emplace_back(wrapSdpAttr(std::move(fmtp))); } } @@ -1255,12 +1254,14 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ ice_renomination = false; switch (type) { case TrackAudio: { - preferred_codec = {CodecAAC, CodecOpus, CodecG711U, CodecG711A}; + //此处调整偏好的编码格式优先级 + preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; rtcp_fb = {"transport-cc"}; extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"}; break; } case TrackVideo: { + //此处调整偏好的编码格式优先级 preferred_codec = {CodecH264, CodecH265}; rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"}; extmap = {"2 urn:ietf:params:rtp-hdrext:toffset", @@ -1441,6 +1442,7 @@ RETRY: //添加媒体plan answer_media.plan.emplace_back(*offer_plan_ptr); + onSelectPlan(answer_media.plan.back(), codec); //添加rtx,red,ulpfec plan if (configure.support_red || configure.support_rtx || configure.support_ulpfec) { @@ -1535,14 +1537,6 @@ void RtcConfigure::setPlayRtspInfo(const string &sdp){ } } -static map toMap(const vector > &fmt) { - map ret; - for (auto &pr : fmt) { - ret.emplace(pr); - } - return ret; -} - static const string kProfile{"profile-level-id"}; static const string kMode{"packetization-mode"}; @@ -1555,21 +1549,21 @@ bool RtcConfigure::onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec){ return true; } if (_rtsp_video_plan && codec == CodecH264 && getCodecId(_rtsp_video_plan->codec) == CodecH264) { - //h264时，匹配packetization-mode和profile-level-id - auto rtc_fmt_map = toMap(plan.fmtp); - auto rtsp_fmt_map = toMap(_rtsp_video_plan->fmtp); - auto &profile = rtsp_fmt_map[kProfile]; - if (!profile.empty() && strcasecmp(profile.data(), rtc_fmt_map[kProfile].data())) { + //h264时，profile-level-id + if (strcasecmp(_rtsp_video_plan->fmtp[kProfile].data(), const_cast(plan).fmtp[kProfile].data())) { //profile-level-id 不匹配 return false; } - auto &mode = rtsp_fmt_map[kMode]; - if (!mode.empty() && atoi(mode.data()) != atoi(rtc_fmt_map[kMode].data())) { - //packetization-mode不匹配 - return false; - } return true; } return true; } + +void RtcConfigure::onSelectPlan(RtcCodecPlan &plan, CodecId codec){ + if (_rtsp_video_plan && codec == CodecH264 && getCodecId(_rtsp_video_plan->codec) == CodecH264) { + //h264时，设置packetization-mod为一致 + auto mode = _rtsp_video_plan->fmtp[kMode]; + plan.fmtp[kMode] = mode.empty() ? "0" : mode; + } +} diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 9247018a..272d6dd9 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -15,6 +15,7 @@ #include #include "assert.h" #include "Extension/Frame.h" +#include "Common/Parser.h" using namespace std; using namespace mediakit; @@ -372,7 +373,7 @@ class SdpAttrFmtp : public SdpItem { public: //fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f uint8_t pt; - vector > arr; + map fmtp; void parse(const string &str) override; string toString() const override; const char* getKey() const override { return "fmtp";} @@ -589,7 +590,7 @@ public: uint32_t channel = 0; //rtcp反馈 vector rtcp_fb; - vector > fmtp; + map fmtp; string getFmtp(const char *key) const; }; @@ -715,6 +716,7 @@ public: private: void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); bool onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec); + void onSelectPlan(RtcCodecPlan &plan, CodecId codec); private: RtcCodecPlan::Ptr _rtsp_video_plan; From e00fe51b267ffc58377c176d2b0e307655e1ff66 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 15 Apr 2021 19:35:49 +0800 Subject: [PATCH 104/218] =?UTF-8?q?=E5=85=B3=E9=97=AD=E6=B5=8B=E8=AF=95?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index afee33be..a74ec4a7 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -555,7 +555,7 @@ void RtcpBye::net2Host(size_t size) { } } -#if 1 +#if 0 #include "Util/onceToken.h" static toolkit::onceToken token([](){ From 9d385b36fde14c1a13dbbef1a072496329d1659b Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 15 Apr 2021 19:37:13 +0800 Subject: [PATCH 105/218] =?UTF-8?q?=E4=BC=98=E5=8C=96=E5=A4=9Aslice?= =?UTF-8?q?=E6=83=85=E5=86=B5=E4=B8=8B=E5=85=B3=E9=94=AE=E5=B8=A7=E5=88=A4?= =?UTF-8?q?=E6=96=AD=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/H264.cpp | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/src/Extension/H264.cpp b/src/Extension/H264.cpp index 04c638e1..1db1f676 100644 --- a/src/Extension/H264.cpp +++ b/src/Extension/H264.cpp @@ -181,17 +181,15 @@ void H264Track::inputFrame_l(const Frame::Ptr &frame){ _pps = string(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize()); break; } - case H264Frame::NAL_IDR: { - insertConfigFrame(frame); - VideoTrack::inputFrame(frame); - break; - } case H264Frame::NAL_AUD: { //忽略AUD帧; break; } default: + if (frame->keyFrame()) { + insertConfigFrame(frame); + } VideoTrack::inputFrame(frame); break; } @@ -281,7 +279,8 @@ Sdp::Ptr H264Track::getSdp() { //////////////////////////////////////////////////////////////////////////////////////////////////// bool H264Frame::keyFrame() const { - return H264_TYPE(_buffer[_prefix_size]) == H264Frame::NAL_IDR; + //多slice 一帧的情况下检查 first_mb_in_slice 是否为0 表示其为一帧的开始 + return H264_TYPE(_buffer[_prefix_size]) == H264Frame::NAL_IDR && (_buffer[_prefix_size + 1] & 0x80); } bool H264Frame::configFrame() const { From d76c38ef72be6736ad7300ff70a03795fb93e1ba Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 15 Apr 2021 19:39:24 +0800 Subject: [PATCH 106/218] =?UTF-8?q?=E8=B0=83=E6=95=B4264=20rtp=E6=89=93?= =?UTF-8?q?=E5=8C=85=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/H264Rtp.cpp | 175 +++++++++++++++++++------------------- src/Extension/H264Rtp.h | 8 +- 2 files changed, 96 insertions(+), 87 deletions(-) diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp index ec45a71f..a9a0c135 100644 --- a/src/Extension/H264Rtp.cpp +++ b/src/Extension/H264Rtp.cpp @@ -184,107 +184,110 @@ H264RtpEncoder::H264RtpEncoder(uint32_t ssrc, uint32_t mtu, uint32_t sample_rate : RtpInfo(ssrc, mtu, sample_rate, pt, interleaved) { } -void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) { - auto ptr = frame->data() + frame->prefixSize(); - auto len = frame->size() - frame->prefixSize(); - auto pts = frame->pts(); - auto nal_type = H264_TYPE(ptr[0]); - if(nal_type == H264Frame::NAL_SEI || nal_type == H264Frame::NAL_AUD){ +void H264RtpEncoder::insertConfigFrame(uint32_t pts){ + if (_sps.empty() || _pps.empty()) { return; } + //gop缓存从sps开始，sps、pps后面还有时间戳相同的关键帧，所以mark bit为false + packRtp(_sps.data(), _sps.size(), pts, false, true); + packRtp(_pps.data(), _pps.size(), pts, false, false); +} - if(nal_type == H264Frame::NAL_SPS){ - _sps = std::string(ptr,len); - return; +void H264RtpEncoder::packRtp(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){ + if (len + 3 <= getMaxSize()) { + //STAP-A模式打包小于MTU + packRtpStapA(ptr, len, pts, is_mark, gop_pos); + } else { + //STAP-A模式打包会大于MTU,所以采用FU-A模式 + packRtpFu(ptr, len, pts, is_mark, gop_pos); } +} - if(nal_type == H264Frame::NAL_PPS){ - _pps = std::string(ptr,len); - return; - } - - if(!_last_frame){ - _last_frame = frame; - return; - } - // 上一帧打包，保证rtp 的mark是正确的 - bool isMark = _last_frame->pts() != frame->pts(); - ptr = _last_frame->data() + _last_frame->prefixSize(); - len = _last_frame->size() - _last_frame->prefixSize(); - pts = _last_frame->pts(); - nal_type = H264_TYPE(ptr[0]); - if(nal_type == H264Frame::NAL_IDR && (ptr[1]&0x80)) - {// 保证每一个I帧前都有SPS与PPS ,为了兼容webrtc 需要在一个rtp包中，并且只能是 STAP-A - // https://blog.csdn.net/momo0853/article/details/88872873 - // 多slice 一帧的情况下检查 first_mb_in_slice 是否为0 表示其为一帧的开始，SPS PPS 只有在帧开始时，才插入 - auto rtp = makeRtp(getTrackType(), nullptr,_sps.size()+_pps.size()+2*2+1,false,pts); - uint8_t *payload = rtp->getPayload(); - payload[0] = 24; - payload[1] = _sps.size() >> 8; - payload[2] = _sps.size() & 0xff; - memcpy(payload+3,(uint8_t *) _sps.data(),_sps.size()); - - payload[_sps.size()+3] = _pps.size() >> 8; - payload[_sps.size()+4] = _pps.size() & 0xff; - - memcpy(payload+3+_sps.size()+2,(uint8_t *) _pps.data(),_pps.size()); - RtpCodec::inputRtp(rtp,true); - } - - - +void H264RtpEncoder::packRtpFu(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){ auto packet_size = getMaxSize() - 2; - //InfoL<<"nal type = "< getBitArray() const; + string dumpString() const; + + template + FCI_NACK(uint16_t pid_h, const Type &type){ + uint16_t blp_h = 0; + int i = kBitSize; + for (auto &item : type) { + --i; + if (item) { + blp_h |= (1 << i); + } + } + blp = htons(blp_h); + pid = htons(pid_h); + } } PACKED; //RTPFB fmt = 3 diff --git a/tests/test_rtcp_fci.cpp b/tests/test_rtcp_fci.cpp index c974d4b0..d8fd1eca 100644 --- a/tests/test_rtcp_fci.cpp +++ b/tests/test_rtcp_fci.cpp @@ -37,5 +37,10 @@ int main() { ptr->net2Host(str.size()); InfoL << ptr->dumpString(); } + { + FCI_NACK nack(1234, vector({1, 0, 0, 0, 1, 0, 1, 0, 1, 0})); + nack.net2Host(); + InfoL << nack.dumpString(); + } return 0; } From 007c04d95899d37ef9b274ae0380612c0b667d3f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 20 Apr 2021 17:17:56 +0800 Subject: [PATCH 110/218] =?UTF-8?q?=E7=BC=96=E8=AF=91=E8=AD=A6=E5=91=8A?= =?UTF-8?q?=E4=BF=AE=E5=A4=8D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Common/MultiMediaSourceMuxer.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/Common/MultiMediaSourceMuxer.cpp b/src/Common/MultiMediaSourceMuxer.cpp index 52e34657..7d707008 100644 --- a/src/Common/MultiMediaSourceMuxer.cpp +++ b/src/Common/MultiMediaSourceMuxer.cpp @@ -308,7 +308,7 @@ void MultiMediaSourceMuxer::setTrackListener(const std::weak_ptrtotalReaderCount() + _rtp_sender.size(); + return _muxer->totalReaderCount() + (int)_rtp_sender.size(); #else return _muxer->totalReaderCount(); #endif From 5db61547079e3d5049f309aa0b97ff967d26f4a6 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 22 Apr 2021 11:43:33 +0800 Subject: [PATCH 111/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0TWCC=E9=83=A8?= =?UTF-8?q?=E5=88=86=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 78 ++++++++++++++++++++++++++++++++++++ src/Rtcp/RtcpFCI.h | 87 +++++++++++++++++++++++++++++++++++++++++ tests/test_rtcp_fci.cpp | 23 +++++++++++ 3 files changed, 188 insertions(+) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 44d22055..61a4cd16 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -167,5 +167,83 @@ string FCI_NACK::dumpString() const { return std::move(printer); } +/////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +RunLengthChunk::RunLengthChunk(SymbolStatus status, uint16_t run_length) { + type = 0; + symbol = (uint8_t)status & 0x03; + run_length_high = (run_length >> 8) & 0x1F; + run_length_low = run_length & 0xFF; +} + +uint16_t RunLengthChunk::getRunLength() const { + CHECK(type == 0); + return run_length_high << 8 | run_length_low; +} + +string RunLengthChunk::dumpString() const{ + _StrPrinter printer; + printer << "run length chunk, symbol:" << (int)symbol << ", run length:" << getRunLength(); + return std::move(printer); +} + +/////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +StatusVecChunk::StatusVecChunk(const vector &status) { + uint16_t value = 0; + type = 1; + if (status.size() == 14) { + symbol = 0; + } else if (status.size() == 7) { + symbol = 1; + } else { + //非法 + CHECK(0); + } + int i = 13; + for (auto &item : status) { + CHECK(item <= SymbolStatus::reserved); + if (!symbol) { + CHECK(item <= SymbolStatus::small_delta); + value |= (int) item << i; + --i; + } else { + value |= (int) item << (i - 1); + i -= 2; + } + } + symbol_list_low = value & 0xFF; + symbol_list_high = (value >> 8 ) & 0x1F; +} + +vector StatusVecChunk::getSymbolList() const { + CHECK(type == 1); + vector ret; + auto thiz = ntohs(*((uint16_t *) this)); + if (symbol == 0) { + //s = 0 时，表示symbollist的每一个bit能表示一个数据包的到达状态 + for (int i = 13; i >= 0; --i) { + SymbolStatus status = (SymbolStatus) ((bool) (thiz & (1 << i))); + ret.emplace_back(status); + } + } else { + //s = 1 时，表示symbollist每两个bit表示一个数据包的状态 + for (int i = 12; i >= 0; i -= 2) { + SymbolStatus status = (SymbolStatus) ((thiz & (3 << i)) >> i); + ret.emplace_back(status); + } + } + return ret; +} + +string StatusVecChunk::dumpString() const { + _StrPrinter printer; + printer << "status vector chunk, symbol:" << (int) symbol << ", symbol list:"; + auto vec = getSymbolList(); + for (auto &item : vec) { + printer << (int) item << " "; + } + return std::move(printer); +} }//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index ed216149..3eedfa6c 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -222,6 +222,7 @@ public: class FCI_NACK { public: static constexpr size_t kBitSize = 16; + static constexpr size_t kSize = 4; // The PID field is used to specify a lost packet. The PID field // refers to the RTP sequence number of the lost packet. @@ -259,6 +260,8 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_TMMBR { public: + static size_t constexpr kSize = 8; + //SSRC (32 bits): The SSRC value of the media sender that is // requested to obey the new maximum bit rate. uint32_t ssrc; @@ -315,8 +318,92 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | recv delta | recv delta | zero padding | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +enum class SymbolStatus : uint8_t{ + //Packet not received + not_received = 0, + //Packet received, small delta （所谓small detal是指能用一个字节表示的数值） + small_delta = 1, + // Packet received, large ornegative delta （large即是能用两个字节表示的数值） + large_delta = 2, + //Reserved + reserved = 3 +}; + +class RunLengthChunk { +public: + static size_t constexpr kSize = 2; + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T| S | Run Length | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +#if __BYTE_ORDER == __BIG_ENDIAN + uint16_t type: 1; + uint16_t symbol: 2; + uint16_t run_length_high: 5; +#else + // Run Length 高5位 + uint16_t run_length_high: 5; + //参考SymbolStatus定义 + uint16_t symbol: 2; + //固定为0 + uint16_t type: 1; +#endif + // Run Length 低8位 + uint16_t run_length_low: 8; + + //获取Run Length + uint16_t getRunLength() const; + //构造函数 + RunLengthChunk(SymbolStatus status, uint16_t run_length); + + string dumpString() const; +} PACKED; + +class StatusVecChunk { +public: + static size_t constexpr kSize = 2; + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T| S | Run Length | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +#if __BYTE_ORDER == __BIG_ENDIAN + uint16_t type: 1; + uint16_t symbol: 1; + uint16_t symbol_list_high: 6; +#else + // symbol_list 高6位 + uint16_t symbol_list_high: 6; + //symbol_list中元素是1个还是2个bit + uint16_t symbol: 1; + //固定为1 + uint16_t type: 1; +#endif + // symbol_list 低8位 + uint16_t symbol_list_low: 8; + + //获取symbollist + vector getSymbolList() const; + //构造函数 + StatusVecChunk(const vector &status); + + string dumpString() const; +} PACKED; + class FCI_TWCC{ public: + static size_t constexpr kSize = 12; + + //base sequence number,基础序号,本次反馈的第一个包的序号;也就是RTP扩展头的序列号 + uint16_t base_seq; + //packet status count, 包个数,本次反馈包含多少个包的状态;从基础序号开始算 + uint16_t pkt_status_count; + //reference time,基准时间,绝对时间;计算该包中每个媒体包的到达时间都要基于这个基准时间计算 + uint8_t ref_time[3]; + //feedback packet count,反馈包号,本包是第几个transport-cc包，每次加1 | + uint8_t fb_pkt_count; } PACKED; diff --git a/tests/test_rtcp_fci.cpp b/tests/test_rtcp_fci.cpp index d8fd1eca..aecc821d 100644 --- a/tests/test_rtcp_fci.cpp +++ b/tests/test_rtcp_fci.cpp @@ -42,5 +42,28 @@ int main() { nack.net2Host(); InfoL << nack.dumpString(); } + { + RunLengthChunk chunk(SymbolStatus::large_delta, 8024); + InfoL << hexdump(&chunk, RunLengthChunk::kSize); + InfoL << chunk.dumpString(); + } + + auto lam = [](const initializer_list &lst){ + vector ret; + for(auto &num : lst){ + ret.emplace_back((SymbolStatus)num); + } + return ret; + }; + { + StatusVecChunk chunk(lam({0, 1, 0, 1, 0, 1, 0, 0, 0, 1, 1, 1, 0, 1})); + InfoL << hexdump(&chunk, StatusVecChunk::kSize); + InfoL << chunk.dumpString(); + } + { + StatusVecChunk chunk(lam({0, 1, 2, 3, 0, 1, 2})); + InfoL << hexdump(&chunk, StatusVecChunk::kSize); + InfoL << chunk.dumpString(); + } return 0; } From 90ad90cb78c78730d4d8c4a409a47034d0575092 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 22 Apr 2021 17:34:26 +0800 Subject: [PATCH 112/218] =?UTF-8?q?=E5=88=9D=E6=AD=A5=E5=AE=8C=E6=88=90TWC?= =?UTF-8?q?C=E5=8C=85=E7=9A=84=E8=A7=A3=E6=9E=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 4 +- src/Rtcp/RtcpFCI.cpp | 88 ++++++++++++++++++++++++++++++++++++++ src/Rtcp/RtcpFCI.h | 72 ++++++++++++++++--------------- webrtc/WebRtcTransport.cpp | 16 +++++++ 4 files changed, 144 insertions(+), 36 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index a74ec4a7..35853e80 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -144,8 +144,8 @@ void RtcpHeader::net2Host(size_t len){ } case RtcpType::RTCP_RTPFB: { - //todo 支持rtcp-fb相关功能 - net2Host(); + RtcpPli *pli = (RtcpPli *)this; + pli->net2Host(len); break; } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 61a4cd16..eec9e7f9 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -246,4 +246,92 @@ string StatusVecChunk::dumpString() const { return std::move(printer); } +/////////////////////////////////////////////////////// + +void FCI_TWCC::net2Host(size_t total_size) { + CHECK(total_size >= kSize); + base_seq = ntohs(base_seq); + pkt_status_count = ntohs(pkt_status_count); +} + +uint32_t FCI_TWCC::getReferenceTime() const { + uint32_t ret = 0; + ret |= ref_time[0] << 16; + ret |= ref_time[1] << 8; + ret |= ref_time[2]; + return ret; +} + +static uint16_t getRecvDelta(SymbolStatus status, uint8_t *&ptr, const uint8_t *end){ + uint16_t delta = 0; + switch (status) { + case SymbolStatus::not_received : { + //丢包， recv delta为0个字节 + delta = 0; + break; + } + case SymbolStatus::small_delta : { + CHECK(ptr + 1 <= end); + //时间戳增量小于256， recv delta为1个字节 + delta = *ptr; + ptr += 1; + break; + } + case SymbolStatus::large_delta : { + CHECK(ptr + 2 <= end); + //时间戳增量256~65535间，recv delta为2个字节 + delta = *ptr << 8 | *(ptr + 1); + ptr += 2; + break; + } + default: + //这个逻辑分支不可达到 + CHECK(0); + break; + } + return delta; +} + +map > FCI_TWCC::getPacketChunkList(size_t total_size) const { + map > ret; + auto ptr = (uint8_t *) this + kSize; + auto end = (uint8_t *) this + total_size; + CHECK(ptr < end); + auto seq = base_seq; + auto stamp = getReferenceTime(); + + for (uint8_t i = 0; i < pkt_status_count;) { + CHECK(ptr + RunLengthChunk::kSize <= end) + RunLengthChunk *chunk = (RunLengthChunk *) ptr; + if (!chunk->type) { + //RunLengthChunk + ptr += RunLengthChunk::kSize; + for (auto j = 0; j < chunk->getRunLength(); ++j) { + ret.emplace(seq++, std::make_pair((SymbolStatus) chunk->symbol, stamp)); + stamp += getRecvDelta((SymbolStatus) chunk->symbol, ptr, end); + ++i; + } + } else { + //StatusVecChunk + StatusVecChunk *chunk = (StatusVecChunk *) ptr; + ptr += StatusVecChunk::kSize; + for (auto &symbol : chunk->getSymbolList()) { + ret.emplace(seq++, std::make_pair(symbol, stamp)); + stamp += getRecvDelta(symbol, ptr, end); + ++i; + } + } + } + return ret; +} + +string FCI_TWCC::dumpString(size_t total_size) const { + _StrPrinter printer; + auto map = getPacketChunkList(total_size); + for (auto &pr : map) { + printer << " seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", stamp:" << pr.second.second << "\n"; + } + return std::move(printer); +} + }//namespace mediakit \ No newline at end of file diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index 3eedfa6c..8df51f32 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -296,29 +296,6 @@ public: } PACKED; -//RTPFB fmt = 15 -//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-3.1 -//https://zhuanlan.zhihu.com/p/206656654 -//0 1 2 3 -// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | base sequence number | packet status count | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | reference time | fb pkt. count | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | packet chunk | packet chunk | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// . . -// . . -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | packet chunk | recv delta | recv delta | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// . . -// . . -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | recv delta | recv delta | zero padding | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - enum class SymbolStatus : uint8_t{ //Packet not received not_received = 0, @@ -333,11 +310,11 @@ enum class SymbolStatus : uint8_t{ class RunLengthChunk { public: static size_t constexpr kSize = 2; - // 0 1 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // |T| S | Run Length | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T| S | Run Length | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ #if __BYTE_ORDER == __BIG_ENDIAN uint16_t type: 1; uint16_t symbol: 2; @@ -364,11 +341,11 @@ public: class StatusVecChunk { public: static size_t constexpr kSize = 2; - // 0 1 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // |T| S | Run Length | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T|S| symbol list | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ #if __BYTE_ORDER == __BIG_ENDIAN uint16_t type: 1; uint16_t symbol: 1; @@ -392,9 +369,31 @@ public: string dumpString() const; } PACKED; +//RTPFB fmt = 15 +//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-3.1 +//https://zhuanlan.zhihu.com/p/206656654 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | base sequence number | packet status count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | reference time | fb pkt. count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | packet chunk | packet chunk | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | packet chunk | recv delta | recv delta | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | recv delta | recv delta | zero padding | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_TWCC{ public: - static size_t constexpr kSize = 12; + static size_t constexpr kSize = 8; //base sequence number,基础序号,本次反馈的第一个包的序号;也就是RTP扩展头的序列号 uint16_t base_seq; @@ -405,6 +404,11 @@ public: //feedback packet count,反馈包号,本包是第几个transport-cc包，每次加1 | uint8_t fb_pkt_count; + void net2Host(size_t total_size); + uint32_t getReferenceTime() const; + map > getPacketChunkList(size_t total_size) const; + string dumpString(size_t total_size) const; + } PACKED; } //namespace mediakit diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9a58fb21..d37e5a88 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -501,6 +501,8 @@ private: function _on_before_sort; }; +#include "RTCP/RtcpFCI.h" + void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { _bytes_usage += len; auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len); @@ -546,6 +548,20 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { //todo 支持pli等更多类型的rtcp break; } + case RtcpType::RTCP_RTPFB: { + //todo 测试打印twcc + RtcpPli *rtpfb = (RtcpPli *)rtcp; + auto fci = (uint8_t *)rtpfb + sizeof (RtcpPli); + if(rtpfb->report_count == 15){ + //TWCC + FCI_TWCC *twcc = (FCI_TWCC *) (fci); + auto fci_size = rtpfb->getSize() - 12; + InfoL << hexdump(fci, fci_size); + twcc->net2Host(fci_size); + InfoL << "\n" << twcc->dumpString(fci_size); + } + break; + } default: break; } } From 99a1d06d9b809b1069ca4b76abbb4eb2e3ca3e5f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 22 Apr 2021 18:56:12 +0800 Subject: [PATCH 113/218] =?UTF-8?q?=E4=BF=AE=E6=AD=A3TWCC=E9=94=99?= =?UTF-8?q?=E8=AF=AF=E7=90=86=E8=A7=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 47 ++++++++++++++++++++++++++++++++++---------- src/Rtcp/RtcpFCI.h | 2 +- 2 files changed, 38 insertions(+), 11 deletions(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index eec9e7f9..3f465aa9 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -261,9 +261,35 @@ uint32_t FCI_TWCC::getReferenceTime() const { ret |= ref_time[2]; return ret; } +//3.1.5. Receive Delta +// +// Deltas are represented as multiples of 250us: +// +// o If the "Packet received, small delta" symbol has been appended to +// the status list, an 8-bit unsigned receive delta will be appended +// to recv delta list, representing a delta in the range [0, 63.75] +// ms. +// +// o If the "Packet received, large or negative delta" symbol has been +// appended to the status list, a 16-bit signed receive delta will be +// appended to recv delta list, representing a delta in the range +// [-8192.0, 8191.75] ms. +// +// o If the delta exceeds even the larger limits, a new feedback +// message must be used, where the 24-bit base receive delta can +// cover very large gaps. +// +// The smaller receive delta upper bound of 63.75 ms means that this is +// only viable at about 1000/25.5 ~= 16 packets per second and above. +// With a packet size of 1200 bytes/packet that amounts to a bitrate of +// about 150 kbit/s. +// +// The 0.25 ms resolution means that up to 4000 packets per second can +// be represented. With a 1200 bytes/packet payload, that amounts to +// 38.4 Mbit/s payload bandwidth. -static uint16_t getRecvDelta(SymbolStatus status, uint8_t *&ptr, const uint8_t *end){ - uint16_t delta = 0; +static int16_t getRecvDelta(SymbolStatus status, uint8_t *&ptr, const uint8_t *end){ + int16_t delta = 0; switch (status) { case SymbolStatus::not_received : { //丢包， recv delta为0个字节 @@ -298,29 +324,29 @@ map > FCI_TWCC::getPacketCh auto end = (uint8_t *) this + total_size; CHECK(ptr < end); auto seq = base_seq; - auto stamp = getReferenceTime(); for (uint8_t i = 0; i < pkt_status_count;) { CHECK(ptr + RunLengthChunk::kSize <= end) RunLengthChunk *chunk = (RunLengthChunk *) ptr; if (!chunk->type) { //RunLengthChunk - ptr += RunLengthChunk::kSize; for (auto j = 0; j < chunk->getRunLength(); ++j) { - ret.emplace(seq++, std::make_pair((SymbolStatus) chunk->symbol, stamp)); - stamp += getRecvDelta((SymbolStatus) chunk->symbol, ptr, end); + ret.emplace(seq++, std::make_pair((SymbolStatus) chunk->symbol, 0)); ++i; } } else { //StatusVecChunk StatusVecChunk *chunk = (StatusVecChunk *) ptr; - ptr += StatusVecChunk::kSize; for (auto &symbol : chunk->getSymbolList()) { - ret.emplace(seq++, std::make_pair(symbol, stamp)); - stamp += getRecvDelta(symbol, ptr, end); + ret.emplace(seq++, std::make_pair(symbol, 0)); ++i; } } + ptr += 2; + } + for (auto &pr : ret) { + CHECK(ptr <= end) + pr.second.second = 250 * getRecvDelta(pr.second.first, ptr, end); } return ret; } @@ -328,8 +354,9 @@ map > FCI_TWCC::getPacketCh string FCI_TWCC::dumpString(size_t total_size) const { _StrPrinter printer; auto map = getPacketChunkList(total_size); + printer << "twcc fci, base_seq:" << base_seq << ",pkt_status_count:" << pkt_status_count << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n"; for (auto &pr : map) { - printer << " seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", stamp:" << pr.second.second << "\n"; + printer << "rtp seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", delta:" << pr.second.second << "\n"; } return std::move(printer); } diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index 8df51f32..6676c42c 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -406,7 +406,7 @@ public: void net2Host(size_t total_size); uint32_t getReferenceTime() const; - map > getPacketChunkList(size_t total_size) const; + map > getPacketChunkList(size_t total_size) const; string dumpString(size_t total_size) const; } PACKED; From 48338af700ce40534d94877e4632fe1093e5a1d5 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 23 Apr 2021 15:06:55 +0800 Subject: [PATCH 114/218] =?UTF-8?q?=E5=AE=8C=E5=96=84FCI=E7=9B=B8=E5=85=B3?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- 3rdpart/assert.h | 2 + src/Rtcp/RtcpFCI.cpp | 241 +++++++++++++++++++++++++++++-------- src/Rtcp/RtcpFCI.h | 159 +++++++++--------------- tests/test_rtcp_fci.cpp | 52 +------- webrtc/Sdp.h | 2 - webrtc/WebRtcTransport.cpp | 1 - 6 files changed, 255 insertions(+), 202 deletions(-) diff --git a/3rdpart/assert.h b/3rdpart/assert.h index e5699be1..00369393 100644 --- a/3rdpart/assert.h +++ b/3rdpart/assert.h @@ -21,6 +21,8 @@ extern void Assert_Throw(int failed, const char *exp, const char *func, const ch } #endif +#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__) + #ifndef NDEBUG #ifdef assert #undef assert diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 3f465aa9..79e09c25 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -8,13 +8,16 @@ * may be found in the AUTHORS file in the root of the source tree. */ - -#include "assert.h" #include "RtcpFCI.h" -#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); +#include "Util/logger.h" +using namespace toolkit; namespace mediakit { +void FCI_SLI::check(size_t size){ + CHECK(size == kSize); +} + FCI_SLI::FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id) { //13 bits first &= 0x1FFF; @@ -27,19 +30,15 @@ FCI_SLI::FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id) { } uint16_t FCI_SLI::getFirst() const { - return data >> 19; + return ntohl(data) >> 19; } uint16_t FCI_SLI::getNumber() const { - return (data >> 6) & 0x1FFF; + return (ntohl(data) >> 6) & 0x1FFF; } uint8_t FCI_SLI::getPicID() const { - return data & 0x3F; -} - -void FCI_SLI::net2Host() { - data = ntohl(data); + return ntohl(data) & 0x3F; } string FCI_SLI::dumpString() const { @@ -48,29 +47,52 @@ string FCI_SLI::dumpString() const { /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -void FCI_FIR::net2Host() { - ssrc = ntohl(ssrc); - reserved = ntohl(reserved) >> 8; +void FCI_FIR::check(size_t size){ + CHECK(size == kSize); +} + +uint32_t FCI_FIR::getSSRC() const{ + return ntohl(ssrc); +} + +uint8_t FCI_FIR::getSeq() const{ + return seq_number; +} + +uint32_t FCI_FIR::getReserved() const{ + return (reserved[0] << 16) | (reserved[1] << 8) | reserved[2]; } string FCI_FIR::dumpString() const { - return StrPrinter << "ssrc:" << ssrc << ", seq_number:" << seq_number << ", reserved:" << reserved; + return StrPrinter << "ssrc:" << getSSRC() << ", seq_number:" << (int)getSeq() << ", reserved:" << getReserved(); } FCI_FIR::FCI_FIR(uint32_t ssrc, uint8_t seq_number, uint32_t reserved) { this->ssrc = htonl(ssrc); this->seq_number = seq_number; - this->reserved = htonl(reserved) >> 8; + this->reserved[0] = (reserved >> 16) & 0xFF; + this->reserved[1] = (reserved >> 8) & 0xFF; + this->reserved[2] = reserved & 0xFF; } /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -string FCI_REMB::create(const initializer_list &ssrcs, uint32_t bitrate) { +static const char kRembMagic[] = "REMB"; + +void FCI_REMB::check(size_t size){ + CHECK(size >= kSize); + CHECK(memcmp(magic, kRembMagic, sizeof(magic)) == 0); + auto num_ssrc = bitrate[0]; + auto expect_size = kSize + 4 * num_ssrc; + CHECK(size == expect_size); +} + +string FCI_REMB::create(const vector &ssrcs, uint32_t bitrate) { CHECK(ssrcs.size() > 0 && ssrcs.size() <= 0xFF); string ret; ret.resize(kSize + ssrcs.size() * 4); FCI_REMB *thiz = (FCI_REMB *) ret.data(); - memcpy(thiz->magic, "REMB", 4); + memcpy(thiz->magic, kRembMagic, sizeof(magic)); /* bitrate --> BR Exp/BR Mantissa */ uint8_t b = 0; @@ -97,24 +119,12 @@ string FCI_REMB::create(const initializer_list &ssrcs, uint32_t bitrat //设置ssrc列表 auto ptr = thiz->ssrc_feedback; - for (auto &ssrc : ssrcs) { + for (auto ssrc : ssrcs) { *(ptr++) = htonl(ssrc); } return ret; } -void FCI_REMB::net2Host(size_t total_size) { - CHECK(total_size >= kSize); - CHECK(memcmp(magic, "REMB", 4) == 0); - auto num_ssrc = bitrate[0]; - auto expect_size = kSize + 4 * num_ssrc; - CHECK(total_size == expect_size); - auto ptr = ssrc_feedback; - while (num_ssrc--) { - *(ptr++) = ntohl(*ptr); - } -} - uint32_t FCI_REMB::getBitRate() const { uint8_t exp = (bitrate[1] >> 2) & 0x3F; uint32_t mantissa = (bitrate[1] & 0x03) << 16; @@ -123,12 +133,12 @@ uint32_t FCI_REMB::getBitRate() const { return mantissa << exp; } -vector FCI_REMB::getSSRC() { - vector ret; +vector FCI_REMB::getSSRC() { + vector ret; auto num_ssrc = bitrate[0]; auto ptr = ssrc_feedback; while (num_ssrc--) { - ret.emplace_back(ptr++); + ret.emplace_back(ntohl(*ptr++)); } return ret; } @@ -137,30 +147,51 @@ string FCI_REMB::dumpString() const { _StrPrinter printer; printer << "bitrate:" << getBitRate() << ", ssrc:"; for (auto &ssrc : ((FCI_REMB *) this)->getSSRC()) { - printer << *ssrc << " "; + printer << ssrc << " "; } return printer; } /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// -void FCI_NACK::net2Host() { - pid = ntohs(pid); - blp = ntohs(blp); +FCI_NACK::FCI_NACK(uint16_t pid_h, const vector &type) { + uint16_t blp_h = 0; + int i = kBitSize; + for (auto item : type) { + --i; + if (item) { + blp_h |= (1 << i); + } + } + blp = htons(blp_h); + pid = htons(pid_h); +} + +void FCI_NACK::check(size_t size){ + CHECK(size == kSize); +} + +uint16_t FCI_NACK::getPid() const { + return ntohs(pid); +} + +uint16_t FCI_NACK::getBlp() const { + return ntohs(blp); } vector FCI_NACK::getBitArray() const { vector ret; ret.resize(kBitSize); + auto blp_h = getBlp(); for (size_t i = 0; i < kBitSize; ++i) { - ret[i] = blp & (1 << (kBitSize - i - 1)); + ret[i] = blp_h & (1 << (kBitSize - i - 1)); } return ret; } string FCI_NACK::dumpString() const { _StrPrinter printer; - printer << "pid:" << pid << ",blp:"; + printer << "pid:" << getPid() << ",blp:"; for (auto &flag : getBitArray()) { printer << flag << " "; } @@ -169,6 +200,37 @@ string FCI_NACK::dumpString() const { /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +class RunLengthChunk { +public: + static size_t constexpr kSize = 2; + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T| S | Run Length | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +#if __BYTE_ORDER == __BIG_ENDIAN + uint16_t type: 1; + uint16_t symbol: 2; + uint16_t run_length_high: 5; +#else + // Run Length 高5位 + uint16_t run_length_high: 5; + //参考SymbolStatus定义 + uint16_t symbol: 2; + //固定为0 + uint16_t type: 1; +#endif + // Run Length 低8位 + uint16_t run_length_low: 8; + + //获取Run Length + uint16_t getRunLength() const; + //构造函数 + RunLengthChunk(SymbolStatus status, uint16_t run_length); + //打印本对象 + string dumpString() const; +} PACKED; + RunLengthChunk::RunLengthChunk(SymbolStatus status, uint16_t run_length) { type = 0; symbol = (uint8_t)status & 0x03; @@ -189,6 +251,37 @@ string RunLengthChunk::dumpString() const{ /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +class StatusVecChunk { +public: + static size_t constexpr kSize = 2; + // 0 1 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |T|S| symbol list | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +#if __BYTE_ORDER == __BIG_ENDIAN + uint16_t type: 1; + uint16_t symbol: 1; + uint16_t symbol_list_high: 6; +#else + // symbol_list 高6位 + uint16_t symbol_list_high: 6; + //symbol_list中元素是1个还是2个bit + uint16_t symbol: 1; + //固定为1 + uint16_t type: 1; +#endif + // symbol_list 低8位 + uint16_t symbol_list_low: 8; + + //获取symbollist + vector getSymbolList() const; + //构造函数 + StatusVecChunk(const vector &status); + //打印本对象 + string dumpString() const; +} PACKED; + StatusVecChunk::StatusVecChunk(const vector &status) { uint16_t value = 0; type = 1; @@ -248,10 +341,16 @@ string StatusVecChunk::dumpString() const { /////////////////////////////////////////////////////// -void FCI_TWCC::net2Host(size_t total_size) { - CHECK(total_size >= kSize); - base_seq = ntohs(base_seq); - pkt_status_count = ntohs(pkt_status_count); +void FCI_TWCC::check(size_t size){ + CHECK(size >= kSize); +} + +uint16_t FCI_TWCC::getBaseSeq() const { + return ntohs(base_seq); +} + +uint16_t FCI_TWCC::getPacketCount() const { + return ntohs(pkt_status_count); } uint32_t FCI_TWCC::getReferenceTime() const { @@ -323,10 +422,10 @@ map > FCI_TWCC::getPacketCh auto ptr = (uint8_t *) this + kSize; auto end = (uint8_t *) this + total_size; CHECK(ptr < end); - auto seq = base_seq; + auto seq = getBaseSeq(); - for (uint8_t i = 0; i < pkt_status_count;) { - CHECK(ptr + RunLengthChunk::kSize <= end) + for (uint8_t i = 0; i < getPacketCount();) { + CHECK(ptr + RunLengthChunk::kSize <= end); RunLengthChunk *chunk = (RunLengthChunk *) ptr; if (!chunk->type) { //RunLengthChunk @@ -345,7 +444,7 @@ map > FCI_TWCC::getPacketCh ptr += 2; } for (auto &pr : ret) { - CHECK(ptr <= end) + CHECK(ptr <= end); pr.second.second = 250 * getRecvDelta(pr.second.first, ptr, end); } return ret; @@ -354,11 +453,55 @@ map > FCI_TWCC::getPacketCh string FCI_TWCC::dumpString(size_t total_size) const { _StrPrinter printer; auto map = getPacketChunkList(total_size); - printer << "twcc fci, base_seq:" << base_seq << ",pkt_status_count:" << pkt_status_count << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n"; + printer << "twcc fci, base_seq:" << getBaseSeq() << ",pkt_status_count:" << getPacketCount() << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n"; for (auto &pr : map) { printer << "rtp seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", delta:" << pr.second.second << "\n"; } return std::move(printer); } -}//namespace mediakit \ No newline at end of file +}//namespace mediakit + +#if 1 +using namespace mediakit; +void testFCI() { + { + FCI_SLI fci(8191, 0, 63); + InfoL << hexdump(&fci, FCI_SLI::kSize) << fci.dumpString(); + } + { + FCI_FIR fci(123456, 139, 456789); + InfoL << hexdump(&fci, FCI_FIR::kSize) << fci.dumpString(); + } + { + auto str = FCI_REMB::create({1234, 2345, 5678}, 4 * 1024 * 1024); + FCI_REMB *ptr = (FCI_REMB *) str.data(); + InfoL << hexdump(str.data(), str.size()) << ptr->dumpString(); + } + { + FCI_NACK nack(1234, vector({1, 0, 0, 0, 1, 0, 1, 0, 1, 0})); + InfoL << hexdump(&nack, FCI_NACK::kSize) << nack.dumpString(); + } + + { + RunLengthChunk chunk(SymbolStatus::large_delta, 8024); + InfoL << hexdump(&chunk, RunLengthChunk::kSize) << chunk.dumpString(); + } + + auto lam = [](const initializer_list &lst) { + vector ret; + for (auto &num : lst) { + ret.emplace_back((SymbolStatus) num); + } + return ret; + }; + { + StatusVecChunk chunk(lam({0, 1, 0, 1, 0, 1, 0, 0, 0, 1, 1, 1, 0, 1})); + InfoL << hexdump(&chunk, StatusVecChunk::kSize) << chunk.dumpString(); + } + { + StatusVecChunk chunk(lam({0, 1, 2, 2, 0, 1, 2})); + InfoL << hexdump(&chunk, StatusVecChunk::kSize) << chunk.dumpString(); + } +} +#endif \ No newline at end of file diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index 6676c42c..a3d5e96e 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -12,6 +12,7 @@ #define ZLMEDIAKIT_RTCPFCI_H #include "Rtcp.h" +#include "assert.h" namespace mediakit { @@ -46,10 +47,11 @@ public: static size_t constexpr kSize = 4; FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id); + + void check(size_t size); uint16_t getFirst() const; uint16_t getNumber() const; uint8_t getPicID() const; - void net2Host(); string dumpString() const; private: @@ -108,15 +110,20 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_FIR { public: - uint32_t ssrc; - uint32_t seq_number: 8; - uint32_t reserved: 24; - static size_t constexpr kSize = 8; + FCI_FIR(uint32_t ssrc, uint8_t seq_number, uint32_t reserved = 0); - void net2Host(); + + void check(size_t size); + uint32_t getSSRC() const; + uint8_t getSeq() const; + uint32_t getReserved() const; string dumpString() const; +private: + uint32_t ssrc; + uint8_t seq_number; + uint8_t reserved[3]; } PACKED; //PSFB fmt = 5 @@ -131,6 +138,13 @@ public: class FCI_TSTR { public: static size_t constexpr kSize = 8; + + void check(size_t size) { + CHECK(size == kSize); + } + +private: + uint8_t data[kSize]; } PACKED; //PSFB fmt = 6 @@ -160,6 +174,13 @@ class FCI_TSTN : public FCI_TSTR{ class FCI_VBCM { public: static size_t constexpr kSize = 12; + + void check(size_t size) { + CHECK(size == kSize); + } + +private: + uint8_t data[kSize]; } PACKED; //PSFB fmt = 15 @@ -194,12 +215,13 @@ class FCI_REMB { public: static size_t constexpr kSize = 8; - static string create(const std::initializer_list &ssrcs, uint32_t bitrate); - void net2Host(size_t total_size); + static string create(const std::vector &ssrcs, uint32_t bitrate); + void check(size_t size); string dumpString() const; uint32_t getBitRate() const; - vector getSSRC(); + vector getSSRC(); +private: //Unique identifier 'R' 'E' 'M' 'B' char magic[4]; //Num SSRC (8 bits)/BR Exp (6 bits)/ BR Mantissa (18 bits) @@ -207,7 +229,6 @@ public: // SSRC feedback (32 bits) Consists of one or more SSRC entries which // this feedback message applies to. uint32_t ssrc_feedback[1]; - } PACKED; /////////////////////////////////////////// RTPFB //////////////////////////////////////////////////// @@ -221,32 +242,25 @@ public: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ class FCI_NACK { public: - static constexpr size_t kBitSize = 16; static constexpr size_t kSize = 4; + FCI_NACK(uint16_t pid_h, const vector &type); + + void check(size_t size); + uint16_t getPid() const; + uint16_t getBlp() const; + vector getBitArray() const; + string dumpString() const; + +private: + static constexpr size_t kBitSize = 16; + +private: // The PID field is used to specify a lost packet. The PID field // refers to the RTP sequence number of the lost packet. uint16_t pid; // bitmask of following lost packets (BLP): 16 bits uint16_t blp; - - void net2Host(); - vector getBitArray() const; - string dumpString() const; - - template - FCI_NACK(uint16_t pid_h, const Type &type){ - uint16_t blp_h = 0; - int i = kBitSize; - for (auto &item : type) { - --i; - if (item) { - blp_h |= (1 << i); - } - } - blp = htons(blp_h); - pid = htons(pid_h); - } } PACKED; //RTPFB fmt = 3 @@ -262,6 +276,11 @@ class FCI_TMMBR { public: static size_t constexpr kSize = 8; + void check(size_t size) { + CHECK(size == kSize); + } + +private: //SSRC (32 bits): The SSRC value of the media sender that is // requested to obey the new maximum bit rate. uint32_t ssrc; @@ -269,17 +288,13 @@ public: // MxTBR Exp (6 bits): The exponential scaling of the mantissa for the // maximum total media bit rate value. The value is an // unsigned integer [0..63]. - uint32_t max_tbr_exp: 6; - // MxTBR Mantissa (17 bits): The mantissa of the maximum total media // bit rate value as an unsigned integer. - uint32_t max_mantissa: 17; - // Measured Overhead (9 bits): The measured average packet overhead // value in bytes. The measurement SHALL be done according // to the description in section 4.2.1.2. The value is an // unsigned integer [0..511]. - uint32_t measured_overhead: 9; + uint32_t max_tbr; } PACKED; //RTPFB fmt = 4 @@ -307,68 +322,6 @@ enum class SymbolStatus : uint8_t{ reserved = 3 }; -class RunLengthChunk { -public: - static size_t constexpr kSize = 2; - // 0 1 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // |T| S | Run Length | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -#if __BYTE_ORDER == __BIG_ENDIAN - uint16_t type: 1; - uint16_t symbol: 2; - uint16_t run_length_high: 5; -#else - // Run Length 高5位 - uint16_t run_length_high: 5; - //参考SymbolStatus定义 - uint16_t symbol: 2; - //固定为0 - uint16_t type: 1; -#endif - // Run Length 低8位 - uint16_t run_length_low: 8; - - //获取Run Length - uint16_t getRunLength() const; - //构造函数 - RunLengthChunk(SymbolStatus status, uint16_t run_length); - - string dumpString() const; -} PACKED; - -class StatusVecChunk { -public: - static size_t constexpr kSize = 2; - // 0 1 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // |T|S| symbol list | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -#if __BYTE_ORDER == __BIG_ENDIAN - uint16_t type: 1; - uint16_t symbol: 1; - uint16_t symbol_list_high: 6; -#else - // symbol_list 高6位 - uint16_t symbol_list_high: 6; - //symbol_list中元素是1个还是2个bit - uint16_t symbol: 1; - //固定为1 - uint16_t type: 1; -#endif - // symbol_list 低8位 - uint16_t symbol_list_low: 8; - - //获取symbollist - vector getSymbolList() const; - //构造函数 - StatusVecChunk(const vector &status); - - string dumpString() const; -} PACKED; - //RTPFB fmt = 15 //https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-3.1 //https://zhuanlan.zhihu.com/p/206656654 @@ -395,6 +348,14 @@ class FCI_TWCC{ public: static size_t constexpr kSize = 8; + void check(size_t size); + string dumpString(size_t total_size) const; + uint16_t getBaseSeq() const; + uint32_t getReferenceTime() const; + uint16_t getPacketCount() const; + map > getPacketChunkList(size_t total_size) const; + +private: //base sequence number,基础序号,本次反馈的第一个包的序号;也就是RTP扩展头的序列号 uint16_t base_seq; //packet status count, 包个数,本次反馈包含多少个包的状态;从基础序号开始算 @@ -403,12 +364,6 @@ public: uint8_t ref_time[3]; //feedback packet count,反馈包号,本包是第几个transport-cc包，每次加1 | uint8_t fb_pkt_count; - - void net2Host(size_t total_size); - uint32_t getReferenceTime() const; - map > getPacketChunkList(size_t total_size) const; - string dumpString(size_t total_size) const; - } PACKED; } //namespace mediakit diff --git a/tests/test_rtcp_fci.cpp b/tests/test_rtcp_fci.cpp index aecc821d..1dda1143 100644 --- a/tests/test_rtcp_fci.cpp +++ b/tests/test_rtcp_fci.cpp @@ -15,55 +15,11 @@ using namespace std; using namespace toolkit; using namespace mediakit; +extern void testFCI(); + int main() { - //初始化日志系统 - Logger::Instance().add(std::make_shared ()); + Logger::Instance().add(std::make_shared()); - { - FCI_SLI fci(0xFFFF, 0, 0xFF); - InfoL << 0b10101010101 << " " << 0b01010101010 << " " << (int) 0b101010 << " " << hexdump(&fci, FCI_SLI::kSize); - fci.net2Host(); - InfoL << fci.dumpString(); - } - { - FCI_FIR fci(123456, 139, 456789); - InfoL << hexdump(&fci, FCI_FIR::kSize); - fci.net2Host(); - InfoL << fci.dumpString(); - } - { - auto str = FCI_REMB::create({1234,2345,5678}, 4 * 1024 * 1024); - FCI_REMB *ptr = (FCI_REMB *)str.data(); - ptr->net2Host(str.size()); - InfoL << ptr->dumpString(); - } - { - FCI_NACK nack(1234, vector({1, 0, 0, 0, 1, 0, 1, 0, 1, 0})); - nack.net2Host(); - InfoL << nack.dumpString(); - } - { - RunLengthChunk chunk(SymbolStatus::large_delta, 8024); - InfoL << hexdump(&chunk, RunLengthChunk::kSize); - InfoL << chunk.dumpString(); - } - - auto lam = [](const initializer_list &lst){ - vector ret; - for(auto &num : lst){ - ret.emplace_back((SymbolStatus)num); - } - return ret; - }; - { - StatusVecChunk chunk(lam({0, 1, 0, 1, 0, 1, 0, 0, 0, 1, 1, 1, 0, 1})); - InfoL << hexdump(&chunk, StatusVecChunk::kSize); - InfoL << chunk.dumpString(); - } - { - StatusVecChunk chunk(lam({0, 1, 2, 3, 0, 1, 2})); - InfoL << hexdump(&chunk, StatusVecChunk::kSize); - InfoL << chunk.dumpString(); - } + testFCI(); return 0; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 272d6dd9..6a20fc40 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -19,8 +19,6 @@ using namespace std; using namespace mediakit; -#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__); - //https://datatracker.ietf.org/doc/rfc4566/?include_text=1 //https://blog.csdn.net/aggresss/article/details/109850434 //https://aggresss.blog.csdn.net/article/details/106436703 diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index d37e5a88..800daa1d 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -557,7 +557,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { FCI_TWCC *twcc = (FCI_TWCC *) (fci); auto fci_size = rtpfb->getSize() - 12; InfoL << hexdump(fci, fci_size); - twcc->net2Host(fci_size); InfoL << "\n" << twcc->dumpString(fci_size); } break; From 93160c0ec7173cf6dfd55b8f562a3d52cdf0c78e Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 23 Apr 2021 15:25:33 +0800 Subject: [PATCH 115/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtpfb=E6=9E=9A?= =?UTF-8?q?=E4=B8=BE=E4=B8=8E=E5=B7=A5=E5=85=B7=E7=B1=BB?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 9 +++++++++ src/Rtcp/Rtcp.h | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 35853e80..33ac59e3 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -42,6 +42,15 @@ const char *psfbTypeToStr(PSFBType type) { } } +const char *rtpfbTypeToStr(RTPFBType type) { + switch (type){ +#define SWITCH_CASE(key, value) case RTPFBType::key : return #value "(" #key ")"; + RTPFB_TYPE_MAP(SWITCH_CASE) +#undef SWITCH_CASE + default: return "unknown transport layer feedback messages fmt type"; + } +} + static size_t alignSize(size_t bytes) { return (size_t)((bytes + 3) / 4) << 2; } diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index 88fa5475..838ddf40 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -133,6 +133,13 @@ enum class PSFBType : uint8_t { #undef XX }; +//rtpfb类型枚举 +enum class RTPFBType : uint8_t { +#define XX(key, value) key = value, + RTPFB_TYPE_MAP(XX) +#undef XX +}; + /** * RtcpType转描述字符串 */ @@ -148,6 +155,11 @@ const char *sdesTypeToStr(SdesType type); */ const char *psfbTypeToStr(PSFBType type); +/** + * rtpfb枚举转描述字符串 + */ +const char *rtpfbTypeToStr(RTPFBType type); + class RtcpHeader { public: #if __BYTE_ORDER == __BIG_ENDIAN From caceb90b40d822c08a911352532d65be32df2e8c Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 23 Apr 2021 16:37:08 +0800 Subject: [PATCH 116/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E9=94=99=E8=AF=AF?= =?UTF-8?q?=E6=8B=BC=E5=86=99=E3=80=81=E5=8A=A0=E5=BF=AB=E6=B5=81=E6=B3=A8?= =?UTF-8?q?=E5=86=8C=E9=80=9F=E5=BA=A6?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- api/tests/pusher.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/api/tests/pusher.c b/api/tests/pusher.c index 3c5d580b..76ca4297 100644 --- a/api/tests/pusher.c +++ b/api/tests/pusher.c @@ -91,7 +91,7 @@ void API_CALL on_mk_play_event_func(void *user_data, int err_code, const char *e if (err_code == 0) { //success log_debug("play success!"); - ctx->media = mk_media_create("__defaultVost__", "live", "test", 0, 0, 0); + ctx->media = mk_media_create("__defaultVhost__", "live", "test", 0, 0, 0); int video_codec = mk_player_video_codecId(ctx->player); int audio_codec = mk_player_audio_codecId(ctx->player); @@ -108,6 +108,7 @@ void API_CALL on_mk_play_event_func(void *user_data, int err_code, const char *e mk_player_audio_channel(ctx->player), mk_player_audio_bit(ctx->player)); } + mk_media_init_complete(ctx->media); mk_media_set_on_regist(ctx->media, on_mk_media_source_regist_func, ctx); } else { From 8702ad101b99138784b4d6d40a7947fec2b39e6c Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 23 Apr 2021 18:27:47 +0800 Subject: [PATCH 117/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtcp=20feedback?= =?UTF-8?q?=E7=9B=B8=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 125 ++++++++++++++++++++++--------------- src/Rtcp/Rtcp.h | 33 ++++++---- webrtc/WebRtcTransport.cpp | 20 ++---- 3 files changed, 99 insertions(+), 79 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 33ac59e3..e3c95927 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -52,7 +52,7 @@ const char *rtpfbTypeToStr(RTPFBType type) { } static size_t alignSize(size_t bytes) { - return (size_t)((bytes + 3) / 4) << 2; + return (size_t)((bytes + 3) >> 2 ) << 2; } static void setupHeader(RtcpHeader *rtcp, RtcpType type, size_t report_count, size_t total_bytes) { @@ -64,23 +64,32 @@ static void setupHeader(RtcpHeader *rtcp, RtcpType type, size_t report_count, si //items总个数 rtcp->report_count = report_count; rtcp->pt = (uint8_t) type; - //不包含rtcp头的长度 - rtcp->length = htons((uint16_t)((total_bytes / 4) - 1)); + rtcp->setSize(total_bytes); } ///////////////////////////////////////////////////////////////////////////// -void RtcpHeader::net2Host() { - length = ntohs(length); -} - string RtcpHeader::dumpHeader() const{ _StrPrinter printer; printer << "version:" << version << "\r\n"; printer << "padding:" << padding << "\r\n"; - printer << "report_count:" << report_count << "\r\n"; + switch ((RtcpType)pt) { + case RtcpType::RTCP_RTPFB : { + printer << "report_count:" << rtpfbTypeToStr((RTPFBType) report_count) << "\r\n"; + break; + } + case RtcpType::RTCP_PSFB : { + printer << "report_count:" << psfbTypeToStr((PSFBType) report_count) << "\r\n"; + break; + } + default : { + printer << "report_count:" << report_count << "\r\n"; + break; + } + } + printer << "pt:" << rtcpTypeToStr((RtcpType)pt) << "\r\n"; - printer << "length:" << length << "\r\n"; + printer << "size:" << getSize() << "\r\n"; printer << "--------\r\n"; return std::move(printer); } @@ -102,8 +111,9 @@ string RtcpHeader::dumpString() const { return rtcp->dumpString(); } + case RtcpType::RTCP_RTPFB: case RtcpType::RTCP_PSFB: { - RtcpPli *rtcp = (RtcpPli *)this; + RtcpFB *rtcp = (RtcpFB *)this; return rtcp->dumpString(); } @@ -112,12 +122,18 @@ string RtcpHeader::dumpString() const { return rtcp->dumpString(); } - default: return StrPrinter << dumpHeader() << hexdump((char *)this + sizeof(*this), length << 2); + default: return StrPrinter << dumpHeader() << hexdump((char *)this + sizeof(*this), getSize() - sizeof(*this)); } } size_t RtcpHeader::getSize() const { - return (1 + length) << 2; + //加上rtcp头长度 + return (1 + ntohs(length)) << 2; +} + +void RtcpHeader::setSize(size_t size) { + //不包含rtcp头的长度 + length = htons((uint16_t)((size >> 2) - 1)); } void RtcpHeader::net2Host(size_t len){ @@ -140,9 +156,10 @@ void RtcpHeader::net2Host(size_t len){ break; } + case RtcpType::RTCP_RTPFB: case RtcpType::RTCP_PSFB: { - RtcpPli *pli = (RtcpPli *)this; - pli->net2Host(len); + RtcpFB *fb = (RtcpFB *)this; + fb->net2Host(len); break; } @@ -152,11 +169,6 @@ void RtcpHeader::net2Host(size_t len){ break; } - case RtcpType::RTCP_RTPFB: { - RtcpPli *pli = (RtcpPli *)this; - pli->net2Host(len); - break; - } default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt)); } } @@ -167,8 +179,8 @@ vector RtcpHeader::loadFromBytes(char *data, size_t len){ char *ptr = data; while (remain > (ssize_t) sizeof(RtcpHeader)) { RtcpHeader *rtcp = (RtcpHeader *) ptr; - auto rtcp_len = (1 + ntohs(rtcp->length)) << 2; - if (remain < rtcp_len) { + auto rtcp_len = rtcp->getSize(); + if (remain < (ssize_t)rtcp_len) { WarnL << "非法的rtcp包,声明的长度超过实际数据长度"; break; } @@ -189,7 +201,6 @@ class BufferRtcp : public Buffer { public: BufferRtcp(std::shared_ptr rtcp) { _rtcp = std::move(rtcp); - _size = (ntohs(_rtcp->length) + 1) << 2; } ~BufferRtcp() override {} @@ -199,7 +210,7 @@ public: } size_t size() const override { - return _size; + return _rtcp->getSize(); } private: @@ -258,21 +269,17 @@ if (size < kMinSize) { \ throw std::out_of_range(StrPrinter << rtcpTypeToStr((RtcpType)pt) << " 长度不足:" << size << " < " << kMinSize); \ } -#define CHECK_LENGTH(size, item_count) \ +#define CHECK_REPORT_COUNT(item_count) \ /*修正个数，防止getItemList时内存越界*/ \ if (report_count != item_count) { \ WarnL << rtcpTypeToStr((RtcpType)pt) << " report_count 字段不正确,已修正为:" << (int)report_count << " -> " << item_count; \ report_count = item_count; \ -} \ -if ((size_t) (length + 1) << 2 != size) { \ - WarnL << rtcpTypeToStr((RtcpType)pt) << " length字段不正确:" << (size_t) (length + 1) << 2 << " != " << size; \ } void RtcpSR::net2Host(size_t size) { static const size_t kMinSize = sizeof(RtcpSR) - sizeof(items); CHECK_MIN_SIZE(size, kMinSize); - RtcpHeader::net2Host(); ssrc = ntohl(ssrc); ntpmsw = ntohl(ntpmsw); ntplsw = ntohl(ntplsw); @@ -287,7 +294,7 @@ void RtcpSR::net2Host(size_t size) { ++ptr; ++item_count; } - CHECK_LENGTH(size, item_count); + CHECK_REPORT_COUNT(item_count); } vector RtcpSR::getItemList(){ @@ -352,7 +359,6 @@ string RtcpRR::dumpString() const{ void RtcpRR::net2Host(size_t size) { static const size_t kMinSize = sizeof(RtcpRR) - sizeof(items); CHECK_MIN_SIZE(size, kMinSize); - RtcpHeader::net2Host(); ssrc = ntohl(ssrc); ReportItem *ptr = &items; @@ -362,7 +368,7 @@ void RtcpRR::net2Host(size_t size) { ++ptr; ++item_count; } - CHECK_LENGTH(size, item_count); + CHECK_REPORT_COUNT(item_count); } vector RtcpRR::getItemList() { @@ -382,7 +388,7 @@ void SdesItem::net2Host() { } size_t SdesItem::totalBytes() const{ - return alignSize(minSize() + length); + return alignSize(minSize() + txt_len); } size_t SdesItem::minSize() { @@ -393,14 +399,14 @@ string SdesItem::dumpString() const{ _StrPrinter printer; printer << "ssrc:" << ssrc << "\r\n"; printer << "type:" << sdesTypeToStr((SdesType) type) << "\r\n"; - printer << "length:" << (int) length << "\r\n"; - printer << "text:" << (length ? string(text, length) : "") << "\r\n"; + printer << "txt_len:" << (int) txt_len << "\r\n"; + printer << "text:" << (txt_len ? string(text, txt_len) : "") << "\r\n"; return std::move(printer); } ///////////////////////////////////////////////////////////////////////////// -std::shared_ptr RtcpSdes::create(const std::initializer_list &item_text) { +std::shared_ptr RtcpSdes::create(const std::vector &item_text) { size_t item_total_size = 0; for (auto &text : item_text) { //统计所有SdesItem对象占用的空间 @@ -410,9 +416,9 @@ std::shared_ptr RtcpSdes::create(const std::initializer_list & auto ptr = (RtcpSdes *) new char[bytes]; auto item_ptr = &ptr->items; for (auto &text : item_text) { - item_ptr->length = (0xFF & text.size()); + item_ptr->txt_len = (0xFF & text.size()); //确保赋值\0为RTCP_SDES_END - memcpy(item_ptr->text, text.data(), item_ptr->length + 1); + memcpy(item_ptr->text, text.data(), item_ptr->txt_len + 1); item_ptr = (SdesItem *) ((char *) item_ptr + item_ptr->totalBytes()); } @@ -437,7 +443,6 @@ string RtcpSdes::dumpString() const { void RtcpSdes::net2Host(size_t size) { static const size_t kMinSize = sizeof(RtcpSdes) - sizeof(items); CHECK_MIN_SIZE(size, kMinSize); - RtcpHeader::net2Host(); SdesItem *ptr = &items; int item_count = 0; for(int i = 0; i < (int)report_count && (char *)(ptr) + SdesItem::minSize() <= (char *)(this) + size; ++i){ @@ -445,7 +450,7 @@ void RtcpSdes::net2Host(size_t size) { ptr = (SdesItem *) ((char *) ptr + ptr->totalBytes()); ++item_count; } - CHECK_LENGTH(size, item_count); + CHECK_REPORT_COUNT(item_count); } vector RtcpSdes::getItemList() { @@ -460,34 +465,52 @@ vector RtcpSdes::getItemList() { //////////////////////////////////////////////////////////////////// -std::shared_ptr RtcpPli::create() { - auto bytes = alignSize(sizeof(RtcpPli)); +std::shared_ptr RtcpFB::create_l(RtcpType type, int fmt, const void *fci, size_t fci_len) { + if (!fci) { + fci_len = 0; + } + auto bytes = alignSize(sizeof(RtcpFB) + fci_len); auto ptr = (RtcpRR *) new char[bytes]; - setupHeader(ptr, RtcpType::RTCP_PSFB, 1, bytes); - return std::shared_ptr((RtcpPli *) ptr, [](RtcpPli *ptr) { + if (fci && fci_len) { + memcpy(ptr + sizeof(RtcpFB), fci, fci_len); + } + setupHeader(ptr, type, fmt, bytes); + return std::shared_ptr((RtcpFB *) ptr, [](RtcpFB *ptr) { delete[] (char *) ptr; }); } -string RtcpPli::dumpString() const { +std::shared_ptr RtcpFB::create(PSFBType fmt, const void *fci, size_t fci_len) { + return RtcpFB::create_l(RtcpType::RTCP_PSFB, (int)fmt, fci, fci_len); +} + +std::shared_ptr RtcpFB::create(RTPFBType fmt, const void *fci, size_t fci_len) { + return RtcpFB::create_l(RtcpType::RTCP_RTPFB, (int)fmt, fci, fci_len); +} + +string RtcpFB::dumpString() const { _StrPrinter printer; printer << RtcpHeader::dumpHeader(); printer << "ssrc:" << ssrc << "\r\n"; - printer << "ssrc_media:" << ssrc_media; + printer << "ssrc_media:" << ssrc_media << "\r\n"; + auto fci = (uint8_t *)&ssrc_media + sizeof(ssrc_media); + auto fci_len = getSize() - sizeof(RtcpFB); + if (fci_len) { + printer << "fci:" << hexdump(fci, fci_len); + } return std::move(printer); } -void RtcpPli::net2Host(size_t size) { - static const size_t kMinSize = sizeof(RtcpPli); +void RtcpFB::net2Host(size_t size) { + static const size_t kMinSize = sizeof(RtcpFB); CHECK_MIN_SIZE(size, kMinSize); - RtcpHeader::net2Host(); ssrc = ntohl(ssrc); ssrc_media = ntohl(ssrc_media); } //////////////////////////////////////////////////////////////////// -std::shared_ptr RtcpBye::create(const std::initializer_list &ssrcs, const string &reason) { +std::shared_ptr RtcpBye::create(const std::vector &ssrcs, const string &reason) { assert(reason.size() <= 0xFF); auto bytes = alignSize(sizeof(RtcpHeader) + sizeof(uint32_t) * ssrcs.size() + 1 + reason.size()); auto ptr = (RtcpBye *) new char[bytes]; @@ -541,8 +564,6 @@ string RtcpBye::dumpString() const { void RtcpBye::net2Host(size_t size) { static const size_t kMinSize = sizeof(RtcpHeader); CHECK_MIN_SIZE(size, kMinSize); - RtcpHeader::net2Host(); - uint32_t *ssrc_ptr = &ssrc; size_t offset = kMinSize; size_t i = 0; @@ -552,7 +573,7 @@ void RtcpBye::net2Host(size_t size) { offset += sizeof(ssrc); } //修正ssrc个数 - CHECK_LENGTH(size, i); + CHECK_REPORT_COUNT(i); if (offset < size) { uint8_t *reason_len_ptr = &reason_len + sizeof(ssrc) * (report_count - 1); diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index 838ddf40..ddc5ae71 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -179,6 +179,8 @@ public: #endif //rtcp类型,RtcpType uint32_t pt: 8; + +private: //长度 uint32_t length: 16; @@ -207,15 +209,16 @@ public: /** * 根据length字段获取rtcp总长度 - * 使用net2Host转换成主机字节序后才可使用此函数 */ size_t getSize() const; -protected: /** - * 网络字节序转换为主机字节序 + * 设置rtcp length字段 + * @param size rtcp总长度，单位字节 */ - void net2Host(); + void setSize(size_t size); + +protected: /** * 打印字段详情 @@ -476,7 +479,7 @@ public: //SdesType uint8_t type; //text长度股，可以为0 - uint8_t length; + uint8_t txt_len; //不定长 char text[1]; //最后以RTCP_SDES_END结尾 @@ -521,7 +524,7 @@ public: * @param item_text SdesItem列表，只赋值length和text部分 * @return SDES包 */ - static std::shared_ptr create(const std::initializer_list &item_text); + static std::shared_ptr create(const std::vector &item_text); /** * 获取SdesItem对象指针列表 @@ -560,8 +563,8 @@ private: // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // : Feedback Control Information (FCI) : // : : - -class RtcpPli : public RtcpHeader { +// rtcpfb和psfb的数据结构一致 +class RtcpFB : public RtcpHeader { public: friend class RtcpHeader; uint32_t ssrc; @@ -569,9 +572,14 @@ public: public: /** - * 创建pli包 + * 创建psfb类型的反馈包 */ - static std::shared_ptr create(); + static std::shared_ptr create(PSFBType fmt, const void *fci = nullptr, size_t fci_len = 0); + + /** + * 创建rtpfb类型的反馈包 + */ + static std::shared_ptr create(RTPFBType fmt, const void *fci = nullptr, size_t fci_len = 0); private: /** @@ -585,6 +593,9 @@ private: * @param size 字节长度，防止内存越界 */ void net2Host(size_t size); + +private: + static std::shared_ptr create_l(RtcpType type, int fmt, const void *fci, size_t fci_len); } PACKED; //BYE @@ -621,7 +632,7 @@ public: * @param reason 原因 * @return rtcp bye包 */ - static std::shared_ptr create(const std::initializer_list &ssrc, const string &reason); + static std::shared_ptr create(const std::vector &ssrc, const string &reason); /** * 获取ssrc列表 diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 800daa1d..aeb8a1ec 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -544,21 +544,9 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { onShutdown(SockException(Err_eof, "rtcp bye message received")); break; } - case RtcpType::RTCP_PSFB: { - //todo 支持pli等更多类型的rtcp - break; - } + case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - //todo 测试打印twcc - RtcpPli *rtpfb = (RtcpPli *)rtcp; - auto fci = (uint8_t *)rtpfb + sizeof (RtcpPli); - if(rtpfb->report_count == 15){ - //TWCC - FCI_TWCC *twcc = (FCI_TWCC *) (fci); - auto fci_size = rtpfb->getSize() - 12; - InfoL << hexdump(fci, fci_size); - InfoL << "\n" << twcc->dumpString(fci_size); - } + InfoL << rtcp->dumpString(); break; } default: break; @@ -591,10 +579,10 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr if (_pli_ticker.elapsedTime() > 2000) { //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); - auto pli = RtcpPli::create(); + auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI); pli->ssrc = htonl(0); pli->ssrc_media = htonl(_recv_video_ssrc); - sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true); + sendRtcpPacket((char *) pli.get(), pli->getSize(), true); } if (_push_src) { _push_src->onWrite(std::move(rtp), false); From fbc8c2286e30191213941979bb726541b3a56408 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 23 Apr 2021 18:30:37 +0800 Subject: [PATCH 118/218] =?UTF-8?q?=E7=AD=89=E5=BE=85=E6=B7=BB=E5=8A=A0fci?= =?UTF-8?q?=E7=9B=B8=E5=85=B3=E9=80=82=E9=85=8D=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index e3c95927..c2c2d920 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -496,6 +496,14 @@ string RtcpFB::dumpString() const { auto fci = (uint8_t *)&ssrc_media + sizeof(ssrc_media); auto fci_len = getSize() - sizeof(RtcpFB); if (fci_len) { + switch ((RtcpType) pt) { + case RtcpType::RTCP_PSFB : { + break; + } + case RtcpType::RTCP_RTPFB : { + break; + } + } printer << "fci:" << hexdump(fci, fci_len); } return std::move(printer); From aa54adb1ab83581d291fd6749dcedd08b593788a Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 26 Apr 2021 20:29:06 +0800 Subject: [PATCH 119/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E7=BC=96=E8=AF=91?= =?UTF-8?q?=E9=97=AE=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 2 +- webrtc/WebRtcTransport.cpp | 2 -- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 79e09c25..28ed2945 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -192,7 +192,7 @@ vector FCI_NACK::getBitArray() const { string FCI_NACK::dumpString() const { _StrPrinter printer; printer << "pid:" << getPid() << ",blp:"; - for (auto &flag : getBitArray()) { + for (auto flag : getBitArray()) { printer << flag << " "; } return std::move(printer); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index aeb8a1ec..d4392385 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -501,8 +501,6 @@ private: function _on_before_sort; }; -#include "RTCP/RtcpFCI.h" - void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { _bytes_usage += len; auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len); From 2d8ef45e4d1ac44333bfd64c525485e9fb526363 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 26 Apr 2021 21:03:04 +0800 Subject: [PATCH 120/218] =?UTF-8?q?=E5=AE=8C=E5=96=84fci=E7=9B=B8=E5=85=B3?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 66 ++++++++++++++++++++++++++++++++------ src/Rtcp/Rtcp.h | 2 +- src/Rtcp/RtcpFCI.cpp | 2 +- webrtc/Sdp.cpp | 23 +++++++++---- webrtc/WebRtcTransport.cpp | 2 +- webrtc/readme.md | 48 +++++++-------------------- 6 files changed, 88 insertions(+), 55 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index c2c2d920..0e541053 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -12,6 +12,7 @@ #include #include "Rtcp.h" #include "Util/logger.h" +#include "RtcpFCI.h" namespace mediakit { @@ -493,18 +494,65 @@ string RtcpFB::dumpString() const { printer << RtcpHeader::dumpHeader(); printer << "ssrc:" << ssrc << "\r\n"; printer << "ssrc_media:" << ssrc_media << "\r\n"; - auto fci = (uint8_t *)&ssrc_media + sizeof(ssrc_media); + auto fci_data = (uint8_t *)&ssrc_media + sizeof(ssrc_media); auto fci_len = getSize() - sizeof(RtcpFB); - if (fci_len) { - switch ((RtcpType) pt) { - case RtcpType::RTCP_PSFB : { - break; - } - case RtcpType::RTCP_RTPFB : { - break; + switch ((RtcpType) pt) { + case RtcpType::RTCP_PSFB : { + switch ((PSFBType) report_count) { + case PSFBType::RTCP_PSFB_SLI : { + FCI_SLI *fci = (FCI_SLI *) fci_data; + fci->check(fci_len); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + break; + } + case PSFBType::RTCP_PSFB_PLI : { + CHECK(fci_len == 0); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count); + break; + } + + case PSFBType::RTCP_PSFB_FIR : { + FCI_FIR *fci = (FCI_FIR *) fci_data; + fci->check(fci_len); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + break; + } + + case PSFBType::RTCP_PSFB_REMB : { + FCI_REMB *fci = (FCI_REMB *) fci_data; + fci->check(fci_len); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + break; + } + default:{ + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << hexdump(fci_data, fci_len); + break; + } } + break; } - printer << "fci:" << hexdump(fci, fci_len); + case RtcpType::RTCP_RTPFB : { + switch ((RTPFBType) report_count) { + case RTPFBType::RTCP_RTPFB_NACK : { + FCI_NACK *fci = (FCI_NACK *) fci_data; + fci->check(fci_len); + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString(); + break; + } + case RTPFBType::RTCP_RTPFB_TWCC : { + FCI_TWCC *fci = (FCI_TWCC *) fci_data; + fci->check(fci_len); + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString(fci_len); + break; + } + default: { + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << hexdump(fci_data, fci_len); + break; + } + } + break; + } + default: /*不可达*/ assert(0); break; } return std::move(printer); } diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index ddc5ae71..5637e4ee 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -85,7 +85,7 @@ namespace mediakit { XX(RTCP_PSFB_TSTR, 5)\ XX(RTCP_PSFB_TSTN, 6)\ XX(RTCP_PSFB_VBCM, 7) \ - XX(RTCP_PSFB_AFB, 15) + XX(RTCP_PSFB_REMB, 15) //https://tools.ietf.org/html/rfc4585#section-6.2 //6.2. Transport Layer Feedback Messages diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 28ed2945..e9627abc 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -453,7 +453,7 @@ map > FCI_TWCC::getPacketCh string FCI_TWCC::dumpString(size_t total_size) const { _StrPrinter printer; auto map = getPacketChunkList(total_size); - printer << "twcc fci, base_seq:" << getBaseSeq() << ",pkt_status_count:" << getPacketCount() << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n"; + printer << "twcc fci, base_seq:" << getBaseSeq() << ", pkt_status_count:" << getPacketCount() << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n"; for (auto &pr : map) { printer << "rtp seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", delta:" << pr.second.second << "\n"; } diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 69feb7de..f6fbc4f8 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1257,18 +1257,29 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ //此处调整偏好的编码格式优先级 preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; rtcp_fb = {"transport-cc"}; - extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"}; + extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level", + "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", + "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", + "4 urn:ietf:params:rtp-hdrext:sdes:mid", + "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", + "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"}; break; } case TrackVideo: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecH264, CodecH265}; rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"}; - extmap = {"2 urn:ietf:params:rtp-hdrext:toffset", - "3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", - "4 urn:3gpp:video-orientation", - "5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", - "6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"}; + extmap = {"14 urn:ietf:params:rtp-hdrext:toffset", + "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", + "13 urn:3gpp:video-orientation", + "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", + "12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", + "11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type", + "7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing", + "8 http://www.webrtc.org/experiments/rtp-hdrext/color-space", + "4 urn:ietf:params:rtp-hdrext:sdes:mid", + "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", + "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"}; break; } case TrackApplication: { diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index d4392385..1a13c656 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -544,7 +544,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - InfoL << rtcp->dumpString(); + InfoL << "\n" << rtcp->dumpString(); break; } default: break; diff --git a/webrtc/readme.md b/webrtc/readme.md index ff832278..2884df14 100644 --- a/webrtc/readme.md +++ b/webrtc/readme.md @@ -1,37 +1,11 @@ -# 致谢与声明 -本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为： - -- ice相关功能： - - IceServer.cpp - - IceServer.hpp - - StunPacket.cpp - - StunPacket.hpp - - Utils.hpp - -- dtls相关功能： - - DtlsTransport.cpp - - DtlsTransport.hpp - -- srtp相关功能： - - SrtpSession.cpp - - SrtpSession.hpp - - -以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者， -用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。 - -同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者， -在集成MediaSoup相关代码前期，主要参考这个项目。 - -另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)， -其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js - -# 现状与规划 -ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有： - -- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。 -- 2、实现rtp重传等相关功能。 -- 3、实现simulecast相关功能。 -- 4、fec、rtp扩展等其他功能。 -- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。 - +14 urn:ietf:params:rtp-hdrext:toffset +2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +13 urn:3gpp:video-orientation +3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type +7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing +8 http://www.webrtc.org/experiments/rtp-hdrext/color-space +4 urn:ietf:params:rtp-hdrext:sdes:mid +5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id \ No newline at end of file From 6c951c8ca986f7a16be5a8a4981b2ff827b6bb4f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 26 Apr 2021 21:50:10 +0800 Subject: [PATCH 121/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtcp=20padding?= =?UTF-8?q?=E7=9B=B8=E5=85=B3=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 46 +++++++++++++++++++++++++++++++++++++++------- src/Rtcp/Rtcp.h | 7 ++++++- 2 files changed, 45 insertions(+), 8 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 0e541053..105d9706 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -68,12 +68,26 @@ static void setupHeader(RtcpHeader *rtcp, RtcpType type, size_t report_count, si rtcp->setSize(total_bytes); } +static void setupPadding(RtcpHeader *rtcp, size_t padding_size) { + if (padding_size) { + rtcp->padding = 1; + ((uint8_t *) rtcp)[rtcp->getSize() - 1] = padding_size & 0xFF; + } else { + rtcp->padding = 0; + } +} + ///////////////////////////////////////////////////////////////////////////// string RtcpHeader::dumpHeader() const{ _StrPrinter printer; printer << "version:" << version << "\r\n"; - printer << "padding:" << padding << "\r\n"; + if (padding) { + printer << "padding:" << padding << " " << getPaddingSize() << "\r\n"; + } else { + printer << "padding:" << padding << "\r\n"; + } + switch ((RtcpType)pt) { case RtcpType::RTCP_RTPFB : { printer << "report_count:" << rtpfbTypeToStr((RTPFBType) report_count) << "\r\n"; @@ -132,6 +146,13 @@ size_t RtcpHeader::getSize() const { return (1 + ntohs(length)) << 2; } +size_t RtcpHeader::getPaddingSize() const{ + if (!padding) { + return 0; + } + return ((uint8_t *) this)[getSize() - 1]; +} + void RtcpHeader::setSize(size_t size) { //不包含rtcp头的长度 length = htons((uint16_t)((size >> 2) - 1)); @@ -226,9 +247,11 @@ Buffer::Ptr RtcpHeader::toBuffer(std::shared_ptr rtcp) { ///////////////////////////////////////////////////////////////////////////// std::shared_ptr RtcpSR::create(size_t item_count) { - auto bytes = alignSize(sizeof(RtcpSR) - sizeof(ReportItem) + item_count * sizeof(ReportItem)); + auto real_size = sizeof(RtcpSR) - sizeof(ReportItem) + item_count * sizeof(ReportItem); + auto bytes = alignSize(real_size); auto ptr = (RtcpSR *) new char[bytes]; setupHeader(ptr, RtcpType::RTCP_SR, item_count, bytes); + setupPadding(ptr, bytes - real_size); return std::shared_ptr(ptr, [](RtcpSR *ptr) { delete[] (char *) ptr; }); @@ -336,9 +359,11 @@ void ReportItem::net2Host() { ///////////////////////////////////////////////////////////////////////////// std::shared_ptr RtcpRR::create(size_t item_count) { - auto bytes = alignSize(sizeof(RtcpRR) - sizeof(ReportItem) + item_count * sizeof(ReportItem)); + auto real_size = sizeof(RtcpRR) - sizeof(ReportItem) + item_count * sizeof(ReportItem); + auto bytes = alignSize(real_size); auto ptr = (RtcpRR *) new char[bytes]; setupHeader(ptr, RtcpType::RTCP_RR, item_count, bytes); + setupPadding(ptr, bytes - real_size); return std::shared_ptr(ptr, [](RtcpRR *ptr) { delete[] (char *) ptr; }); @@ -413,7 +438,8 @@ std::shared_ptr RtcpSdes::create(const std::vector &item_text) //统计所有SdesItem对象占用的空间 item_total_size += alignSize(SdesItem::minSize() + (0xFF & text.size())); } - auto bytes = alignSize(sizeof(RtcpSdes) - sizeof(SdesItem) + item_total_size); + auto real_size = sizeof(RtcpSdes) - sizeof(SdesItem) + item_total_size; + auto bytes = alignSize(real_size); auto ptr = (RtcpSdes *) new char[bytes]; auto item_ptr = &ptr->items; for (auto &text : item_text) { @@ -424,6 +450,7 @@ std::shared_ptr RtcpSdes::create(const std::vector &item_text) } setupHeader(ptr, RtcpType::RTCP_SDES, item_text.size(), bytes); + setupPadding(ptr, bytes - real_size); return std::shared_ptr(ptr, [](RtcpSdes *ptr) { delete [] (char *) ptr; }); @@ -470,12 +497,14 @@ std::shared_ptr RtcpFB::create_l(RtcpType type, int fmt, const void *fci if (!fci) { fci_len = 0; } - auto bytes = alignSize(sizeof(RtcpFB) + fci_len); + auto real_size = sizeof(RtcpFB) + fci_len; + auto bytes = alignSize(real_size); auto ptr = (RtcpRR *) new char[bytes]; if (fci && fci_len) { memcpy(ptr + sizeof(RtcpFB), fci, fci_len); } setupHeader(ptr, type, fmt, bytes); + setupPadding(ptr, bytes - real_size); return std::shared_ptr((RtcpFB *) ptr, [](RtcpFB *ptr) { delete[] (char *) ptr; }); @@ -495,7 +524,8 @@ string RtcpFB::dumpString() const { printer << "ssrc:" << ssrc << "\r\n"; printer << "ssrc_media:" << ssrc_media << "\r\n"; auto fci_data = (uint8_t *)&ssrc_media + sizeof(ssrc_media); - auto fci_len = getSize() - sizeof(RtcpFB); + auto fci_len = (ssize_t)getSize() - getPaddingSize() - sizeof(RtcpFB); + CHECK(fci_len >= 0); switch ((RtcpType) pt) { case RtcpType::RTCP_PSFB : { switch ((PSFBType) report_count) { @@ -568,9 +598,11 @@ void RtcpFB::net2Host(size_t size) { std::shared_ptr RtcpBye::create(const std::vector &ssrcs, const string &reason) { assert(reason.size() <= 0xFF); - auto bytes = alignSize(sizeof(RtcpHeader) + sizeof(uint32_t) * ssrcs.size() + 1 + reason.size()); + auto real_size = sizeof(RtcpHeader) + sizeof(uint32_t) * ssrcs.size() + 1 + reason.size(); + auto bytes = alignSize(real_size); auto ptr = (RtcpBye *) new char[bytes]; setupHeader(ptr, RtcpType::RTCP_BYE, ssrcs.size(), bytes); + setupPadding(ptr, bytes - real_size); auto *ssrc_ptr = &(((RtcpBye *) ptr)->ssrc); for (auto ssrc : ssrcs) { diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index 5637e4ee..836699d2 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -172,7 +172,7 @@ public: #else //reception report count uint32_t report_count: 5; - //padding，固定为0 + //padding，末尾是否有追加填充 uint32_t padding: 1; //版本号，固定为2 uint32_t version: 2; @@ -212,6 +212,11 @@ public: */ size_t getSize() const; + /** + * 后面追加padding数据长度 + */ + size_t getPaddingSize() const; + /** * 设置rtcp length字段 * @param size rtcp总长度，单位字节 From 1d84bb54585e986015986cb4702c65c7a166fecc Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 26 Apr 2021 21:50:31 +0800 Subject: [PATCH 122/218] =?UTF-8?q?=E5=AE=8C=E5=96=84twcc=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index e9627abc..dc48be52 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -423,22 +423,26 @@ map > FCI_TWCC::getPacketCh auto end = (uint8_t *) this + total_size; CHECK(ptr < end); auto seq = getBaseSeq(); - - for (uint8_t i = 0; i < getPacketCount();) { + auto rtp_count = getPacketCount(); + for (uint8_t i = 0; i < rtp_count;) { CHECK(ptr + RunLengthChunk::kSize <= end); RunLengthChunk *chunk = (RunLengthChunk *) ptr; if (!chunk->type) { //RunLengthChunk for (auto j = 0; j < chunk->getRunLength(); ++j) { ret.emplace(seq++, std::make_pair((SymbolStatus) chunk->symbol, 0)); - ++i; + if (++i >= rtp_count) { + break; + } } } else { //StatusVecChunk StatusVecChunk *chunk = (StatusVecChunk *) ptr; for (auto &symbol : chunk->getSymbolList()) { ret.emplace(seq++, std::make_pair(symbol, 0)); - ++i; + if (++i >= rtp_count) { + break; + } } } ptr += 2; From 26cd5660cd58737e8a949369a375f7e9aefa264e Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 27 Apr 2021 01:16:01 +0800 Subject: [PATCH 123/218] =?UTF-8?q?sdp=E8=BF=94=E5=9B=9E=E7=9C=9F=E5=AE=9E?= =?UTF-8?q?ip=E5=92=8C=E7=AB=AF=E5=8F=A3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.h | 4 ++++ webrtc/WebRtcTransport.cpp | 17 ++++++++++++++++- 2 files changed, 20 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 6a20fc40..df3e529a 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -97,6 +97,10 @@ public: } virtual const char* getKey() const = 0; + void reset() { + value.clear(); + } + protected: mutable string value; }; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 1a13c656..80b3d6ae 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -408,7 +408,22 @@ void WebRtcTransportImp::onStartWebRTC() { void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ WebRtcTransport::onCheckSdp(type, sdp); - if (type != SdpType::answer || !canSendRtp()) { + if (type != SdpType::answer) { + return; + } + + GET_CONFIG(string, extern_ip, RTC::kExternIP); + for (auto &m : sdp.media) { + m.addr.reset(); + m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip; + m.rtcp_addr.reset(); + m.rtcp_addr.address = m.addr.address; + m.rtcp_addr.port = _socket->get_local_port(); + m.port = m.rtcp_addr.port; + sdp.origin.address = m.addr.address; + } + + if (!canSendRtp()) { return; } From ac55ae79fb6c66ccd6f328f6b5c689ec38921dd7 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 13:55:00 +0800 Subject: [PATCH 124/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dfci=E5=86=85=E5=AD=98?= =?UTF-8?q?=E8=B6=8A=E7=95=8C=E9=97=AE=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 105d9706..16e9c9fa 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -499,9 +499,9 @@ std::shared_ptr RtcpFB::create_l(RtcpType type, int fmt, const void *fci } auto real_size = sizeof(RtcpFB) + fci_len; auto bytes = alignSize(real_size); - auto ptr = (RtcpRR *) new char[bytes]; + auto ptr = (RtcpFB *) new char[bytes]; if (fci && fci_len) { - memcpy(ptr + sizeof(RtcpFB), fci, fci_len); + memcpy((char *)ptr + sizeof(RtcpFB), fci, fci_len); } setupHeader(ptr, type, fmt, bytes); setupPadding(ptr, bytes - real_size); From 9396270ce26dc5a0f3a97f3780169fa5f0c3b9fa Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 15:07:15 +0800 Subject: [PATCH 125/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0twcc=E7=9B=B8?= =?UTF-8?q?=E5=85=B3api?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 110 +++++++++++++++++++++++++++++++------------------ webrtc/Sdp.h | 9 ++-- 2 files changed, 75 insertions(+), 44 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index f6fbc4f8..6317c87b 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -921,7 +921,7 @@ void RtcSession::loadFrom(const string &str, bool check) { } for (auto rtpfb_it = rtcpfb_map.find(pt); rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) { - plan.rtcp_fb.emplace_back(rtpfb_it->second.rtcp_type); + plan.rtcp_fb.emplace(rtpfb_it->second.rtcp_type); } } } @@ -1241,6 +1241,20 @@ const RtcMedia *RtcSession::getMedia(TrackType type) const{ return nullptr; } +static string const kTWCCRtcpFb = "transport-cc"; +static string const kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; + + +void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ + if (!enable) { + rtcp_fb.erase(kTWCCRtcpFb); + extmap.erase(kTWCCExtMap); + } else { + rtcp_fb.emplace(kTWCCRtcpFb); + extmap.emplace(kTWCCExtMap); + } +} + void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ enable = true; rtcp_mux = true; @@ -1256,30 +1270,34 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ case TrackAudio: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; - rtcp_fb = {"transport-cc"}; - extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level", - "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", - "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", - "4 urn:ietf:params:rtp-hdrext:sdes:mid", - "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", - "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"}; + rtcp_fb = {kTWCCRtcpFb}; + extmap = { + kTWCCExtMap, + "urn:ietf:params:rtp-hdrext:ssrc-audio-level", + "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", + "urn:ietf:params:rtp-hdrext:sdes:mid", + "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", + "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id" + }; break; } case TrackVideo: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecH264, CodecH265}; - rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"}; - extmap = {"14 urn:ietf:params:rtp-hdrext:toffset", - "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", - "13 urn:3gpp:video-orientation", - "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", - "12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", - "11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type", - "7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing", - "8 http://www.webrtc.org/experiments/rtp-hdrext/color-space", - "4 urn:ietf:params:rtp-hdrext:sdes:mid", - "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", - "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"}; + rtcp_fb = {kTWCCRtcpFb, "nack", "ccm fir", "nack pli", "goog-remb"}; + extmap = { + kTWCCExtMap, + "urn:ietf:params:rtp-hdrext:toffset", + "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", + "urn:3gpp:video-orientation", + "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", + "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type", + "http://www.webrtc.org/experiments/rtp-hdrext/video-timing", + "http://www.webrtc.org/experiments/rtp-hdrext/color-space", + "urn:ietf:params:rtp-hdrext:sdes:mid", + "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", + "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id" + }; break; } case TrackApplication: { @@ -1333,6 +1351,29 @@ void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType typ } } +void RtcConfigure::enableTWCC(bool enable, TrackType type){ + switch (type) { + case TrackAudio: { + audio.enableTWCC(enable); + break; + } + case TrackVideo: { + video.enableTWCC(enable); + break; + } + case TrackApplication: { + application.enableTWCC(enable); + break; + } + default: { + audio.enableTWCC(enable); + video.enableTWCC(enable); + application.enableTWCC(enable); + break; + } + } +} + shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ shared_ptr ret = std::make_shared(); ret->version = offer.version; @@ -1479,35 +1520,22 @@ RETRY: } } - //这是我们支持的扩展 - unordered_set extmap_set; - for (auto &ext : configure.extmap) { - SdpAttrExtmap ext_cfg; - ext_cfg.parse(ext); - extmap_set.emplace(ext_cfg.ext); - } - - //对方和我方都支持的扩展，那么我们都支持 + //对方和我方都支持的扩展，那么我们才支持 for (auto &ext : offer_media.extmap) { - if (extmap_set.find(ext.ext) != extmap_set.end()) { + if (configure.extmap.find(ext.ext) != configure.extmap.end()) { answer_media.extmap.emplace_back(ext); } } - //我们支持的rtcp类型 - unordered_set rtcp_fb_set; - for (auto &fp : configure.rtcp_fb) { - rtcp_fb_set.emplace(fp); - } - vector offer_rtcp_fb; + + auto &rtcp_fb_ref = answer_media.plan[0].rtcp_fb; + rtcp_fb_ref.clear(); + //对方和我方都支持的rtcpfb，那么我们才支持 for (auto &fp : offer_plan_ptr->rtcp_fb) { - if (rtcp_fb_set.find(fp) != rtcp_fb_set.end()) { + if (configure.rtcp_fb.find(fp) != configure.rtcp_fb.end()) { //对方该rtcp被我们支持 - offer_rtcp_fb.emplace_back(fp); + rtcp_fb_ref.emplace(fp); } } - - //修改为我们支持的rtcp-fb类型 - answer_media.plan[0].rtcp_fb.swap(offer_rtcp_fb); ret->media.emplace_back(answer_media); return; } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index df3e529a..6c9784b0 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -591,7 +591,7 @@ public: //音频时有效 uint32_t channel = 0; //rtcp反馈 - vector rtcp_fb; + set rtcp_fb; map fmtp; string getFmtp(const char *key) const; @@ -693,12 +693,13 @@ public: RtpDirection direction{RtpDirection::invalid}; SdpAttrFingerprint fingerprint; - vector rtcp_fb; + set rtcp_fb; + set extmap; vector preferred_codec; - vector extmap; vector candidate; void setDefaultSetting(TrackType type); + void enableTWCC(bool enable = true); }; RtcTrackConfigure video; @@ -715,6 +716,8 @@ public: void setPlayRtspInfo(const string &sdp); + void enableTWCC(bool enable = true, TrackType type = TrackInvalid); + private: void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); bool onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec); From 726e9096018431dd7e1cacd8fa9a9d6dc54fd9a0 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 15:41:36 +0800 Subject: [PATCH 126/218] =?UTF-8?q?=E6=94=AF=E6=8C=81remb=E6=8E=A7?= =?UTF-8?q?=E5=88=B6=E6=8E=A8=E6=B5=81=E6=AF=94=E7=89=B9=E7=8E=87?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- conf/config.ini | 2 ++ webrtc/Sdp.cpp | 10 +++++++++- webrtc/Sdp.h | 1 + webrtc/WebRtcTransport.cpp | 36 ++++++++++++++++++++++++++++++++---- webrtc/WebRtcTransport.h | 4 +++- 5 files changed, 47 insertions(+), 6 deletions(-) diff --git a/conf/config.ini b/conf/config.ini index c7330ed2..35a6cc58 100644 --- a/conf/config.ini +++ b/conf/config.ini @@ -211,6 +211,8 @@ timeoutSec=15 timeoutSec=15 #本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip externIP= +#设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质 +rembBitRate=1000000 [rtsp] #rtsp专有鉴权方式是采用base64还是md5方式 diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 6317c87b..8a671f66 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1241,10 +1241,18 @@ const RtcMedia *RtcSession::getMedia(TrackType type) const{ return nullptr; } +bool RtcSession::supportRtcpFb(const string &name, TrackType type) const { + auto media = getMedia(type); + if (!media) { + return false; + } + auto &ref = media->plan[0].rtcp_fb; + return ref.find(name) != ref.end(); +} + static string const kTWCCRtcpFb = "transport-cc"; static string const kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; - void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ if (!enable) { rtcp_fb.erase(kTWCCRtcpFb); diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 6c9784b0..bcd1783c 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -667,6 +667,7 @@ public: string toString() const; string toRtspSdp() const; const RtcMedia *getMedia(TrackType type) const; + bool supportRtcpFb(const string &name, TrackType type = TrackType::TrackVideo) const; private: RtcSessionSdp::Ptr toRtcSessionSdp() const; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 80b3d6ae..178daca6 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -11,7 +11,9 @@ #include "WebRtcTransport.h" #include #include "Rtcp/Rtcp.h" +#include "Rtcp/RtcpFCI.h" #include "Rtsp/RtpReceiver.h" + #define RTX_SSRC_OFFSET 2 #define RTP_CNAME "zlmediakit-rtp" #define RTX_CNAME "zlmediakit-rtx" @@ -23,10 +25,13 @@ namespace RTC { const string kTimeOutSec = RTC_FIELD"timeoutSec"; //服务器外网ip const string kExternIP = RTC_FIELD"externIP"; +//设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质 +const string kRembBitRate = RTC_FIELD"rembBitRate"; static onceToken token([]() { mINI::Instance()[kTimeOutSec] = 15; mINI::Instance()[kExternIP] = ""; + mINI::Instance()[kRembBitRate] = 0; }); }//namespace RTC @@ -134,6 +139,22 @@ RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{ return _ice_server->GetSelectedTuple(); } +void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) { + auto remb = FCI_REMB::create({ssrc}, (uint32_t)bit_rate); + auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size()); + fb->ssrc = htonl(0); + fb->ssrc_media = htonl(ssrc); + sendRtcpPacket((char *) fb.get(), fb->getSize(), true); + TraceL << ssrc << " " << bit_rate; +} + +void WebRtcTransport::sendRtcpPli(uint32_t ssrc) { + auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI); + pli->ssrc = htonl(0); + pli->ssrc_media = htonl(ssrc); + sendRtcpPacket((char *) pli.get(), pli->getSize(), true); +} + string getFingerprint(const string &algorithm_str, const std::shared_ptr &transport){ auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str); for (auto &finger_prints : transport->GetLocalFingerprints()) { @@ -163,6 +184,12 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ } } +void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const { + //开启remb后关闭twcc，因为开启twcc后remb无效 + GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); + configure.enableTWCC(!remb_bit_rate); +} + std::string WebRtcTransport::getAnswerSdp(const string &offer){ try { //// 解析offer sdp //// @@ -389,6 +416,10 @@ void WebRtcTransportImp::onStartWebRTC() { if (canRecvRtp()) { _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp()); + GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); + if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb("goog-remb")) { + sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); + } } if (canSendRtp()) { _reader = _play_src->getRing()->attach(getPoller(), true); @@ -592,10 +623,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr if (_pli_ticker.elapsedTime() > 2000) { //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); - auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI); - pli->ssrc = htonl(0); - pli->ssrc_media = htonl(_recv_video_ssrc); - sendRtcpPacket((char *) pli.get(), pli->getSize(), true); + sendRtcpPli(_recv_video_ssrc); } if (_push_src) { _push_src->onWrite(std::move(rtp), false); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 50f45604..9cc64664 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -93,7 +93,7 @@ protected: protected: virtual void onStartWebRTC() = 0; - virtual void onRtcConfigure(RtcConfigure &configure) const {} + virtual void onRtcConfigure(RtcConfigure &configure) const; virtual void onCheckSdp(SdpType type, RtcSession &sdp); virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0; @@ -104,6 +104,8 @@ protected: protected: const RtcSession& getSdp(SdpType type) const; RTC::TransportTuple* getSelectedTuple() const; + void sendRtcpRemb(uint32_t ssrc, size_t bit_rate); + void sendRtcpPli(uint32_t ssrc); private: void onSendSockData(const char *buf, size_t len, bool flush = true); From 514a028eac7225783326a73ae58512def6184f7f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 16:03:05 +0800 Subject: [PATCH 127/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dfu-a=E6=89=93?= =?UTF-8?q?=E5=8C=85=E6=97=B6=EF=BC=8C=E6=97=A0rtp=E8=B4=9F=E8=BD=BD?= =?UTF-8?q?=E7=9A=84=E6=83=85=E5=86=B5?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/H264Rtp.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp index 51f899d2..f9f4f570 100644 --- a/src/Extension/H264Rtp.cpp +++ b/src/Extension/H264Rtp.cpp @@ -204,9 +204,9 @@ void H264RtpEncoder::packRtp(const char *ptr, size_t len, uint32_t pts, bool is_ } void H264RtpEncoder::packRtpFu(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){ - auto packet_size = getMaxSize() - 2; + auto packet_size = getMaxSize() - 1; if (len <= packet_size) { - //小于FU-A打包最小字节长度要求，采用STAP-A模式，如果frame长度正好是mtu-2, 那么打包的rtp长度是mtu + 1 + //小于FU-A打包最小字节长度要求，采用STAP-A模式 packRtpStapA(ptr, len, pts, is_mark, gop_pos); return; } From e021279fab6dc025f2930336ed03fdca18b97431 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 16:07:01 +0800 Subject: [PATCH 128/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8D"Too=20short=20data?= =?UTF-8?q?=20for=20FU-A=20H.264=20RTP=20packet"=E7=9A=84=E9=97=AE?= =?UTF-8?q?=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/H264Rtp.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp index f9f4f570..0db7585a 100644 --- a/src/Extension/H264Rtp.cpp +++ b/src/Extension/H264Rtp.cpp @@ -204,8 +204,8 @@ void H264RtpEncoder::packRtp(const char *ptr, size_t len, uint32_t pts, bool is_ } void H264RtpEncoder::packRtpFu(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){ - auto packet_size = getMaxSize() - 1; - if (len <= packet_size) { + auto packet_size = getMaxSize() - 2; + if (len <= packet_size + 1) { //小于FU-A打包最小字节长度要求，采用STAP-A模式 packRtpStapA(ptr, len, pts, is_mark, gop_pos); return; From 6ef69d4ef101188cbf7029680e0a694c70133fed Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Wed, 28 Apr 2021 17:32:16 +0800 Subject: [PATCH 129/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dbug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- api/tests/pusher.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/api/tests/pusher.c b/api/tests/pusher.c index 76ca4297..95f33dc7 100644 --- a/api/tests/pusher.c +++ b/api/tests/pusher.c @@ -48,9 +48,9 @@ void release_pusher(mk_media *ptr) { void release_context(Context **ptr){ if (ptr && *ptr) { - release_pusher((*ptr)->pusher); - release_media((*ptr)->media); - release_player((*ptr)->player); + release_pusher(&(*ptr)->pusher); + release_media(&(*ptr)->media); + release_player(&(*ptr)->player); free(*ptr); *ptr = NULL; } From 5145c7e658a4062d4fed22b36c35317a69624be8 Mon Sep 17 00:00:00 2001 From: xgj Date: Thu, 29 Apr 2021 18:36:04 +0800 Subject: [PATCH 130/218] optimize one track ready speed not wait 3000ms --- src/Rtsp/RtspMediaSourceImp.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/Rtsp/RtspMediaSourceImp.h b/src/Rtsp/RtspMediaSourceImp.h index 37731f1e..35ce3f5e 100644 --- a/src/Rtsp/RtspMediaSourceImp.h +++ b/src/Rtsp/RtspMediaSourceImp.h @@ -89,6 +89,8 @@ public: _muxer->addTrack(track); track->addDelegate(_muxer); } + if(this->_all_demuxer_track_ready) + this->addTrackCompleted(); } /** @@ -108,6 +110,7 @@ public: if (_muxer) { _muxer->addTrackCompleted(); } + this->_all_demuxer_track_ready = true; } void resetTracks() override { @@ -141,6 +144,7 @@ private: RtspDemuxer::Ptr _demuxer; MultiMediaSourceMuxer::Ptr _muxer; bool _all_track_ready = false; + bool _all_demuxer_track_ready = false; }; } /* namespace mediakit */ From 88e1c323c31179180464d943b479967615d184cf Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 09:24:09 +0800 Subject: [PATCH 131/218] =?UTF-8?q?=E5=88=A0=E9=99=A4=E6=97=A0=E6=95=88?= =?UTF-8?q?=E9=85=8D=E7=BD=AE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Common/config.cpp | 6 ------ src/Common/config.h | 4 ---- 2 files changed, 10 deletions(-) diff --git a/src/Common/config.cpp b/src/Common/config.cpp index 5fc2dc3a..47b8177a 100644 --- a/src/Common/config.cpp +++ b/src/Common/config.cpp @@ -190,16 +190,10 @@ namespace Rtp { //RTP打包最大MTU,公网情况下更小 const string kVideoMtuSize = RTP_FIELD"videoMtuSize"; const string kAudioMtuSize = RTP_FIELD"audioMtuSize"; -//RTP排序缓存最大个数 -const string kMaxRtpCount = RTP_FIELD"maxRtpCount"; -//如果RTP序列正确次数累计达到该数字就启动清空排序缓存 -const string kClearCount = RTP_FIELD"clearCount"; onceToken token([](){ mINI::Instance()[kVideoMtuSize] = 1400; mINI::Instance()[kAudioMtuSize] = 600; - mINI::Instance()[kMaxRtpCount] = 50; - mINI::Instance()[kClearCount] = 10; },nullptr); } //namespace Rtsp diff --git a/src/Common/config.h b/src/Common/config.h index 6b26ab3c..a104a789 100644 --- a/src/Common/config.h +++ b/src/Common/config.h @@ -247,10 +247,6 @@ namespace Rtp { extern const string kVideoMtuSize; //RTP打包最大MTU,公网情况下更小 extern const string kAudioMtuSize; -//RTP排序缓存最大个数 -extern const string kMaxRtpCount; -//如果RTP序列正确次数累计达到该数字就启动清空排序缓存 -extern const string kClearCount; } //namespace Rtsp ////////////组播配置/////////// From 7ebdce38e964a8d69ddee2def0c6fbc49e19af12 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 11:41:35 +0800 Subject: [PATCH 132/218] =?UTF-8?q?=E4=BD=9C=E8=80=85=E5=88=97=E8=A1=A8?= =?UTF-8?q?=E5=8B=98=E8=AF=AF?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- AUTHORS | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/AUTHORS b/AUTHORS index 3456f103..cb4eba4b 100644 --- a/AUTHORS +++ b/AUTHORS @@ -23,13 +23,12 @@ sunhui mirs Kevin Cheng Liu Jiang -along +along(https://github.com/alongl) qingci lyg1949 zhlong -Luke 大裤衩 <3503207480@qq.com> droid.chow [陈晓林](https://github.com/musicwood) [老衲不出家](https://github.com/monktan89) -[big panda](<2381267071@qq.com>) \ No newline at end of file +big panda <2381267071@qq.com> \ No newline at end of file From af5195374a7a510bd585a856b9fbbc03afd55bcc Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 11:41:58 +0800 Subject: [PATCH 133/218] =?UTF-8?q?=E6=9B=B4=E6=96=B0=E9=93=BE=E6=8E=A5?= =?UTF-8?q?=E5=9C=B0=E5=9D=80?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- README.md | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/README.md b/README.md index 1322dfa5..e3d677c9 100644 --- a/README.md +++ b/README.md @@ -143,7 +143,7 @@ bash build_docker_images.sh - [基于ZLMediaKit分支的管理WEB网站](https://github.com/chenxiaolei/ZLMediaKit_NVR_UI) - 流媒体管理平台 - - [功能强大的流媒体控制管理接口平台,支持GB28181](https://github.com/chatop2020/StreamNode) + - [功能强大的流媒体控制管理接口平台,支持GB28181](https://github.com/chatop2020/AKStream) - [GB28181-2016网络视频平台](https://github.com/648540858/wvp-GB28181-pro) - [node-js版本的GB28181平台](https://gitee.com/hfwudao/GB28181_Node_Http) - [Go实现的海康ehome服务器](https://github.com/tsingeye/FreeEhome) @@ -218,11 +218,10 @@ bash build_docker_images.sh [mirs](fangpengcheng@bilibili.com>) [Kevin Cheng](kevin__cheng@outlook.com>) [Liu Jiang](root@oopy.org>) -[along](alongl@users.noreply.github.com>) +[along](https://github.com/alongl) [qingci](xpy66swsry@gmail.com>) [lyg1949](zh.ghlong@qq.com>) [zhlong](zh.ghlong@qq.com>) -[Luke](automan@easydarwin.org>) [大裤衩](3503207480@qq.com>) [droid.chow](droid.chow@gmail.com>) [陈晓林](https://github.com/musicwood) @@ -240,6 +239,6 @@ bash build_docker_images.sh 欢迎捐赠以便更好的推动项目的发展，谢谢您的支持! 同时欢迎捐赠公网服务器用于在线展示效果。 -[支付宝](https://gitee.com/xia-chu/other/raw/master/IMG_3919.JPG) +[支付宝](https://github.com/xia-chu/other/blob/master/IMG_3919.JPG) -[微信](https://gitee.com/xia-chu/other/raw/master/IMG_3920.JPG) +[微信](https://github.com/xia-chu/other/blob/master/IMG_3920.JPG) From 6274cbdd2845533fd82042860b44e77046d6fa12 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 14:31:08 +0800 Subject: [PATCH 134/218] =?UTF-8?q?=E7=BB=9F=E4=B8=80=E4=BB=A3=E7=A0=81?= =?UTF-8?q?=E9=A3=8E=E6=A0=BC?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtsp/RtspMediaSourceImp.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/Rtsp/RtspMediaSourceImp.h b/src/Rtsp/RtspMediaSourceImp.h index 35ce3f5e..53e9d4d0 100644 --- a/src/Rtsp/RtspMediaSourceImp.h +++ b/src/Rtsp/RtspMediaSourceImp.h @@ -89,8 +89,9 @@ public: _muxer->addTrack(track); track->addDelegate(_muxer); } - if(this->_all_demuxer_track_ready) + if (_all_demuxer_track_ready) { this->addTrackCompleted(); + } } /** @@ -110,7 +111,7 @@ public: if (_muxer) { _muxer->addTrackCompleted(); } - this->_all_demuxer_track_ready = true; + _all_demuxer_track_ready = true; } void resetTracks() override { From 8f17fcdf1d1e2594c09fe94bf2de34bcea0e1193 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 14:49:06 +0800 Subject: [PATCH 135/218] =?UTF-8?q?remb=E6=94=B9=E6=88=90=E5=9B=BA?= =?UTF-8?q?=E5=AE=9A=E9=97=B4=E9=9A=94=E5=8F=91=E9=80=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 178daca6..cfe70132 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -416,10 +416,6 @@ void WebRtcTransportImp::onStartWebRTC() { if (canRecvRtp()) { _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp()); - GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); - if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb("goog-remb")) { - sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); - } } if (canSendRtp()) { _reader = _play_src->getRing()->attach(getPoller(), true); @@ -624,6 +620,12 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); sendRtcpPli(_recv_video_ssrc); + + //开启remb，则发送remb包调节比特率 + GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); + if (remb_bit_rate) { + sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); + } } if (_push_src) { _push_src->onWrite(std::move(rtp), false); From dc2c8829948da7b6d29d445ec2d5326eaebf5114 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 15:08:43 +0800 Subject: [PATCH 136/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0remb=E5=BC=80?= =?UTF-8?q?=E5=85=B3?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 20 ++++++++++++++++---- webrtc/Sdp.h | 1 + 2 files changed, 17 insertions(+), 4 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 8a671f66..3c137fde 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1252,6 +1252,8 @@ bool RtcSession::supportRtcpFb(const string &name, TrackType type) const { static string const kTWCCRtcpFb = "transport-cc"; static string const kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; +static string const kRembRtcpFb = "goog-remb"; +static string const kRembExtMap = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ if (!enable) { @@ -1263,6 +1265,16 @@ void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ } } +void RtcConfigure::RtcTrackConfigure::enableREMB(bool enable){ + if (!enable) { + rtcp_fb.erase(kRembRtcpFb); + extmap.erase(kRembExtMap); + } else { + rtcp_fb.emplace(kRembRtcpFb); + extmap.emplace(kRembExtMap); + } +} + void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ enable = true; rtcp_mux = true; @@ -1278,11 +1290,11 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ case TrackAudio: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; - rtcp_fb = {kTWCCRtcpFb}; + rtcp_fb = {kTWCCRtcpFb, kRembRtcpFb}; extmap = { kTWCCExtMap, + kRembExtMap, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", - "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", "urn:ietf:params:rtp-hdrext:sdes:mid", "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id" @@ -1292,11 +1304,11 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ case TrackVideo: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecH264, CodecH265}; - rtcp_fb = {kTWCCRtcpFb, "nack", "ccm fir", "nack pli", "goog-remb"}; + rtcp_fb = {kTWCCRtcpFb, kRembRtcpFb, "nack", "ccm fir", "nack pli"}; extmap = { kTWCCExtMap, + kRembExtMap, "urn:ietf:params:rtp-hdrext:toffset", - "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", "urn:3gpp:video-orientation", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type", diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index bcd1783c..a180c5ee 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -701,6 +701,7 @@ public: void setDefaultSetting(TrackType type); void enableTWCC(bool enable = true); + void enableREMB(bool enable = true); }; RtcTrackConfigure video; From c77c4cfd4dd33c1f1b779abb5829ef4752437dd3 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 30 Apr 2021 15:15:35 +0800 Subject: [PATCH 137/218] =?UTF-8?q?sdp=E7=9B=B8=E5=85=B3=E7=89=B9=E6=80=A7?= =?UTF-8?q?=E6=9A=B4=E9=9C=B2=E6=88=90=E5=B8=B8=E9=87=8F?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 36 ++++++++++++++++++------------------ webrtc/Sdp.h | 12 ++++++++++++ webrtc/WebRtcTransport.cpp | 2 +- 3 files changed, 31 insertions(+), 19 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 3c137fde..987d095f 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1250,28 +1250,28 @@ bool RtcSession::supportRtcpFb(const string &name, TrackType type) const { return ref.find(name) != ref.end(); } -static string const kTWCCRtcpFb = "transport-cc"; -static string const kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; -static string const kRembRtcpFb = "goog-remb"; -static string const kRembExtMap = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; +string const SdpConst::kTWCCRtcpFb = "transport-cc"; +string const SdpConst::kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; +string const SdpConst::kRembRtcpFb = "goog-remb"; +string const SdpConst::kRembExtMap = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ if (!enable) { - rtcp_fb.erase(kTWCCRtcpFb); - extmap.erase(kTWCCExtMap); + rtcp_fb.erase(SdpConst::kTWCCRtcpFb); + extmap.erase(SdpConst::kTWCCExtMap); } else { - rtcp_fb.emplace(kTWCCRtcpFb); - extmap.emplace(kTWCCExtMap); + rtcp_fb.emplace(SdpConst::kTWCCRtcpFb); + extmap.emplace(SdpConst::kTWCCExtMap); } } void RtcConfigure::RtcTrackConfigure::enableREMB(bool enable){ if (!enable) { - rtcp_fb.erase(kRembRtcpFb); - extmap.erase(kRembExtMap); + rtcp_fb.erase(SdpConst::kRembRtcpFb); + extmap.erase(SdpConst::kRembExtMap); } else { - rtcp_fb.emplace(kRembRtcpFb); - extmap.emplace(kRembExtMap); + rtcp_fb.emplace(SdpConst::kRembRtcpFb); + extmap.emplace(SdpConst::kRembExtMap); } } @@ -1290,10 +1290,10 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ case TrackAudio: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; - rtcp_fb = {kTWCCRtcpFb, kRembRtcpFb}; + rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb}; extmap = { - kTWCCExtMap, - kRembExtMap, + SdpConst::kTWCCExtMap, + SdpConst::kRembExtMap, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "urn:ietf:params:rtp-hdrext:sdes:mid", "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", @@ -1304,10 +1304,10 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ case TrackVideo: { //此处调整偏好的编码格式优先级 preferred_codec = {CodecH264, CodecH265}; - rtcp_fb = {kTWCCRtcpFb, kRembRtcpFb, "nack", "ccm fir", "nack pli"}; + rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli"}; extmap = { - kTWCCExtMap, - kRembExtMap, + SdpConst::kTWCCExtMap, + SdpConst::kRembExtMap, "urn:ietf:params:rtp-hdrext:toffset", "urn:3gpp:video-orientation", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index a180c5ee..fa7f31a3 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -730,5 +730,17 @@ private: RtcCodecPlan::Ptr _rtsp_audio_plan; }; +class SdpConst { +public: + static string const kTWCCRtcpFb; + static string const kTWCCExtMap; + static string const kRembRtcpFb; + static string const kRembExtMap; + +private: + SdpConst() = delete; + ~SdpConst() = delete; +}; + #endif //ZLMEDIAKIT_SDP_H diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index cfe70132..484400db 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -623,7 +623,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr //开启remb，则发送remb包调节比特率 GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); - if (remb_bit_rate) { + if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) { sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); } } From 521d9f906b921b219078b39a2cd52add202d3701 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 6 May 2021 10:40:21 +0800 Subject: [PATCH 138/218] =?UTF-8?q?=E5=AE=8C=E5=96=84remb=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E6=8E=A7=E5=88=B6=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 23 +++++++++++++++++++++++ webrtc/Sdp.h | 1 + webrtc/WebRtcTransport.cpp | 8 +++++++- 3 files changed, 31 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 987d095f..9abe83b2 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1394,6 +1394,29 @@ void RtcConfigure::enableTWCC(bool enable, TrackType type){ } } +void RtcConfigure::enableREMB(bool enable, TrackType type){ + switch (type) { + case TrackAudio: { + audio.enableREMB(enable); + break; + } + case TrackVideo: { + video.enableREMB(enable); + break; + } + case TrackApplication: { + application.enableREMB(enable); + break; + } + default: { + audio.enableREMB(enable); + video.enableREMB(enable); + application.enableREMB(enable); + break; + } + } +} + shared_ptr RtcConfigure::createAnswer(const RtcSession &offer){ shared_ptr ret = std::make_shared(); ret->version = offer.version; diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index fa7f31a3..cd044e23 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -719,6 +719,7 @@ public: void setPlayRtspInfo(const string &sdp); void enableTWCC(bool enable = true, TrackType type = TrackInvalid); + void enableREMB(bool enable = true, TrackType type = TrackInvalid); private: void matchMedia(shared_ptr &ret, TrackType type, const vector &medias, const RtcTrackConfigure &configure); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 484400db..7ce870ae 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -187,7 +187,13 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const { //开启remb后关闭twcc，因为开启twcc后remb无效 GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); - configure.enableTWCC(!remb_bit_rate); + if (remb_bit_rate) { + configure.enableREMB(true); + configure.enableTWCC(false); + } else { + configure.enableREMB(false); + configure.enableTWCC(true); + } } std::string WebRtcTransport::getAnswerSdp(const string &offer){ From f9ebd82e6d7150bbe0abfc056d2c6caa35649417 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 6 May 2021 10:54:06 +0800 Subject: [PATCH 139/218] =?UTF-8?q?=E6=94=AF=E6=8C=81=E6=9B=B4=E5=A4=9Artp?= =?UTF-8?q?=20ext?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 9abe83b2..1c1f0fed 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1297,7 +1297,8 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "urn:ietf:params:rtp-hdrext:sdes:mid", "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", - "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id" + "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id", + "urn:ietf:params:rtp-hdrext:csrc-audio-level" }; break; } From cbee3210a0abee763358ca13b86e938b478017af Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 6 May 2021 12:01:56 +0800 Subject: [PATCH 140/218] =?UTF-8?q?=E5=AE=8C=E5=96=84twcc=20fci?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index dc48be52..d0f4fb62 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -392,7 +392,6 @@ static int16_t getRecvDelta(SymbolStatus status, uint8_t *&ptr, const uint8_t *e switch (status) { case SymbolStatus::not_received : { //丢包， recv delta为0个字节 - delta = 0; break; } case SymbolStatus::small_delta : { @@ -409,6 +408,10 @@ static int16_t getRecvDelta(SymbolStatus status, uint8_t *&ptr, const uint8_t *e ptr += 2; break; } + case SymbolStatus::reserved : { + //没有时间戳 + break; + } default: //这个逻辑分支不可达到 CHECK(0); From 842257edaaa4b4a864aa5aacc4372dda328b608c Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 6 May 2021 12:02:16 +0800 Subject: [PATCH 141/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtcp=20ext?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtsp/Rtsp.cpp | 13 +++++++++++-- src/Rtsp/Rtsp.h | 2 ++ 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/src/Rtsp/Rtsp.cpp b/src/Rtsp/Rtsp.cpp index 73f84434..bd2f165c 100644 --- a/src/Rtsp/Rtsp.cpp +++ b/src/Rtsp/Rtsp.cpp @@ -458,18 +458,27 @@ size_t RtpHeader::getExtSize() const { return 0; } auto ext_ptr = &payload + getCsrcSize(); - uint16_t reserved = AV_RB16(ext_ptr); + //uint16_t reserved = AV_RB16(ext_ptr); //每个ext占用4字节 return AV_RB16(ext_ptr + 2) << 2; } +uint16_t RtpHeader::getExtReserved() const{ + //rtp有ext + if (!ext) { + return 0; + } + auto ext_ptr = &payload + getCsrcSize(); + return AV_RB16(ext_ptr); +} + uint8_t *RtpHeader::getExtData() { if (!ext) { return nullptr; } auto ext_ptr = &payload + getCsrcSize(); //多出的4个字节分别为reserved、ext_len - return ext_ptr + 4 + getExtSize(); + return ext_ptr + 4; } size_t RtpHeader::getPayloadOffset() const { diff --git a/src/Rtsp/Rtsp.h b/src/Rtsp/Rtsp.h index 20307f45..aa3b81fb 100644 --- a/src/Rtsp/Rtsp.h +++ b/src/Rtsp/Rtsp.h @@ -119,6 +119,8 @@ public: //返回ext字段字节长度 size_t getExtSize() const; + //返回ext reserved值 + uint16_t getExtReserved() const; //返回ext段首地址，不存在时返回nullptr uint8_t *getExtData(); From d337b8d86478dd86667fd109a217521336a97c74 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Thu, 6 May 2021 16:25:18 +0800 Subject: [PATCH 142/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0rtp=20ext=E8=A7=A3?= =?UTF-8?q?=E6=9E=90=E7=9B=B8=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 157 ++++++++++++++++++++++++++++++++++++++++++++++ webrtc/RtpExt.h | 29 +++++++++ 2 files changed, 186 insertions(+) create mode 100644 webrtc/RtpExt.cpp create mode 100644 webrtc/RtpExt.h diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp new file mode 100644 index 00000000..b28306f0 --- /dev/null +++ b/webrtc/RtpExt.cpp @@ -0,0 +1,157 @@ +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#include "RtpExt.h" + +#if defined(_WIN32) +#pragma pack(push, 1) +#endif // defined(_WIN32) + +//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 +//https://tools.ietf.org/html/rfc5285 + +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | 0xBE | 0xDE | length=3 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | L=0 | data | ID | L=1 | data... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// ...data | 0 (pad) | 0 (pad) | ID | L=3 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | data | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class RtpExtOneByte { +public: + static constexpr uint16_t kMinSize = 1; + size_t getSize() const; + uint8_t getId() const; + uint8_t* getData(); + +private: +#if __BYTE_ORDER == __BIG_ENDIAN + uint8_t id: 4; + uint8_t len: 4; +#else + uint8_t len: 4; + uint8_t id: 4; +#endif + uint8_t data[1]; +} PACKED; + +//0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | 0x100 |appbits| length=3 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | L=0 | ID | L=1 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | data | 0 (pad) | ID | L=4 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | data | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +class RtpExtTwoByte { +public: + static constexpr uint16_t kMinSize = 2; + + size_t getSize() const; + uint8_t getId() const; + uint8_t* getData(); + +private: + uint8_t id; + uint8_t len; + uint8_t data[1]; +} PACKED; + +#if defined(_WIN32) +#pragma pack(pop) +#endif // defined(_WIN32) + +////////////////////////////////////////////////////////////////// + +size_t RtpExtOneByte::getSize() const { + return len + 1; +} + +uint8_t RtpExtOneByte::getId() const { + return id; +} + +uint8_t *RtpExtOneByte::getData() { + return data; +} + +////////////////////////////////////////////////////////////////// + +size_t RtpExtTwoByte::getSize() const { + return len; +} + +uint8_t RtpExtTwoByte::getId() const { + return id; +} + +uint8_t *RtpExtTwoByte::getData() { + return data; +} + +////////////////////////////////////////////////////////////////// + +static constexpr uint16_t kOneByteHeader = 0xBEDE; +static constexpr uint16_t kTwoByteHeader = 0x1000; + +map RtpExt::getExtValue(const RtpHeader *header) { + map ret; + assert(header); + auto ext_size = header->getExtSize(); + if (!ext_size) { + return ret; + } + auto reserved = header->getExtReserved(); + auto ptr = const_cast(header)->getExtData(); + auto end = ptr + ext_size; + + if (reserved == kOneByteHeader) { + while (ptr < end) { + RtpExtOneByte *ext = reinterpret_cast(ptr); + if (ext->getId() == 0) { + //padding，忽略 + ++ptr; + continue; + } + //15类型的rtp ext为保留 + CHECK(ext->getId() != 15); + CHECK(reinterpret_cast(ext) + RtpExtOneByte::kMinSize <= end); + CHECK(ext->getData() + ext->getSize() <= end); + ret.emplace(ext->getId(), string(reinterpret_cast(ext->getData()), ext->getSize())); + ptr += RtpExtOneByte::kMinSize + ext->getSize(); + } + return ret; + } + + if ((reserved & 0xFFF0) >> 4 == kTwoByteHeader) { + while (ptr < end) { + RtpExtTwoByte *ext = reinterpret_cast(ptr); + if (ext->getId() == 0) { + //padding，忽略 + ++ptr; + continue; + } + CHECK(reinterpret_cast(ext) + RtpExtTwoByte::kMinSize <= end); + CHECK(ext->getData() + ext->getSize() <= end); + ret.emplace(ext->getId(), string(reinterpret_cast(ext->getData()), ext->getSize())); + ptr += RtpExtTwoByte::kMinSize + ext->getSize(); + } + return ret; + } + + return ret; +} diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h new file mode 100644 index 00000000..92825e40 --- /dev/null +++ b/webrtc/RtpExt.h @@ -0,0 +1,29 @@ +/* + * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. + * + * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef ZLMEDIAKIT_RTPEXT_H +#define ZLMEDIAKIT_RTPEXT_H + +#include +#include +#include +#include "Common/macros.h" +#include "Rtsp/Rtsp.h" + +using namespace std; +using namespace mediakit; + +class RtpExt { +public: + static map getExtValue(const RtpHeader *header); +}; + + +#endif //ZLMEDIAKIT_RTPEXT_H From b2be30fd51d3f1c1e84f74f655163430c1b8bb77 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 11:19:12 +0800 Subject: [PATCH 143/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0bom=E5=A4=B4?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 2 +- webrtc/RtpExt.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index b28306f0..7fb69cd1 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -1,4 +1,4 @@ -/* +/* * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 92825e40..23c04d94 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -1,4 +1,4 @@ -/* +/* * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). From 2e8a7be5eadf540f49382228399240a43d2fecca Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 12:00:26 +0800 Subject: [PATCH 144/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtp=20ext=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 58 ++++++++++++++++++++++++++++++++++++++++------- webrtc/RtpExt.h | 32 ++++++++++++++++++++++++-- 2 files changed, 80 insertions(+), 10 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 7fb69cd1..83ce7bbb 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -108,8 +108,8 @@ uint8_t *RtpExtTwoByte::getData() { static constexpr uint16_t kOneByteHeader = 0xBEDE; static constexpr uint16_t kTwoByteHeader = 0x1000; -map RtpExt::getExtValue(const RtpHeader *header) { - map ret; +map RtpExt::getExtValue(const RtpHeader *header, const RtcMedia &media) { + map ret; assert(header); auto ext_size = header->getExtSize(); if (!ext_size) { @@ -118,20 +118,20 @@ map RtpExt::getExtValue(const RtpHeader *header) auto reserved = header->getExtReserved(); auto ptr = const_cast(header)->getExtData(); auto end = ptr + ext_size; - + RtpExtType type; if (reserved == kOneByteHeader) { while (ptr < end) { RtpExtOneByte *ext = reinterpret_cast(ptr); - if (ext->getId() == 0) { + if (ext->getId() == (uint8_t) RtpExtType::padding) { //padding，忽略 ++ptr; continue; } //15类型的rtp ext为保留 - CHECK(ext->getId() != 15); + CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); CHECK(reinterpret_cast(ext) + RtpExtOneByte::kMinSize <= end); CHECK(ext->getData() + ext->getSize() <= end); - ret.emplace(ext->getId(), string(reinterpret_cast(ext->getData()), ext->getSize())); + ret.emplace(ext->getId(), RtpExt(type, reinterpret_cast(ext->getData()), ext->getSize())); ptr += RtpExtOneByte::kMinSize + ext->getSize(); } return ret; @@ -140,14 +140,16 @@ map RtpExt::getExtValue(const RtpHeader *header) if ((reserved & 0xFFF0) >> 4 == kTwoByteHeader) { while (ptr < end) { RtpExtTwoByte *ext = reinterpret_cast(ptr); - if (ext->getId() == 0) { + if (ext->getId() == (uint8_t) RtpExtType::padding) { //padding，忽略 ++ptr; continue; } + //15类型的rtp ext为保留 + CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); CHECK(reinterpret_cast(ext) + RtpExtTwoByte::kMinSize <= end); CHECK(ext->getData() + ext->getSize() <= end); - ret.emplace(ext->getId(), string(reinterpret_cast(ext->getData()), ext->getSize())); + ret.emplace(ext->getId(), RtpExt(type, reinterpret_cast(ext->getData()), ext->getSize())); ptr += RtpExtTwoByte::kMinSize + ext->getSize(); } return ret; @@ -155,3 +157,43 @@ map RtpExt::getExtValue(const RtpHeader *header) return ret; } + +#define RTP_EXT_MAP(XX) \ + XX(RtpExtType::ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \ + XX(RtpExtType::abs_send_time, "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time") \ + XX(RtpExtType::transport_cc, "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01") \ + XX(RtpExtType::sdes_mid, "urn:ietf:params:rtp-hdrext:sdes:mid") \ + XX(RtpExtType::sdes_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") \ + XX(RtpExtType::sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \ + XX(RtpExtType::video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \ + XX(RtpExtType::color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \ + XX(RtpExtType::video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \ + XX(RtpExtType::playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \ + XX(RtpExtType::video_orientation, "urn:3gpp:video-orientation") \ + XX(RtpExtType::toffset, "urn:ietf:params:rtp-hdrext:toffset") + +#define XX(type, url) {type , url}, +static unordered_map s_type_to_url = {RTP_EXT_MAP(XX)}; +#undef XX + + +#define XX(type, url) {url, type}, +static unordered_map s_url_to_type = {RTP_EXT_MAP(XX)}; +#undef XX + +RtpExtType RtpExt::getRtpExtType(const string &url) { + auto it = s_url_to_type.find(url); + if (it == s_url_to_type.end()) { + throw std::invalid_argument(string("未识别的rtp ext url类型:") + url); + } + return it->second; +} + +const string &RtpExt::getRtpExtUrl(RtpExtType type) { + auto it = s_type_to_url.find(type); + if (it == s_type_to_url.end()) { + throw std::invalid_argument(string("未识别的rtp ext类型:") + to_string((int) type)); + } + return it->second; +} + diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 23c04d94..c89ee560 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -20,9 +20,37 @@ using namespace std; using namespace mediakit; -class RtpExt { +enum class RtpExtType : uint8_t { + padding = 0, + ssrc_audio_level = 1, + abs_send_time = 2, + transport_cc = 3, + sdes_mid = 4, + sdes_rtp_stream_id = 5, + sdes_repaired_rtp_stream_id = 6, + video_timing = 7, + color_space = 8, + video_content_type = 11, + playout_delay = 12, + video_orientation = 13, + toffset = 14, + reserved = 15, +}; + +class RtcMedia; + +class RtpExt : public std::string { public: - static map getExtValue(const RtpHeader *header); + ~RtpExt() = default; + static map getExtValue(const RtpHeader *header, const RtcMedia &media); + static RtpExtType getRtpExtType(const string &url); + static const string& getRtpExtUrl(RtpExtType type); + +private: + RtpExt(RtpExtType type, const char *str, size_t size) : std::string(str, size), _type(type) {} + +private: + RtpExtType _type; }; From 9f25392f69285dc35a796291cf23fbac393f8bd2 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 14:01:24 +0800 Subject: [PATCH 145/218] =?UTF-8?q?=E7=B2=BE=E7=AE=80api=E5=91=BD=E5=90=8D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 4 ++-- webrtc/RtpExt.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 83ce7bbb..1be988a9 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -181,7 +181,7 @@ static unordered_map s_type_to_url = {RTP_EXT_M static unordered_map s_url_to_type = {RTP_EXT_MAP(XX)}; #undef XX -RtpExtType RtpExt::getRtpExtType(const string &url) { +RtpExtType RtpExt::getExtType(const string &url) { auto it = s_url_to_type.find(url); if (it == s_url_to_type.end()) { throw std::invalid_argument(string("未识别的rtp ext url类型:") + url); @@ -189,7 +189,7 @@ RtpExtType RtpExt::getRtpExtType(const string &url) { return it->second; } -const string &RtpExt::getRtpExtUrl(RtpExtType type) { +const string &RtpExt::getExtUrl(RtpExtType type) { auto it = s_type_to_url.find(type); if (it == s_type_to_url.end()) { throw std::invalid_argument(string("未识别的rtp ext类型:") + to_string((int) type)); diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index c89ee560..05aa85c5 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -43,8 +43,8 @@ class RtpExt : public std::string { public: ~RtpExt() = default; static map getExtValue(const RtpHeader *header, const RtcMedia &media); - static RtpExtType getRtpExtType(const string &url); - static const string& getRtpExtUrl(RtpExtType type); + static RtpExtType getExtType(const string &url); + static const string& getExtUrl(RtpExtType type); private: RtpExt(RtpExtType type, const char *str, size_t size) : std::string(str, size), _type(type) {} From cd96267dc8530881a8dc36b728ea0ad469a6a436 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 14:02:03 +0800 Subject: [PATCH 146/218] =?UTF-8?q?extmap=E7=9B=B8=E5=85=B3=E8=AE=BE?= =?UTF-8?q?=E7=BD=AE=E6=94=B9=E6=88=90=E6=9E=9A=E4=B8=BE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 47 ++++++++++++++++++++++------------------------- webrtc/Sdp.h | 5 ++--- 2 files changed, 24 insertions(+), 28 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 1c1f0fed..6adc4451 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1251,27 +1251,25 @@ bool RtcSession::supportRtcpFb(const string &name, TrackType type) const { } string const SdpConst::kTWCCRtcpFb = "transport-cc"; -string const SdpConst::kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; string const SdpConst::kRembRtcpFb = "goog-remb"; -string const SdpConst::kRembExtMap = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ if (!enable) { rtcp_fb.erase(SdpConst::kTWCCRtcpFb); - extmap.erase(SdpConst::kTWCCExtMap); + extmap.erase(RtpExtType::abs_send_time); } else { rtcp_fb.emplace(SdpConst::kTWCCRtcpFb); - extmap.emplace(SdpConst::kTWCCExtMap); + extmap.emplace(RtpExtType::transport_cc); } } void RtcConfigure::RtcTrackConfigure::enableREMB(bool enable){ if (!enable) { rtcp_fb.erase(SdpConst::kRembRtcpFb); - extmap.erase(SdpConst::kRembExtMap); + extmap.erase(RtpExtType::abs_send_time); } else { rtcp_fb.emplace(SdpConst::kRembRtcpFb); - extmap.emplace(SdpConst::kRembExtMap); + extmap.emplace(RtpExtType::transport_cc); } } @@ -1292,13 +1290,12 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus}; rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb}; extmap = { - SdpConst::kTWCCExtMap, - SdpConst::kRembExtMap, - "urn:ietf:params:rtp-hdrext:ssrc-audio-level", - "urn:ietf:params:rtp-hdrext:sdes:mid", - "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", - "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id", - "urn:ietf:params:rtp-hdrext:csrc-audio-level" + RtpExtType::ssrc_audio_level, + RtpExtType::abs_send_time, + RtpExtType::transport_cc, + RtpExtType::sdes_mid, + RtpExtType::sdes_rtp_stream_id, + RtpExtType::sdes_repaired_rtp_stream_id }; break; } @@ -1307,17 +1304,17 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ preferred_codec = {CodecH264, CodecH265}; rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli"}; extmap = { - SdpConst::kTWCCExtMap, - SdpConst::kRembExtMap, - "urn:ietf:params:rtp-hdrext:toffset", - "urn:3gpp:video-orientation", - "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", - "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type", - "http://www.webrtc.org/experiments/rtp-hdrext/video-timing", - "http://www.webrtc.org/experiments/rtp-hdrext/color-space", - "urn:ietf:params:rtp-hdrext:sdes:mid", - "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", - "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id" + RtpExtType::abs_send_time, + RtpExtType::transport_cc, + RtpExtType::sdes_mid, + RtpExtType::sdes_rtp_stream_id, + RtpExtType::sdes_repaired_rtp_stream_id, + RtpExtType::video_timing, + RtpExtType::color_space, + RtpExtType::video_content_type, + RtpExtType::playout_delay, + RtpExtType::video_orientation, + RtpExtType::toffset }; break; } @@ -1566,7 +1563,7 @@ RETRY: //对方和我方都支持的扩展，那么我们才支持 for (auto &ext : offer_media.extmap) { - if (configure.extmap.find(ext.ext) != configure.extmap.end()) { + if (configure.extmap.find(RtpExt::getExtType(ext.ext)) != configure.extmap.end()) { answer_media.extmap.emplace_back(ext); } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index cd044e23..2195ccaf 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -13,6 +13,7 @@ #include #include +#include "RtpExt.h" #include "assert.h" #include "Extension/Frame.h" #include "Common/Parser.h" @@ -695,7 +696,7 @@ public: SdpAttrFingerprint fingerprint; set rtcp_fb; - set extmap; + set extmap; vector preferred_codec; vector candidate; @@ -734,9 +735,7 @@ private: class SdpConst { public: static string const kTWCCRtcpFb; - static string const kTWCCExtMap; static string const kRembRtcpFb; - static string const kRembExtMap; private: SdpConst() = delete; From c5fff23040e08bc547f37f23e779cd37d9b5491b Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 14:08:43 +0800 Subject: [PATCH 147/218] =?UTF-8?q?extmap=E6=94=B9=E7=94=A8map=E5=AD=98?= =?UTF-8?q?=E6=94=BE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 26 ++++++++++++++++---------- webrtc/Sdp.h | 4 ++-- 2 files changed, 18 insertions(+), 12 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 6adc4451..830cd52a 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -535,8 +535,8 @@ string SdpAttrSetup::toString() const { void SdpAttrExtmap::parse(const string &str) { char buf[128] = {0}; char direction_buf[32] = {0}; - if (sscanf(str.data(), "%" SCNd32 "/%31[^ ] %127s", &index, direction_buf, buf) != 3) { - if (sscanf(str.data(), "%" SCNd32 " %127s", &index, buf) != 2) { + if (sscanf(str.data(), "%" SCNd8 "/%31[^ ] %127s", &id, direction_buf, buf) != 3) { + if (sscanf(str.data(), "%" SCNd8 " %127s", &id, buf) != 2) { SDP_THROW(); } direction = RtpDirection::sendrecv; @@ -549,9 +549,9 @@ void SdpAttrExtmap::parse(const string &str) { string SdpAttrExtmap::toString() const { if (value.empty()) { if(direction == RtpDirection::invalid || direction == RtpDirection::sendrecv){ - value = to_string(index) + " " + ext; + value = to_string((int)id) + " " + ext; } else { - value = to_string(index) + "/" + getRtpDirectionString(direction) + " " + ext; + value = to_string((int)id) + "/" + getRtpDirectionString(direction) + " " + ext; } } return SdpItem::toString(); @@ -806,7 +806,13 @@ void RtcSession::loadFrom(const string &str, bool check) { } rtc_media.rtcp_addr = media.getItemClass('a', "rtcp"); rtc_media.direction = media.getDirection(); - rtc_media.extmap = media.getAllItem('a', "extmap"); + { + rtc_media.extmap.clear(); + auto arr = media.getAllItem('a', "extmap"); + for (auto &ext : arr) { + rtc_media.extmap.emplace(ext.id, ext); + } + } rtc_media.rtcp_mux = media.getItem('a', "rtcp-mux").operator bool(); rtc_media.rtcp_rsize = media.getItem('a', "rtcp-rsize").operator bool(); @@ -1055,8 +1061,8 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ if (m.ice_lite) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-lite"))); } - for (auto &ext : m.extmap) { - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(ext))); + for (auto &pr : m.extmap) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(pr.second))); } if (m.direction != RtpDirection::invalid) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.direction))); @@ -1562,9 +1568,9 @@ RETRY: } //对方和我方都支持的扩展，那么我们才支持 - for (auto &ext : offer_media.extmap) { - if (configure.extmap.find(RtpExt::getExtType(ext.ext)) != configure.extmap.end()) { - answer_media.extmap.emplace_back(ext); + for (auto &pr : offer_media.extmap) { + if (configure.extmap.find(RtpExt::getExtType(pr.second.ext)) != configure.extmap.end()) { + answer_media.extmap.emplace(pr); } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 2195ccaf..bfff21eb 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -337,7 +337,7 @@ class SdpAttrExtmap : public SdpItem { public: //https://aggresss.blog.csdn.net/article/details/106436703 //a=extmap:1[/sendonly] urn:ietf:params:rtp-hdrext:ssrc-audio-level - uint32_t index; + uint8_t id; RtpDirection direction{RtpDirection::invalid}; string ext; void parse(const string &str) override; @@ -636,7 +636,7 @@ public: SdpAttrFingerprint fingerprint; //////// extmap //////// - vector extmap; + map extmap; //////// sctp //////////// SdpAttrSctpMap sctpmap; From f263dd5d9d5eae63fe0042ce9dd07953126635a1 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 14:56:01 +0800 Subject: [PATCH 148/218] =?UTF-8?q?=E4=BC=98=E5=8C=96rtp=20ext=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 107 ++++++++++++++++++++++++++-------------------- webrtc/RtpExt.h | 9 ++-- 2 files changed, 66 insertions(+), 50 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 1be988a9..9d291cdc 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -9,6 +9,7 @@ */ #include "RtpExt.h" +#include "Sdp.h" #if defined(_WIN32) #pragma pack(push, 1) @@ -108,8 +109,35 @@ uint8_t *RtpExtTwoByte::getData() { static constexpr uint16_t kOneByteHeader = 0xBEDE; static constexpr uint16_t kTwoByteHeader = 0x1000; -map RtpExt::getExtValue(const RtpHeader *header, const RtcMedia &media) { - map ret; +static RtpExtType getExtTypeById(uint8_t id, const RtcMedia &media){ + auto it = media.extmap.find(id); + if (it == media.extmap.end()) { + return RtpExtType::padding; + } + return RtpExt::getExtType(it->second.ext); +} + +template +static void appendExt( map &ret,const RtcMedia &media, uint8_t *ptr, const uint8_t *end){ + while (ptr < end) { + auto ext = reinterpret_cast(ptr); + if (ext->getId() == (uint8_t) RtpExtType::padding) { + //padding，忽略 + ++ptr; + continue; + } + //15类型的rtp ext为保留 + CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); + CHECK(reinterpret_cast(ext) + Type::kMinSize <= end); + CHECK(ext->getData() + ext->getSize() <= end); + auto type = getExtTypeById(ext->getId(), media); + ret.emplace(type, RtpExt(type, ext->getId(), reinterpret_cast(ext->getData()), ext->getSize())); + ptr += Type::kMinSize + ext->getSize(); + } +} + +map RtpExt::getExtValue(const RtpHeader *header, const RtcMedia &media) { + map ret; assert(header); auto ext_size = header->getExtSize(); if (!ext_size) { @@ -118,66 +146,37 @@ map RtpExt::getExtValue(const RtpHeader *header, auto reserved = header->getExtReserved(); auto ptr = const_cast(header)->getExtData(); auto end = ptr + ext_size; - RtpExtType type; if (reserved == kOneByteHeader) { - while (ptr < end) { - RtpExtOneByte *ext = reinterpret_cast(ptr); - if (ext->getId() == (uint8_t) RtpExtType::padding) { - //padding，忽略 - ++ptr; - continue; - } - //15类型的rtp ext为保留 - CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); - CHECK(reinterpret_cast(ext) + RtpExtOneByte::kMinSize <= end); - CHECK(ext->getData() + ext->getSize() <= end); - ret.emplace(ext->getId(), RtpExt(type, reinterpret_cast(ext->getData()), ext->getSize())); - ptr += RtpExtOneByte::kMinSize + ext->getSize(); - } + appendExt(ret, media, ptr, end); return ret; } - if ((reserved & 0xFFF0) >> 4 == kTwoByteHeader) { - while (ptr < end) { - RtpExtTwoByte *ext = reinterpret_cast(ptr); - if (ext->getId() == (uint8_t) RtpExtType::padding) { - //padding，忽略 - ++ptr; - continue; - } - //15类型的rtp ext为保留 - CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); - CHECK(reinterpret_cast(ext) + RtpExtTwoByte::kMinSize <= end); - CHECK(ext->getData() + ext->getSize() <= end); - ret.emplace(ext->getId(), RtpExt(type, reinterpret_cast(ext->getData()), ext->getSize())); - ptr += RtpExtTwoByte::kMinSize + ext->getSize(); - } + appendExt(ret, media, ptr, end); return ret; } - return ret; } #define RTP_EXT_MAP(XX) \ - XX(RtpExtType::ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \ - XX(RtpExtType::abs_send_time, "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time") \ - XX(RtpExtType::transport_cc, "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01") \ - XX(RtpExtType::sdes_mid, "urn:ietf:params:rtp-hdrext:sdes:mid") \ - XX(RtpExtType::sdes_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") \ - XX(RtpExtType::sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \ - XX(RtpExtType::video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \ - XX(RtpExtType::color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \ - XX(RtpExtType::video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \ - XX(RtpExtType::playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \ - XX(RtpExtType::video_orientation, "urn:3gpp:video-orientation") \ - XX(RtpExtType::toffset, "urn:ietf:params:rtp-hdrext:toffset") + XX(ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \ + XX(abs_send_time, "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time") \ + XX(transport_cc, "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01") \ + XX(sdes_mid, "urn:ietf:params:rtp-hdrext:sdes:mid") \ + XX(sdes_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") \ + XX(sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \ + XX(video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \ + XX(color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \ + XX(video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \ + XX(playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \ + XX(video_orientation, "urn:3gpp:video-orientation") \ + XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") -#define XX(type, url) {type , url}, +#define XX(type, url) {RtpExtType::type , url}, static unordered_map s_type_to_url = {RTP_EXT_MAP(XX)}; #undef XX -#define XX(type, url) {url, type}, +#define XX(type, url) {url, RtpExtType::type}, static unordered_map s_url_to_type = {RTP_EXT_MAP(XX)}; #undef XX @@ -197,3 +196,17 @@ const string &RtpExt::getExtUrl(RtpExtType type) { return it->second; } +const char *RtpExt::getExtName(RtpExtType type) { +#define XX(type, url) case RtpExtType::type: return #type; + switch (type) { + RTP_EXT_MAP(XX) + default: return "unknown ext type"; + } +#undef XX +} + +string RtpExt::dumpString() const { + _StrPrinter printer; + printer << getExtName(_type) << ", id:" << (int)_id << " " << hexdump(data(), size()); + return std::move(printer); +} diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 05aa85c5..742ebd30 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -41,16 +41,19 @@ class RtcMedia; class RtpExt : public std::string { public: + RtpExt(RtpExtType type, uint8_t id, const char *str, size_t size) : std::string(str, size), _type(type), _id(id) {} ~RtpExt() = default; - static map getExtValue(const RtpHeader *header, const RtcMedia &media); + + static map getExtValue(const RtpHeader *header, const RtcMedia &media); static RtpExtType getExtType(const string &url); static const string& getExtUrl(RtpExtType type); + static const char *getExtName(RtpExtType type); -private: - RtpExt(RtpExtType type, const char *str, size_t size) : std::string(str, size), _type(type) {} + string dumpString() const; private: RtpExtType _type; + uint8_t _id; }; From 0b8eb6a76f92737091a1b8a5dbf931c31fc6dd29 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 17:47:53 +0800 Subject: [PATCH 149/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtp=20ext=E8=A7=A3?= =?UTF-8?q?=E6=9E=90=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 292 +++++++++++++++++++++++++++++++++++++++++++++- webrtc/RtpExt.h | 34 ++++++ webrtc/Sdp.cpp | 4 +- 3 files changed, 327 insertions(+), 3 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 9d291cdc..e1e181a6 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -166,10 +166,13 @@ map RtpExt::getExtValue(const RtpHeader *hea XX(sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \ XX(video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \ XX(color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \ + XX(csrc_audio_level, "urn:ietf:params:rtp-hdrext:csrc-audio-level") \ + XX(framemarking, "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07") \ XX(video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \ XX(playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \ XX(video_orientation, "urn:3gpp:video-orientation") \ - XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") + XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") \ + XX(encrypt, "urn:ietf:params:rtp-hdrext:encrypt") #define XX(type, url) {RtpExtType::type , url}, static unordered_map s_type_to_url = {RTP_EXT_MAP(XX)}; @@ -207,6 +210,291 @@ const char *RtpExt::getExtName(RtpExtType type) { string RtpExt::dumpString() const { _StrPrinter printer; - printer << getExtName(_type) << ", id:" << (int)_id << " " << hexdump(data(), size()); + switch (_type) { + case RtpExtType::ssrc_audio_level : { + bool vad; + printer << "audio level:" << (int) getAudioLevel(&vad) << ", vad:" << vad; + break; + } + case RtpExtType::abs_send_time : { + printer << "abs send time:" << getAbsSendTime(); + break; + } + case RtpExtType::transport_cc : { + printer << "twcc seq:" << getTransportCCSeq(); + break; + } + case RtpExtType::sdes_mid : { + printer << "sdes mid:" << getSdesMid(); + break; + } + case RtpExtType::sdes_rtp_stream_id : { + printer << "rtp stream id:" << getRtpStreamId(); + break; + } + case RtpExtType::sdes_repaired_rtp_stream_id : { + printer << "rtp repaired stream id:" << getRepairedRtpStreamId(); + break; + } + case RtpExtType::video_timing : { + uint8_t flags; + uint16_t encode_start, encode_finish, packetization_complete, last_pkt_left_pacer, reserved_net0, reserved_net1; + getVideoTiming(flags, encode_start, encode_finish, packetization_complete, last_pkt_left_pacer, + reserved_net0, reserved_net1); + printer << "video timing, flags:" << (int) flags + << ",encode:" << encode_start << "-" << encode_finish + << ",packetization_complete:" << packetization_complete + << ",last_pkt_left_pacer:" << last_pkt_left_pacer + << ",reserved_net0:" << reserved_net0 + << ",reserved_net1:" << reserved_net1; + break; + } + case RtpExtType::video_content_type : { + printer << "video content type:" << (int)getVideoContentType(); + break; + } + case RtpExtType::video_orientation : { + bool camera_bit, flip_bit, first_rotation, second_rotation; + getVideoOrientation(camera_bit, flip_bit, first_rotation, second_rotation); + printer << "video orientation:" << camera_bit << "-" << flip_bit << "-" << first_rotation << "-" << second_rotation; + break; + } + case RtpExtType::playout_delay : { + uint16_t min_delay, max_delay; + getPlayoutDelay(min_delay, max_delay); + printer << "playout delay:" << min_delay << "-" << max_delay; + break; + } + case RtpExtType::toffset : { + printer << "toffset:" << getTransmissionOffset(); + break; + } + case RtpExtType::framemarking : { + printer << "framemarking tid:" << getFramemarkingTID(); + break; + } + default: { + printer << getExtName(_type) << ", id:" << (int) _id << ", "; + printer << "hex:" << hexdump(data(), size()); + break; + } + } return std::move(printer); } + +//https://tools.ietf.org/html/rfc6464 +// 0 1 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | len=0 |V| level | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Figure 1: Sample Audio Level Encoding Using the +// One-Byte Header Format +// +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | len=1 |V| level | 0 (pad) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Figure 2: Sample Audio Level Encoding Using the +// Two-Byte Header Format +uint8_t RtpExt::getAudioLevel(bool *vad) const{ + CHECK(_type == RtpExtType::ssrc_audio_level && size() >= 1); + auto &byte = (*this)[0]; + if (vad) { + *vad = byte & 0x80; + } + return byte & 0x7F; +} + +//http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +//Wire format: 1-byte extension, 3 bytes of data. total 4 bytes extra per packet (plus shared 4 bytes for all extensions present: 2 byte magic word 0xBEDE, 2 byte # of extensions). Will in practice replace the “toffset” extension so we should see no long term increase in traffic as a result. +// +//Encoding: Timestamp is in seconds, 24 bit 6.18 fixed point, yielding 64s wraparound and 3.8us resolution (one increment for each 477 bytes going out on a 1Gbps interface). +// +//Relation to NTP timestamps: abs_send_time_24 = (ntp_timestamp_64 >> 14) & 0x00ffffff ; NTP timestamp is 32 bits for whole seconds, 32 bits fraction of second. +// +//Notes: Packets are time stamped when going out, preferably close to metal. Intermediate RTP relays (entities possibly altering the stream) should remove the extension or set its own timestamp. +uint32_t RtpExt::getAbsSendTime() const { + CHECK(_type == RtpExtType::abs_send_time && size() >= 3); + uint32_t ret = 0; + ret |= (*this)[0] << 16; + ret |= (*this)[1] << 8; + ret |= (*this)[2]; + return ret; +} + +//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | 0xBE | 0xDE | length=1 | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | L=1 |transport-wide sequence number | zero padding | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +uint16_t RtpExt::getTransportCCSeq() const { + CHECK(_type == RtpExtType::transport_cc && size() >= 2); + uint16_t ret; + ret |= (*this)[0] << 8; + ret |= (*this)[1]; + return ret; +} + +//https://tools.ietf.org/html/draft-ietf-avtext-sdes-hdr-ext-07 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | len | SDES Item text value ... | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +const string &RtpExt::getSdesMid() const { + CHECK(_type == RtpExtType::sdes_mid && size() >= 1); + return *this; +} + + +//https://tools.ietf.org/html/draft-ietf-avtext-rid-06 +//用于simulecast +//3.1. RTCP 'RtpStreamId' SDES Extension +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |RtpStreamId=TBD| length | RtpStreamId ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// +// The RtpStreamId payload is UTF-8 encoded and is not null-terminated. +// +// RFC EDITOR NOTE: Please replace TBD with the assigned SDES +// identifier value. + +//3.2. RTCP 'RepairedRtpStreamId' SDES Extension +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |Repaired...=TBD| length | RepairRtpStreamId ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// +// The RepairedRtpStreamId payload is UTF-8 encoded and is not null- +// terminated. +// +// RFC EDITOR NOTE: Please replace TBD with the assigned SDES +// identifier value. + +string RtpExt::getRtpStreamId() const { + CHECK(_type == RtpExtType::sdes_rtp_stream_id && size() >= 1); + return *this; +} + +string RtpExt::getRepairedRtpStreamId() const { + CHECK(_type == RtpExtType::sdes_repaired_rtp_stream_id && size() >= 1); + return *this; +} + + +//http://www.webrtc.org/experiments/rtp-hdrext/video-timing +//Wire format: 1-byte extension, 13 bytes of data. Total 14 bytes extra per packet (plus 1-3 padding byte in some cases, plus shared 4 bytes for all extensions present: 2 byte magic word 0xBEDE, 2 byte # of extensions). +// +//First byte is a flags field. Defined flags: +// +//0x01 - extension is set due to timer. +//0x02 - extension is set because the frame is larger than usual. +//Both flags may be set at the same time. All remaining 6 bits are reserved and should be ignored. +// +//Next, 6 timestamps are stored as 16-bit values in big-endian order, representing delta from the capture time of a packet in ms. Timestamps are, in order: +// +//Encode start. +//Encode finish. +//Packetization complete. +//Last packet left the pacer. +//Reserved for network. +//Reserved for network (2). + +void RtpExt::getVideoTiming(uint8_t &flags, + uint16_t &encode_start, + uint16_t &encode_finish, + uint16_t &packetization_complete, + uint16_t &last_pkt_left_pacer, + uint16_t &reserved_net0, + uint16_t &reserved_net1) const { + CHECK(_type == RtpExtType::video_timing && size() >= 13); + flags = (*this)[0]; + encode_start = (*this)[1] << 8 | (*this)[2]; + encode_finish = (*this)[3] << 8 | (*this)[4]; + packetization_complete = (*this)[5] << 8 | (*this)[6]; + last_pkt_left_pacer = (*this)[7] << 8 | (*this)[8]; + reserved_net0 = (*this)[9] << 8 | (*this)[10]; + reserved_net1 = (*this)[11] << 8 | (*this)[12]; +} + + +//http://www.webrtc.org/experiments/rtp-hdrext/color-space +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | L = 3 | primaries | transfer | matrix | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |range+chr.sit. | +// +-+-+-+-+-+-+-+-+ + + +//http://www.webrtc.org/experiments/rtp-hdrext/video-content-type +//Values: +//0x00: Unspecified. Default value. Treated the same as an absence of an extension. +//0x01: Screenshare. Video stream is of a screenshare type. +//0x02: 摄像头？ +//Notes: Extension shoud be present only in the last packet of key-frames. +// If attached to other packets it should be ignored. +// If extension is absent, Unspecified value is assumed. +uint8_t RtpExt::getVideoContentType() const { + CHECK(_type == RtpExtType::video_content_type && size() >= 1); + return (*this)[0]; +} + +//http://www.3gpp.org/ftp/Specs/html-info/26114.htm +void RtpExt::getVideoOrientation(bool &camera_bit, bool &flip_bit, bool &first_rotation, bool &second_rotation) const { + CHECK(_type == RtpExtType::video_orientation && size() >= 1); + uint8_t byte = (*this)[0]; + camera_bit = (byte & 0x08) >> 3; + flip_bit = (byte & 0x04) >> 2; + first_rotation = (byte & 0x02) >> 1; + second_rotation = byte & 0x01; +} + +//http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +//+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +//| ID | len=2 | MIN delay | MAX delay | +//+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +void RtpExt::getPlayoutDelay(uint16_t &min_delay, uint16_t &max_delay) const { + CHECK(_type == RtpExtType::playout_delay && size() >= 3); + uint32_t bytes = (*this)[0] << 16 | (*this)[1] << 8 | (*this)[2]; + min_delay = (bytes & 0x00FFF000) >> 12; + max_delay = bytes & 0x00000FFF; +} + +//urn:ietf:params:rtp-hdrext:toffset +//https://tools.ietf.org/html/rfc5450 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID | len=2 | transmission offset | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +uint32_t RtpExt::getTransmissionOffset() const { + CHECK(_type == RtpExtType::toffset && size() >= 3); + return (*this)[0] << 16 | (*this)[1] << 8 | (*this)[2]; +} + +//http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07 +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | ID=? | L=2 |S|E|I|D|B| TID | LID | TL0PICIDX | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +uint8_t RtpExt::getFramemarkingTID() const { + CHECK(_type == RtpExtType::framemarking && size() >= 3); + return (*this)[0] & 0x07; +} diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 742ebd30..0decaeac 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -30,11 +30,17 @@ enum class RtpExtType : uint8_t { sdes_repaired_rtp_stream_id = 6, video_timing = 7, color_space = 8, + //for firefox + csrc_audio_level = 9, + //svc ? + framemarking = 10, video_content_type = 11, playout_delay = 12, video_orientation = 13, toffset = 14, reserved = 15, + // e2e ? + encrypt }; class RtcMedia; @@ -51,6 +57,34 @@ public: string dumpString() const; + uint8_t getAudioLevel(bool *vad) const; + uint32_t getAbsSendTime() const; + uint16_t getTransportCCSeq() const; + const string& getSdesMid() const; + string getRtpStreamId() const; + string getRepairedRtpStreamId() const; + + void getVideoTiming(uint8_t &flags, + uint16_t &encode_start, + uint16_t &encode_finish, + uint16_t &packetization_complete, + uint16_t &last_pkt_left_pacer, + uint16_t &reserved_net0, + uint16_t &reserved_net1) const; + + uint8_t getVideoContentType() const; + + void getVideoOrientation(bool &camera_bit, + bool &flip_bit, + bool &first_rotation, + bool &second_rotation) const; + + void getPlayoutDelay(uint16_t &min_delay, uint16_t &max_delay) const; + + uint32_t getTransmissionOffset() const; + + uint8_t getFramemarkingTID() const; + private: RtpExtType _type; uint8_t _id; diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 830cd52a..0dbb1fdc 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1297,6 +1297,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb}; extmap = { RtpExtType::ssrc_audio_level, + RtpExtType::csrc_audio_level, RtpExtType::abs_send_time, RtpExtType::transport_cc, RtpExtType::sdes_mid, @@ -1320,7 +1321,8 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ RtpExtType::video_content_type, RtpExtType::playout_delay, RtpExtType::video_orientation, - RtpExtType::toffset + RtpExtType::toffset, + RtpExtType::framemarking }; break; } From cdbd210a10eec51caa408431aa600c010b7f7294 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 17:58:10 +0800 Subject: [PATCH 150/218] =?UTF-8?q?=E9=BB=98=E8=AE=A4=E5=85=B3=E9=97=ADmed?= =?UTF-8?q?iasoup=E7=9A=84=E6=97=A5=E5=BF=97?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/logger.h | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/webrtc/logger.h b/webrtc/logger.h index 67da6219..c708d425 100644 --- a/webrtc/logger.h +++ b/webrtc/logger.h @@ -2,6 +2,7 @@ #include #include +#if 0 #define MS_TRACE() #define MS_ERROR(fmt, ...) printf("error:" fmt "\n", ##__VA_ARGS__) #define MS_THROW_ERROR(fmt, ...) do{ printf("throw:" fmt "\n", ##__VA_ARGS__); throw std::runtime_error("error"); } while(false); @@ -12,4 +13,17 @@ #define MS_ASSERT(con, fmt, ...) do{if(!(con)) { printf("assert failed:%s" fmt "\n", #con, ##__VA_ARGS__);} assert(con); } while(false); #define MS_ABORT(fmt, ...) do{ printf("abort:" fmt "\n", ##__VA_ARGS__); abort(); } while(false); #define MS_WARN_TAG(tag,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__) -#define MS_DEBUG_DEV(fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) \ No newline at end of file +#define MS_DEBUG_DEV(fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__) +#else +#define MS_TRACE() +#define MS_ERROR(fmt, ...) +#define MS_THROW_ERROR(fmt, ...) +#define MS_DUMP(fmt, ...) +#define MS_DEBUG_2TAGS(tag1, tag2,fmt, ...) +#define MS_WARN_2TAGS(tag1, tag2,fmt, ...) +#define MS_DEBUG_TAG(tag,fmt, ...) +#define MS_ASSERT(con, fmt, ...) +#define MS_ABORT(fmt, ...) +#define MS_WARN_TAG(tag,fmt, ...) +#define MS_DEBUG_DEV(fmt, ...) +#endif \ No newline at end of file From e9f11a89fd2aacc40f3f05893389fde223208506 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 18:33:45 +0800 Subject: [PATCH 151/218] =?UTF-8?q?=E6=94=AF=E6=8C=81=E4=BF=AE=E6=94=B9ext?= =?UTF-8?q?=20id?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 44 ++++++++++++++++++++++++++++++++++++++++---- webrtc/RtpExt.h | 15 ++++++++++++--- 2 files changed, 52 insertions(+), 7 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index e1e181a6..10542843 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -34,6 +34,7 @@ public: static constexpr uint16_t kMinSize = 1; size_t getSize() const; uint8_t getId() const; + void setId(uint8_t id); uint8_t* getData(); private: @@ -64,6 +65,7 @@ public: size_t getSize() const; uint8_t getId() const; + void setId(uint8_t id); uint8_t* getData(); private: @@ -86,6 +88,10 @@ uint8_t RtpExtOneByte::getId() const { return id; } +void RtpExtOneByte::setId(uint8_t in) { + id = in & 0x0F; +} + uint8_t *RtpExtOneByte::getData() { return data; } @@ -100,6 +106,10 @@ uint8_t RtpExtTwoByte::getId() const { return id; } +void RtpExtTwoByte::setId(uint8_t in) { + id = in; +} + uint8_t *RtpExtTwoByte::getData() { return data; } @@ -118,7 +128,17 @@ static RtpExtType getExtTypeById(uint8_t id, const RtcMedia &media){ } template -static void appendExt( map &ret,const RtcMedia &media, uint8_t *ptr, const uint8_t *end){ +static bool isOneByteExt(){ + return false; +} + +template<> +static bool isOneByteExt(){ + return true; +} + +template +static void appendExt(map &ret, const RtcMedia &media, uint8_t *ptr, const uint8_t *end) { while (ptr < end) { auto ext = reinterpret_cast(ptr); if (ext->getId() == (uint8_t) RtpExtType::padding) { @@ -131,11 +151,17 @@ static void appendExt( map &ret,const RtcMedia &media, uint8 CHECK(reinterpret_cast(ext) + Type::kMinSize <= end); CHECK(ext->getData() + ext->getSize() <= end); auto type = getExtTypeById(ext->getId(), media); - ret.emplace(type, RtpExt(type, ext->getId(), reinterpret_cast(ext->getData()), ext->getSize())); + ret.emplace(type, RtpExt(ext, isOneByteExt(), type, reinterpret_cast(ext->getData()), ext->getSize())); ptr += Type::kMinSize + ext->getSize(); } } +RtpExt::RtpExt(void *ptr, bool one_byte_ext, RtpExtType type, const char *str, size_t size) : std::string(str, size) { + _ptr = ptr; + _one_byte_ext = one_byte_ext; + _type = type; +} + map RtpExt::getExtValue(const RtpHeader *header, const RtcMedia &media) { map ret; assert(header); @@ -274,8 +300,7 @@ string RtpExt::dumpString() const { break; } default: { - printer << getExtName(_type) << ", id:" << (int) _id << ", "; - printer << "hex:" << hexdump(data(), size()); + printer << getExtName(_type) << ", hex:" << hexdump(data(), size()); break; } } @@ -498,3 +523,14 @@ uint8_t RtpExt::getFramemarkingTID() const { CHECK(_type == RtpExtType::framemarking && size() >= 3); return (*this)[0] & 0x07; } + +void RtpExt::setExtId(uint8_t ext_id) { + assert(ext_id > (int) RtpExtType::padding && ext_id <= (int) RtpExtType::reserved && _ptr); + if (_one_byte_ext) { + auto ptr = reinterpret_cast(_ptr); + ptr->setId(ext_id); + } else { + auto ptr = reinterpret_cast(_ptr); + ptr->setId(ext_id); + } +} diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 0decaeac..2557dc6d 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -40,14 +40,16 @@ enum class RtpExtType : uint8_t { toffset = 14, reserved = 15, // e2e ? - encrypt + encrypt = reserved }; class RtcMedia; class RtpExt : public std::string { public: - RtpExt(RtpExtType type, uint8_t id, const char *str, size_t size) : std::string(str, size), _type(type), _id(id) {} + template + friend void appendExt(map &ret, const RtcMedia &media, uint8_t *ptr, const uint8_t *end); + ~RtpExt() = default; static map getExtValue(const RtpHeader *header, const RtcMedia &media); @@ -85,9 +87,16 @@ public: uint8_t getFramemarkingTID() const; + //危险函数，必须保证关联的RtpHeader对象有效 + void setExtId(uint8_t ext_id); + private: + RtpExt(void *ptr, bool one_byte_ext, RtpExtType type, const char *str, size_t size); + +private: + void *_ptr = nullptr; + bool _one_byte_ext = true; RtpExtType _type; - uint8_t _id; }; From 5756ed46e04097aefbc4152bc77e8a2c804ad182 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Fri, 7 May 2021 18:34:04 +0800 Subject: [PATCH 152/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0ext=E8=B0=83=E8=AF=95?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 7ce870ae..eda327a0 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -10,6 +10,7 @@ #include "WebRtcTransport.h" #include +#include "RtpExt.h" #include "Rtcp/Rtcp.h" #include "Rtcp/RtcpFCI.h" #include "Rtsp/RtpReceiver.h" @@ -187,13 +188,7 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const { //开启remb后关闭twcc，因为开启twcc后remb无效 GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); - if (remb_bit_rate) { - configure.enableREMB(true); - configure.enableTWCC(false); - } else { - configure.enableREMB(false); - configure.enableTWCC(true); - } + configure.enableTWCC(!remb_bit_rate); } std::string WebRtcTransport::getAnswerSdp(const string &offer){ @@ -633,6 +628,19 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); } } + + auto header = rtp->getHeader(); + auto ext_map = RtpExt::getExtValue(header, *(info.media)); + for (auto &pr : ext_map) { + if (rtp->type == TrackVideo) { + InfoL << pr.second.dumpString(); + } else { + DebugL << pr.second.dumpString(); + } + //推流时修改ext id为统一的id，播放时再修改为对方设置的ext id + pr.second.setExtId((uint8_t) pr.first); + } + if (_push_src) { _push_src->onWrite(std::move(rtp), false); } From 9f95f743e70756f9d19808406eb171dccad2ee59 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 10:32:08 +0800 Subject: [PATCH 153/218] =?UTF-8?q?=E8=A7=A3=E6=9E=90rtp=20ext=E6=97=B6?= =?UTF-8?q?=E4=B8=8D=E5=86=8D=E8=B5=8B=E5=80=BC=E7=B1=BB=E5=9E=8B?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 28 +++++++++++----------------- webrtc/RtpExt.h | 9 +++++---- 2 files changed, 16 insertions(+), 21 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 10542843..8189b9e9 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -119,14 +119,6 @@ uint8_t *RtpExtTwoByte::getData() { static constexpr uint16_t kOneByteHeader = 0xBEDE; static constexpr uint16_t kTwoByteHeader = 0x1000; -static RtpExtType getExtTypeById(uint8_t id, const RtcMedia &media){ - auto it = media.extmap.find(id); - if (it == media.extmap.end()) { - return RtpExtType::padding; - } - return RtpExt::getExtType(it->second.ext); -} - template static bool isOneByteExt(){ return false; @@ -138,7 +130,7 @@ static bool isOneByteExt(){ } template -static void appendExt(map &ret, const RtcMedia &media, uint8_t *ptr, const uint8_t *end) { +static void appendExt(map &ret, uint8_t *ptr, const uint8_t *end) { while (ptr < end) { auto ext = reinterpret_cast(ptr); if (ext->getId() == (uint8_t) RtpExtType::padding) { @@ -150,20 +142,18 @@ static void appendExt(map &ret, const RtcMedia &media, uint8 CHECK(ext->getId() < (uint8_t) RtpExtType::reserved); CHECK(reinterpret_cast(ext) + Type::kMinSize <= end); CHECK(ext->getData() + ext->getSize() <= end); - auto type = getExtTypeById(ext->getId(), media); - ret.emplace(type, RtpExt(ext, isOneByteExt(), type, reinterpret_cast(ext->getData()), ext->getSize())); + ret.emplace(ext->getId(), RtpExt(ext, isOneByteExt(), reinterpret_cast(ext->getData()), ext->getSize())); ptr += Type::kMinSize + ext->getSize(); } } -RtpExt::RtpExt(void *ptr, bool one_byte_ext, RtpExtType type, const char *str, size_t size) : std::string(str, size) { +RtpExt::RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size) : std::string(str, size) { _ptr = ptr; _one_byte_ext = one_byte_ext; - _type = type; } -map RtpExt::getExtValue(const RtpHeader *header, const RtcMedia &media) { - map ret; +map RtpExt::getExtValue(const RtpHeader *header) { + map ret; assert(header); auto ext_size = header->getExtSize(); if (!ext_size) { @@ -173,11 +163,11 @@ map RtpExt::getExtValue(const RtpHeader *hea auto ptr = const_cast(header)->getExtData(); auto end = ptr + ext_size; if (reserved == kOneByteHeader) { - appendExt(ret, media, ptr, end); + appendExt(ret, ptr, end); return ret; } if ((reserved & 0xFFF0) >> 4 == kTwoByteHeader) { - appendExt(ret, media, ptr, end); + appendExt(ret, ptr, end); return ret; } return ret; @@ -534,3 +524,7 @@ void RtpExt::setExtId(uint8_t ext_id) { ptr->setId(ext_id); } } + +void RtpExt::setType(RtpExtType type) { + _type = type; +} diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 2557dc6d..d3975410 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -48,15 +48,16 @@ class RtcMedia; class RtpExt : public std::string { public: template - friend void appendExt(map &ret, const RtcMedia &media, uint8_t *ptr, const uint8_t *end); + friend void appendExt(map &ret, uint8_t *ptr, const uint8_t *end); ~RtpExt() = default; - static map getExtValue(const RtpHeader *header, const RtcMedia &media); + static map getExtValue(const RtpHeader *header); static RtpExtType getExtType(const string &url); static const string& getExtUrl(RtpExtType type); static const char *getExtName(RtpExtType type); + void setType(RtpExtType type); string dumpString() const; uint8_t getAudioLevel(bool *vad) const; @@ -91,12 +92,12 @@ public: void setExtId(uint8_t ext_id); private: - RtpExt(void *ptr, bool one_byte_ext, RtpExtType type, const char *str, size_t size); + RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size); private: void *_ptr = nullptr; bool _one_byte_ext = true; - RtpExtType _type; + RtpExtType _type = RtpExtType::padding; }; From d5eb486f5445fbdd1cc39de1202dd2af31252b03 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 10:49:09 +0800 Subject: [PATCH 154/218] =?UTF-8?q?=E4=BF=AE=E6=94=B9rtp=E7=9A=84ext=20id?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 优化代码结构 --- webrtc/WebRtcTransport.cpp | 81 +++++++++++++++++++++++++------------- webrtc/WebRtcTransport.h | 8 ++-- 2 files changed, 59 insertions(+), 30 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index eda327a0..74f544dd 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -390,10 +390,8 @@ bool WebRtcTransportImp::canRecvRtp() const{ } void WebRtcTransportImp::onStartWebRTC() { + //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 for (auto &m : getSdp(SdpType::offer).media) { - if (m.type == TrackVideo) { - _recv_video_ssrc = m.rtp_ssrc.ssrc; - } for (auto &plan : m.plan) { auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); if (!hit_pan) { @@ -413,6 +411,14 @@ void WebRtcTransportImp::onStartWebRTC() { onBeforeSortedRtp(ref, rtp); }); } + if (m.type != TrackApplication) { + //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id + for (auto &pr : m.extmap) { + auto ext_type = RtpExt::getExtType(pr.second.ext); + _rtp_ext_id_to_type.emplace(pr.second.id, ext_type); + _rtp_ext_type_to_id.emplace(ext_type, pr.second.id); + } + } } if (canRecvRtp()) { @@ -431,6 +437,19 @@ void WebRtcTransportImp::onStartWebRTC() { strongSelf->onSendRtp(rtp, ++i == pkt->size()); }); }); + + RtcSession rtsp_send_sdp; + rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); + for (auto &m : getSdp(SdpType::answer).media) { + if (m.type == TrackApplication) { + continue; + } + auto rtsp_media = rtsp_send_sdp.getMedia(m.type); + if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { + //记录发送rtp时约定的pt，届时发送rtp时需要修改pt + _send_rtp_pt[m.type] = m.plan[0].pt; + } + } } } @@ -440,6 +459,7 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ return; } + //修改sdp的ip、端口信息 GET_CONFIG(string, extern_ip, RTC::kExternIP); for (auto &m : sdp.media) { m.addr.reset(); @@ -455,9 +475,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ return; } - RtcSession rtsp_send_sdp; - rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); - for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; @@ -470,11 +487,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc; m.rtx_ssrc.cname = RTX_CNAME; } - auto rtsp_media = rtsp_send_sdp.getMedia(m.type); - if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { - //记录发送rtp的pt - _send_rtp_pt[m.type] = m.plan[0].pt; - } } } @@ -557,6 +569,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { it->second->rtcp_context_recv->onRtcp(sr); auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); + } else { + WarnL << "未识别的sr rtcp包:" << sr->ssrc; } break; } @@ -568,6 +582,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { if (it != _rtp_info_ssrc.end()) { auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); + } else { + WarnL << "未识别的rr rtcp包:" << rr->ssrc; } break; } @@ -577,6 +593,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { for (auto ssrc : bye->getSSRC()) { auto it = _rtp_info_ssrc.find(*ssrc); if (it == _rtp_info_ssrc.end()) { + WarnL << "未识别的bye rtcp包:" << *ssrc; continue; } _rtp_info_pt.erase(it->second->plan->pt); @@ -587,7 +604,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - InfoL << "\n" << rtcp->dumpString(); break; } default: break; @@ -617,36 +633,46 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr //todo rtx/red/ulpfec类型的rtp先未处理 return; } - if (_pli_ticker.elapsedTime() > 2000) { + if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); - sendRtcpPli(_recv_video_ssrc); + sendRtcpPli(rtp->getSSRC()); //开启remb，则发送remb包调节比特率 GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) { - sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); + sendRtcpRemb(rtp->getSSRC(), remb_bit_rate); } } - auto header = rtp->getHeader(); - auto ext_map = RtpExt::getExtValue(header, *(info.media)); - for (auto &pr : ext_map) { - if (rtp->type == TrackVideo) { - InfoL << pr.second.dumpString(); - } else { - DebugL << pr.second.dumpString(); - } - //推流时修改ext id为统一的id，播放时再修改为对方设置的ext id - pr.second.setExtId((uint8_t) pr.first); - } - if (_push_src) { _push_src->onWrite(std::move(rtp), false); } } +void setExtType(RtpExt &ext, uint8_t tp) {} +void setExtType(RtpExt &ext, RtpExtType tp) { + ext.setType(tp); +} + +template +static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { + auto header = rtp->getHeader(); + auto ext_map = RtpExt::getExtValue(header); + for (auto &pr : ext_map) { + auto it = map.find((Type::key_type) pr.first); + if (it == map.end()) { + WarnL << "未处理的rtp ext, 类型不识别:" << (int) pr.first; + continue; + } + setExtType(pr.second, it->first); + setExtType(pr.second, it->second); + pr.second.setExtId((uint8_t) it->second); + } +} + void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { + changeRtpExtId(rtp, _rtp_ext_id_to_type); //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } @@ -657,6 +683,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //忽略，对方不支持该编码类型 return; } + changeRtpExtId(rtp, _rtp_ext_type_to_id); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); //统计rtp发送情况，好做sr汇报 diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 9cc64664..84394534 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -211,8 +211,6 @@ private: Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; - //rtc rtp推流的视频ssrc - uint32_t _recv_video_ssrc; //记录协商的rtp的pt类型 uint8_t _send_rtp_pt[2] = {0xFF, 0xFF}; //复合udp端口，接收一切rtp与rtcp @@ -225,8 +223,12 @@ private: RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 unordered_map _rtp_info_pt; - //根据推流端rtp的ssrc获取相关信息 + //根据推流端rtcp的ssrc获取相关信息 unordered_map _rtp_info_ssrc; + //发送rtp时需要修改rtp ext id + unordered_map _rtp_ext_type_to_id; + //接收rtp时需要修改rtp ext id + unordered_map _rtp_ext_id_to_type; }; From f625b3fbf391bb2000650f490d5644849e073cca Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 11:37:54 +0800 Subject: [PATCH 155/218] =?UTF-8?q?=E6=97=A0=E6=B3=95=E5=A4=84=E7=90=86?= =?UTF-8?q?=E7=9A=84rtp=20ext=E6=B8=85=E7=A9=BA?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 11 +++++++++++ webrtc/RtpExt.h | 1 + webrtc/WebRtcTransport.cpp | 1 + 3 files changed, 13 insertions(+) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 8189b9e9..a0f47bf1 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -525,6 +525,17 @@ void RtpExt::setExtId(uint8_t ext_id) { } } +void RtpExt::clearExt(){ + assert(_ptr); + if (_one_byte_ext) { + auto ptr = reinterpret_cast(_ptr); + memset(ptr, (int) RtpExtType::padding, RtpExtOneByte::kMinSize + ptr->getSize()); + } else { + auto ptr = reinterpret_cast(_ptr); + memset(ptr, (int) RtpExtType::padding, RtpExtTwoByte::kMinSize + ptr->getSize()); + } +} + void RtpExt::setType(RtpExtType type) { _type = type; } diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index d3975410..3d916def 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -90,6 +90,7 @@ public: //危险函数，必须保证关联的RtpHeader对象有效 void setExtId(uint8_t ext_id); + void clearExt(); private: RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size); diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 74f544dd..325debf4 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -663,6 +663,7 @@ static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { auto it = map.find((Type::key_type) pr.first); if (it == map.end()) { WarnL << "未处理的rtp ext, 类型不识别:" << (int) pr.first; + pr.second.clearExt(); continue; } setExtType(pr.second, it->first); From 0d8a3ef790c18d71ef7414d198ef43d0601aaf46 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 11:40:18 +0800 Subject: [PATCH 156/218] =?UTF-8?q?=E5=AE=8C=E5=96=84=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 325debf4..8e74c0de 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -650,8 +650,8 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr } } -void setExtType(RtpExt &ext, uint8_t tp) {} -void setExtType(RtpExt &ext, RtpExtType tp) { +static void setExtType(RtpExt &ext, uint8_t tp) {} +static void setExtType(RtpExt &ext, RtpExtType tp) { ext.setType(tp); } From c10296d432ae090f95cb79421f9a92b2f85e5b6a Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 14:50:19 +0800 Subject: [PATCH 157/218] =?UTF-8?q?=E6=9C=AA=E9=80=89=E4=B8=AD=E7=9A=84pla?= =?UTF-8?q?n=E4=B9=9F=E8=BF=94=E5=9B=9E=E5=88=B0sdp=E4=B8=AD?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 0dbb1fdc..a178ce35 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1464,7 +1464,7 @@ RETRY: WarnL << "answer sdp配置为ice_lite模式，与offer sdp中的ice_lite模式冲突"; continue; } - const RtcCodecPlan *offer_plan_ptr = nullptr; + const RtcCodecPlan *selected_plan = nullptr; for (auto &plan : offer_media.plan) { //先检查编码格式是否为偏好 if (check_codec && getCodecId(plan.codec) != codec) { @@ -1475,10 +1475,10 @@ RETRY: continue; } //找到中意的codec - offer_plan_ptr = &plan; + selected_plan = &plan; break; } - if (!offer_plan_ptr) { + if (!selected_plan) { //offer中该媒体的所有的codec都不支持 continue; } @@ -1542,27 +1542,32 @@ RETRY: } //添加媒体plan - answer_media.plan.emplace_back(*offer_plan_ptr); + answer_media.plan.emplace_back(*selected_plan); onSelectPlan(answer_media.plan.back(), codec); + set pt_selected = {selected_plan->pt}; + //添加rtx,red,ulpfec plan if (configure.support_red || configure.support_rtx || configure.support_ulpfec) { for (auto &plan : offer_media.plan) { if (!strcasecmp(plan.codec.data(), "rtx")) { - if (configure.support_rtx && atoi(plan.getFmtp("apt").data()) == offer_plan_ptr->pt) { + if (configure.support_rtx && atoi(plan.getFmtp("apt").data()) == selected_plan->pt) { answer_media.plan.emplace_back(plan); + pt_selected.emplace(plan.pt); } continue; } if (!strcasecmp(plan.codec.data(), "red")) { if (configure.support_red) { answer_media.plan.emplace_back(plan); + pt_selected.emplace(plan.pt); } continue; } if (!strcasecmp(plan.codec.data(), "ulpfec")) { if (configure.support_ulpfec) { answer_media.plan.emplace_back(plan); + pt_selected.emplace(plan.pt); } continue; } @@ -1579,12 +1584,19 @@ RETRY: auto &rtcp_fb_ref = answer_media.plan[0].rtcp_fb; rtcp_fb_ref.clear(); //对方和我方都支持的rtcpfb，那么我们才支持 - for (auto &fp : offer_plan_ptr->rtcp_fb) { + for (auto &fp : selected_plan->rtcp_fb) { if (configure.rtcp_fb.find(fp) != configure.rtcp_fb.end()) { //对方该rtcp被我们支持 rtcp_fb_ref.emplace(fp); } } + + //其他plan放在后面 + for (auto &plan : offer_media.plan) { + if (pt_selected.find(plan.pt) == pt_selected.end()) { + answer_media.plan.emplace_back(plan); + } + } ret->media.emplace_back(answer_media); return; } From 1403923138e6156f5f277b4bfedbf82f54e18c8b Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 15:01:14 +0800 Subject: [PATCH 158/218] =?UTF-8?q?=E8=AE=BE=E7=BD=AErtx=20ssrc?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 8e74c0de..ac48d0b1 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -17,7 +17,9 @@ #define RTX_SSRC_OFFSET 2 #define RTP_CNAME "zlmediakit-rtp" -#define RTX_CNAME "zlmediakit-rtx" +#define RTP_LABEL "zlmediakit-label" +#define RTP_MSLABEL "zlmediakit-mslabel" +#define RTP_MSID RTP_MSLABEL " " RTP_LABEL //RTC配置项目 namespace RTC { @@ -482,10 +484,14 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ //添加answer sdp的ssrc信息 m.rtp_ssrc.ssrc = _play_src->getSsrc(m.type); m.rtp_ssrc.cname = RTP_CNAME; + m.rtp_ssrc.label = RTP_LABEL; + m.rtp_ssrc.mslabel = RTP_MSLABEL; + m.rtp_ssrc.msid = RTP_MSID; + //todo 先屏蔽rtx，因为chrome报错 - if (false && m.getRelatedRtxPlan(m.plan[0].pt)) { - m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc; - m.rtx_ssrc.cname = RTX_CNAME; + if (m.getRelatedRtxPlan(m.plan[0].pt)) { + m.rtx_ssrc = m.rtp_ssrc; + m.rtx_ssrc.ssrc += RTX_SSRC_OFFSET; } } } From e84d16213c98404db88c265bde2be68fe89d84a8 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 15:01:30 +0800 Subject: [PATCH 159/218] =?UTF-8?q?sdp=E4=B8=AD=E6=B7=BB=E5=8A=A0msid?= =?UTF-8?q?=E5=B1=9E=E6=80=A7?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index a178ce35..f50f3b63 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1098,6 +1098,16 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ } } + if (!m.rtp_ssrc.empty()) { + auto msid = std::make_shared(); + if (!m.rtp_ssrc.msid.empty()) { + msid->parse(m.rtp_ssrc.msid); + } else { + msid->parse("mslabel label"); + } + sdp_media.items.emplace_back(wrapSdpAttr(std::move(msid))); + } + if (!m.rtp_ssrc.empty() && !m.rtx_ssrc.empty()) { auto group = std::make_shared(); group->type = "FID"; From 16fa359cc577d702de048c06a3156e3a63da5405 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 15:05:13 +0800 Subject: [PATCH 160/218] =?UTF-8?q?=E9=BB=98=E8=AE=A4=E4=B8=8D=E6=B7=BB?= =?UTF-8?q?=E5=8A=A0=E6=97=A0=E6=95=88=E7=9A=84codec=20plan?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index f50f3b63..affbec7b 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1601,12 +1601,14 @@ RETRY: } } - //其他plan放在后面 +#if 0 + //todo 此处为添加无效的plan，webrtc sdp通过调节plan pt顺序选择匹配的codec，意味着后面的codec其实放在sdp中是无意义的 for (auto &plan : offer_media.plan) { if (pt_selected.find(plan.pt) == pt_selected.end()) { answer_media.plan.emplace_back(plan); } } +#endif ret->media.emplace_back(answer_media); return; } From 26762b6a6f2a552b8fc734a7c50bb3478aa90b90 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 15:15:05 +0800 Subject: [PATCH 161/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0offer=20simulcast=20s?= =?UTF-8?q?dp?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/offer-simulcast.sdp | 129 ++++++++++++++++++++++++++++++++----- 1 file changed, 114 insertions(+), 15 deletions(-) diff --git a/webrtc/offer-simulcast.sdp b/webrtc/offer-simulcast.sdp index 91701061..d6f1ff84 100644 --- a/webrtc/offer-simulcast.sdp +++ b/webrtc/offer-simulcast.sdp @@ -1,5 +1,6 @@ +# chrome的sdp v=0 -o=- 99044479296054817 2 IN IP4 127.0.0.1 +o=- 403371946498103831 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 @@ -8,10 +9,10 @@ a=msid-semantic: WMS m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 -a=ice-ufrag:s95k -a=ice-pwd:JFAzK9rInMLPOosVj4Z7lC+Q +a=ice-ufrag:pW4Z +a=ice-pwd:S38S++HW3eTcPTyytsNI1XVp a=ice-options:trickle -a=fingerprint:sha-256 50:76:BB:FA:A8:D3:B7:19:81:C7:8D:19:60:C7:A9:7E:0E:5F:AA:3F:FE:68:60:98:B1:93:95:BC:A9:29:E4:D1 +a=fingerprint:sha-256 04:32:7B:56:7D:F7:D4:EC:65:7C:04:6C:F8:0B:03:F0:35:A9:1A:C3:43:3E:18:95:67:E6:0D:D1:EE:C9:16:8C a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level @@ -21,7 +22,7 @@ a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv -a=msid:- 83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 +a=msid:- a3c6a137-1291-45cd-b985-07a9bd365452 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc @@ -38,17 +39,17 @@ a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 -a=ssrc:1320289824 cname:YOYQAT5W3y5SSliB -a=ssrc:1320289824 msid:- 83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 -a=ssrc:1320289824 mslabel:- -a=ssrc:1320289824 label:83e7b60d-ce8f-4d7d-a7b9-d2209747bf55 -m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 +a=ssrc:3626257331 cname:JSFJMbaE9Pu5tevN +a=ssrc:3626257331 msid:- a3c6a137-1291-45cd-b985-07a9bd365452 +a=ssrc:3626257331 mslabel:- +a=ssrc:3626257331 label:a3c6a137-1291-45cd-b985-07a9bd365452 +m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 -a=ice-ufrag:s95k -a=ice-pwd:JFAzK9rInMLPOosVj4Z7lC+Q +a=ice-ufrag:pW4Z +a=ice-pwd:S38S++HW3eTcPTyytsNI1XVp a=ice-options:trickle -a=fingerprint:sha-256 50:76:BB:FA:A8:D3:B7:19:81:C7:8D:19:60:C7:A9:7E:0E:5F:AA:3F:FE:68:60:98:B1:93:95:BC:A9:29:E4:D1 +a=fingerprint:sha-256 04:32:7B:56:7D:F7:D4:EC:65:7C:04:6C:F8:0B:03:F0:35:A9:1A:C3:43:3E:18:95:67:E6:0D:D1:EE:C9:16:8C a=setup:actpass a=mid:1 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset @@ -63,7 +64,7 @@ a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv -a=msid:- cd8e4810-dadf-4f11-8799-eed2560fe196 +a=msid:- 261e5384-9cf6-479d-9d59-aaf924d1a2ea a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 @@ -128,6 +129,14 @@ a=rtcp-fb:108 nack pli a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 +a=rtpmap:35 AV1X/90000 +a=rtcp-fb:35 goog-remb +a=rtcp-fb:35 transport-cc +a=rtcp-fb:35 ccm fir +a=rtcp-fb:35 nack +a=rtcp-fb:35 nack pli +a=rtpmap:36 rtx/90000 +a=fmtp:36 apt=35 a=rtpmap:124 H264/90000 a=rtcp-fb:124 goog-remb a=rtcp-fb:124 transport-cc @@ -153,4 +162,94 @@ a=rtpmap:116 ulpfec/90000 a=rid:q send a=rid:h send a=rid:f send -a=simulcast:send q;h;f \ No newline at end of file +a=simulcast:send q;h;f + +#firefox的sdp +v=0 +o=mozilla...THIS_IS_SDPARTA-88.0.1 3954544078885279475 0 IN IP4 0.0.0.0 +s=- +t=0 0 +a=fingerprint:sha-256 9B:4F:D1:D2:A5:ED:08:BC:E8:D7:DD:D8:59:2C:E6:3D:19:F9:4C:67:9C:D9:9B:7B:C9:47:7A:3A:1F:05:C8:96 +a=group:BUNDLE 0 1 +a=ice-options:trickle +a=msid-semantic:WMS * +m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 +c=IN IP4 0.0.0.0 +a=sendrecv +a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level +a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level +a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid +a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1 +a=fmtp:101 0-15 +a=ice-pwd:92a9ced6d734f7ff2a45cde8b29572a9 +a=ice-ufrag:b986b945 +a=mid:0 +a=msid:- {ea61729a-c244-4c79-aeb7-b57765fefa26} +a=rtcp-mux +a=rtpmap:109 opus/48000/2 +a=rtpmap:9 G722/8000/1 +a=rtpmap:0 PCMU/8000 +a=rtpmap:8 PCMA/8000 +a=rtpmap:101 telephone-event/8000/1 +a=setup:actpass +a=ssrc:3000327501 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc} +m=video 9 UDP/TLS/RTP/SAVPF 120 124 121 125 126 127 97 98 +c=IN IP4 0.0.0.0 +a=sendrecv +a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid +a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time +a=extmap:5 urn:ietf:params:rtp-hdrext:toffset +a=extmap:6/recvonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay +a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 +a=extmap:8/sendonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id +a=extmap:9/sendonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id +a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1 +a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1 +a=fmtp:120 max-fs=12288;max-fr=60 +a=fmtp:124 apt=120 +a=fmtp:121 max-fs=12288;max-fr=60 +a=fmtp:125 apt=121 +a=fmtp:127 apt=126 +a=fmtp:98 apt=97 +a=ice-pwd:92a9ced6d734f7ff2a45cde8b29572a9 +a=ice-ufrag:b986b945 +a=mid:1 +a=msid:- {3bfe1b80-20eb-4b42-b8b7-fac45fb281bf} +a=rid:q send +a=rid:h send +a=rid:f send +a=rtcp-fb:120 nack +a=rtcp-fb:120 nack pli +a=rtcp-fb:120 ccm fir +a=rtcp-fb:120 goog-remb +a=rtcp-fb:120 transport-cc +a=rtcp-fb:121 nack +a=rtcp-fb:121 nack pli +a=rtcp-fb:121 ccm fir +a=rtcp-fb:121 goog-remb +a=rtcp-fb:121 transport-cc +a=rtcp-fb:126 nack +a=rtcp-fb:126 nack pli +a=rtcp-fb:126 ccm fir +a=rtcp-fb:126 goog-remb +a=rtcp-fb:126 transport-cc +a=rtcp-fb:97 nack +a=rtcp-fb:97 nack pli +a=rtcp-fb:97 ccm fir +a=rtcp-fb:97 goog-remb +a=rtcp-fb:97 transport-cc +a=rtcp-mux +a=rtcp-rsize +a=rtpmap:120 VP8/90000 +a=rtpmap:124 rtx/90000 +a=rtpmap:121 VP9/90000 +a=rtpmap:125 rtx/90000 +a=rtpmap:126 H264/90000 +a=rtpmap:127 rtx/90000 +a=rtpmap:97 H264/90000 +a=rtpmap:98 rtx/90000 +a=setup:actpass +a=simulcast:send q;h;f +a=ssrc:2581133096 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc} +a=ssrc:773854125 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc} +a=ssrc:4100728001 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc} From e81d91988d8c8d7320afc986b7bbdbf36b65ee85 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 18:39:35 +0800 Subject: [PATCH 162/218] =?UTF-8?q?=E5=AE=8C=E5=96=84sdp=E5=AF=B9ssrc?= =?UTF-8?q?=E7=9A=84=E5=A4=84=E7=90=86?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- .gitignore | 1 + webrtc/Sdp.cpp | 190 +++++++++++++++++++++---------------- webrtc/Sdp.h | 29 +++--- webrtc/WebRtcTransport.cpp | 19 ++-- 4 files changed, 130 insertions(+), 109 deletions(-) diff --git a/.gitignore b/.gitignore index ffb0e5b3..37c326cd 100644 --- a/.gitignore +++ b/.gitignore @@ -46,3 +46,4 @@ /3rdpart/media-server/.idea/ /ios/ /cmake-build-* +/3rdpart/ZLToolKit/cmake-build-mq/ diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index affbec7b..35cad68e 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -659,21 +659,13 @@ string SdpAttrSSRC::toString() const { void SdpAttrSSRCGroup::parse(const string &str) { auto vec = split(str, " "); - if (vec.size() == 3) { - if (vec[0] != "FID") { - SDP_THROW(); - } + if (vec.size() >= 3) { type = std::move(vec[0]); - u.fid.rtp_ssrc = atoll(vec[1].data()) & 0xFFFFFFFF; - u.fid.rtx_ssrc = atoll(vec[2].data()) & 0xFFFFFFFF; - } else if (vec.size() == 4) { - if (vec[0] != "SIM") { - SDP_THROW(); + CHECK(isFID() || isSIM()); + vec.erase(vec.begin()); + for (auto ssrc : vec) { + ssrcs.emplace_back((uint32_t)atoll(ssrc.data())); } - type = std::move(vec[0]); - u.sim.rtp_ssrc_low = atoll(vec[1].data()) & 0xFFFFFFFF; - u.sim.rtp_ssrc_mid = atoll(vec[2].data()) & 0xFFFFFFFF; - u.sim.rtp_ssrc_high = atoll(vec[3].data()) & 0xFFFFFFFF; } else { SDP_THROW(); } @@ -681,12 +673,12 @@ void SdpAttrSSRCGroup::parse(const string &str) { string SdpAttrSSRCGroup::toString() const { if (value.empty()) { - if (isFID()) { - value = type + " " + to_string(u.fid.rtp_ssrc) + " " + to_string(u.fid.rtx_ssrc); - } else if (isSIM()) { - value = type + " " + to_string(u.sim.rtp_ssrc_low) + " " + to_string(u.sim.rtp_ssrc_mid) + " " + to_string(u.sim.rtp_ssrc_high); - } else { - SDP_THROW2(); + value = type; + //最少要求2个ssrc + CHECK(ssrcs.size() >= 2); + for (auto &ssrc : ssrcs) { + value += ' '; + value += to_string(ssrc); } } return SdpItem::toString(); @@ -758,6 +750,34 @@ string SdpAttrCandidate::toString() const { return SdpItem::toString(); } +void SdpAttrSimulcast::parse(const string &str) { + //https://www.meetecho.com/blog/simulcast-janus-ssrc/ + //a=simulcast:send/recv q;h;f + //a=simulcast:send/recv [rid=]q;h;f + //a=simulcast: recv h;m;l + // + auto vec = split(str, " "); + if (vec.size() != 2) { + SDP_THROW(); + } + direction = vec[0]; + vec = split(vec[1], ";"); + rid0 = vec[0]; + if (vec.size() > 1) { + rid1 = vec[1]; + if (vec.size() > 2) { + rid2 = vec[2]; + } + } +} + +string SdpAttrSimulcast::toString() const { + if (value.empty()) { + value = direction + " " + rid0 + ";" + rid1 + ";" + rid2; + } + return SdpItem::toString(); +} + void RtcSession::loadFrom(const string &str, bool check) { RtcSessionSdp sdp; sdp.parse(str); @@ -838,44 +858,43 @@ void RtcSession::loadFrom(const string &str, bool check) { } } - uint32_t ssrc_rtp = 0, ssrc_rtx = 0, ssrc_rtp_low = 0, ssrc_rtp_mid = 0, ssrc_rtp_high = 0; auto ssrc_groups = media.getAllItem('a', "ssrc-group"); + SdpAttrSSRCGroup *ssrc_group_sim = nullptr; + SdpAttrSSRCGroup *ssrc_group_fid = nullptr; for (auto &group : ssrc_groups) { if (group.isFID()) { - ssrc_rtp = group.u.fid.rtp_ssrc; - ssrc_rtx = group.u.fid.rtx_ssrc; + ssrc_group_fid = &group; } else if (group.isSIM()) { - ssrc_rtp_low = group.u.sim.rtp_ssrc_low; - ssrc_rtp_mid = group.u.sim.rtp_ssrc_mid; - ssrc_rtp_high = group.u.sim.rtp_ssrc_high; + ssrc_group_sim = &group; } } - if (!ssrc_rtp) { - //没有指定ssrc-group字段，那么只有一个ssrc - if (rtc_ssrc_map.size() > 1) { - throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却有多个"); + if (ssrc_group_fid) { + //指定了ssrc-group:FID字段 + for (auto ssrc : ssrc_group_fid->ssrcs) { + auto it = rtc_ssrc_map.find(ssrc); + if (it == rtc_ssrc_map.end()) { + throw std::invalid_argument("a=ssrc-group:FID字段指定的ssrc未找到"); + } + rtc_media.rtp_rtx_ssrc.emplace_back(it->second); } + CHECK(rtc_media.rtp_rtx_ssrc.size() == 2); + } else { + //没有指定ssrc-group:FID字段，那么只有1个或0个ssrc if (rtc_ssrc_map.size() == 1) { - ssrc_rtp = rtc_ssrc_map.begin()->second.ssrc; + rtc_media.rtp_rtx_ssrc.emplace_back(rtc_ssrc_map.begin()->second); + } else if (rtc_ssrc_map.size() > 1) { + throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却大于1个"); } } - for (auto &pr : rtc_ssrc_map) { - auto &rtc_ssrc = pr.second; - if (rtc_ssrc.ssrc == ssrc_rtp) { - rtc_media.rtp_ssrc = rtc_ssrc; - } - if (rtc_ssrc.ssrc == ssrc_rtx) { - rtc_media.rtx_ssrc = rtc_ssrc; - } - if (rtc_ssrc.ssrc == ssrc_rtp_low) { - rtc_media.rtp_ssrc_low = rtc_ssrc; - } - if (rtc_ssrc.ssrc == ssrc_rtp_mid) { - rtc_media.rtp_ssrc_mid = rtc_ssrc; - } - if (rtc_ssrc.ssrc == ssrc_rtp_high) { - rtc_media.rtp_ssrc_high = rtc_ssrc; + + if (ssrc_group_sim) { + for (auto ssrc : ssrc_group_sim->ssrcs) { + auto it = rtc_ssrc_map.find(ssrc); + if (it == rtc_ssrc_map.end()) { + throw std::invalid_argument("a=ssrc-group:SIM字段指定的ssrc未找到"); + } + rtc_media.rtp_ssrc_sim.emplace_back(it->second); } } @@ -1081,12 +1100,14 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ rtp_map->codec = p.codec; rtp_map->sample_rate = p.sample_rate; rtp_map->channel = p.channel; + //添加a=rtpmap sdp_media.items.emplace_back(wrapSdpAttr(std::move(rtp_map))); for (auto &fb : p.rtcp_fb) { auto rtcp_fb = std::make_shared(); rtcp_fb->pt = p.pt; rtcp_fb->rtcp_type = fb; + //添加a=rtcp-fb sdp_media.items.emplace_back(wrapSdpAttr(std::move(rtcp_fb))); } @@ -1094,29 +1115,26 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ auto fmtp = std::make_shared(); fmtp->pt = p.pt; fmtp->fmtp = p.fmtp; + //添加a=fmtp sdp_media.items.emplace_back(wrapSdpAttr(std::move(fmtp))); } } - if (!m.rtp_ssrc.empty()) { - auto msid = std::make_shared(); - if (!m.rtp_ssrc.msid.empty()) { - msid->parse(m.rtp_ssrc.msid); - } else { - msid->parse("mslabel label"); + { + //添加a=msid字段 + if (!m.rtp_rtx_ssrc.empty()) { + auto msid = std::make_shared(); + if (!m.rtp_rtx_ssrc[0].msid.empty()) { + msid->parse(m.rtp_rtx_ssrc[0].msid); + } else { + msid->parse("mslabel label"); + } + sdp_media.items.emplace_back(wrapSdpAttr(std::move(msid))); } - sdp_media.items.emplace_back(wrapSdpAttr(std::move(msid))); - } - - if (!m.rtp_ssrc.empty() && !m.rtx_ssrc.empty()) { - auto group = std::make_shared(); - group->type = "FID"; - group->u.fid.rtp_ssrc = m.rtp_ssrc.ssrc; - group->u.fid.rtx_ssrc = m.rtx_ssrc.ssrc; - sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); } static auto addSSRCItem = [](const RtcSSRC &rtp_ssrc, vector &items) { + CHECK(!rtp_ssrc.empty()); SdpAttrSSRC ssrc; ssrc.ssrc = rtp_ssrc.ssrc; @@ -1142,27 +1160,34 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ items.emplace_back(wrapSdpAttr(std::make_shared(ssrc))); } }; - if (!m.rtp_ssrc.empty()) { - addSSRCItem(m.rtp_ssrc, sdp_media.items); - } - if (!m.rtx_ssrc.empty()) { - addSSRCItem(m.rtx_ssrc, sdp_media.items); + + { + auto group = std::make_shared(); + for (auto &ssrc : m.rtp_rtx_ssrc) { + //添加a=ssrc字段 + addSSRCItem(ssrc, sdp_media.items); + group->ssrcs.emplace_back(ssrc.ssrc); + } + if (group->ssrcs.size() >= 2) { + group->type = "FID"; + //生成a=ssrc-group:FID字段 + sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); + } } - bool enable_sim = false; - if (!m.rtp_ssrc_low.empty() && !m.rtp_ssrc_mid.empty() && !m.rtp_ssrc_high.empty()) { - auto group = std::make_shared(); - group->type = "SIM"; - group->u.sim.rtp_ssrc_low = m.rtp_ssrc_low.ssrc; - group->u.sim.rtp_ssrc_mid = m.rtp_ssrc_mid.ssrc; - group->u.sim.rtp_ssrc_high = m.rtp_ssrc_high.ssrc; - sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); - enable_sim = true; - } - if (enable_sim) { - addSSRCItem(m.rtp_ssrc_low, sdp_media.items); - addSSRCItem(m.rtp_ssrc_mid, sdp_media.items); - addSSRCItem(m.rtp_ssrc_high, sdp_media.items); + { + if (m.rtp_ssrc_sim.size() >= 2) { + //simulcast 要求 2~3路 + auto group = std::make_shared(); + for (auto &ssrc : m.rtp_ssrc_sim) { + //添加simulcast的ssrc + addSSRCItem(ssrc, sdp_media.items); + group->ssrcs.emplace_back(ssrc.ssrc); + } + //添加a=ssrc-group:SIM字段 + group->type = "SIM"; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); + } } } else { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.sctpmap))); @@ -1227,12 +1252,11 @@ void RtcMedia::checkValid() const{ CHECK(direction != RtpDirection::invalid || type == TrackApplication); CHECK(!plan.empty() || type == TrackApplication ); bool send_rtp = (direction == RtpDirection::sendonly || direction == RtpDirection::sendrecv); - CHECK(!rtp_ssrc.empty() || !send_rtp); + CHECK(!rtp_rtx_ssrc.empty() || !send_rtp); auto rtx_plan = getPlan("rtx"); if (rtx_plan) { //开启rtx后必须指定rtx_ssrc - //todo 此处不确定 -// CHECK(!rtx_ssrc.empty() || !send_rtp); + CHECK(rtp_rtx_ssrc.size() >= 2 || !send_rtp); } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index bfff21eb..0f0a5c1c 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -414,17 +414,7 @@ public: //a=ssrc-group:SIM 360918977 360918978 360918980 // 在 Chrome 独有的 SDP munging 风格的 simulcast 中使用，将三组编码质量由低到高的 MediaStreamTrack 关联在一起。 string type{"FID"}; - union { - struct { - uint32_t rtp_ssrc; - uint32_t rtx_ssrc; - } fid; - struct { - uint32_t rtp_ssrc_low; - uint32_t rtp_ssrc_mid; - uint32_t rtp_ssrc_high; - } sim; - } u; + vector ssrcs; bool isFID() const { return type == "FID"; } bool isSIM() const { return type == "SIM"; } @@ -480,8 +470,16 @@ public: class SdpAttrSimulcast : public SdpItem{ public: - //todo + //https://www.meetecho.com/blog/simulcast-janus-ssrc/ + //https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-14 const char* getKey() const override { return "simulcast";} + void parse(const string &str) override; + string toString() const override; + + string direction; + string rid0; + string rid1; + string rid2; }; class SdpAttrRid : public SdpItem{ @@ -610,13 +608,10 @@ public: vector plan; //////// rtp //////// - RtcSSRC rtp_ssrc; - RtcSSRC rtx_ssrc; + vector rtp_rtx_ssrc; //////// simulcast //////// - RtcSSRC rtp_ssrc_low; - RtcSSRC rtp_ssrc_mid; - RtcSSRC rtp_ssrc_high; + vector rtp_ssrc_sim; //////// rtcp //////// bool rtcp_mux{false}; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index ac48d0b1..f2e232d8 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -401,7 +401,7 @@ void WebRtcTransportImp::onStartWebRTC() { } //获取offer端rtp的ssrc和pt相关信息 auto &ref = _rtp_info_pt[plan.pt]; - _rtp_info_ssrc[m.rtp_ssrc.ssrc] = &ref; + _rtp_info_ssrc[m.rtp_rtx_ssrc[0].ssrc] = &ref; ref.plan = &plan; ref.media = &m; ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; @@ -482,16 +482,17 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ continue; } //添加answer sdp的ssrc信息 - m.rtp_ssrc.ssrc = _play_src->getSsrc(m.type); - m.rtp_ssrc.cname = RTP_CNAME; - m.rtp_ssrc.label = RTP_LABEL; - m.rtp_ssrc.mslabel = RTP_MSLABEL; - m.rtp_ssrc.msid = RTP_MSID; + m.rtp_rtx_ssrc.emplace_back(); + m.rtp_rtx_ssrc[0].ssrc = _play_src->getSsrc(m.type); + m.rtp_rtx_ssrc[0].cname = RTP_CNAME; + m.rtp_rtx_ssrc[0].label = RTP_LABEL; + m.rtp_rtx_ssrc[0].mslabel = RTP_MSLABEL; + m.rtp_rtx_ssrc[0].msid = RTP_MSID; - //todo 先屏蔽rtx，因为chrome报错 if (m.getRelatedRtxPlan(m.plan[0].pt)) { - m.rtx_ssrc = m.rtp_ssrc; - m.rtx_ssrc.ssrc += RTX_SSRC_OFFSET; + m.rtp_rtx_ssrc.emplace_back(); + m.rtp_rtx_ssrc[1] = m.rtp_rtx_ssrc[0]; + m.rtp_rtx_ssrc[1].ssrc += RTX_SSRC_OFFSET; } } } From 0377796c031275a8186108c015fa8e418c6e05fb Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 19:33:51 +0800 Subject: [PATCH 163/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dclang=E4=B8=8B?= =?UTF-8?q?=E7=BC=96=E8=AF=91=E8=AD=A6=E5=91=8A=E5=92=8C=E9=94=99=E8=AF=AF?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 7 +++---- src/Rtcp/Rtcp.h | 2 +- src/Rtcp/RtcpFCI.cpp | 2 +- webrtc/RtpExt.cpp | 6 +++--- webrtc/WebRtcTransport.cpp | 2 +- webrtc/WebRtcTransport.h | 2 +- 6 files changed, 10 insertions(+), 11 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 16e9c9fa..4b7dbd25 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -236,7 +236,6 @@ public: } private: - std::size_t _size; std::shared_ptr _rtcp; }; @@ -604,7 +603,7 @@ std::shared_ptr RtcpBye::create(const std::vector &ssrcs, con setupHeader(ptr, RtcpType::RTCP_BYE, ssrcs.size(), bytes); setupPadding(ptr, bytes - real_size); - auto *ssrc_ptr = &(((RtcpBye *) ptr)->ssrc); + auto ssrc_ptr = ((RtcpBye *) ptr)->ssrc; for (auto ssrc : ssrcs) { *ssrc_ptr = htonl(ssrc); ++ssrc_ptr; @@ -623,7 +622,7 @@ std::shared_ptr RtcpBye::create(const std::vector &ssrcs, con vector RtcpBye::getSSRC() { vector ret; - uint32_t *ssrc_ptr = &ssrc; + auto ssrc_ptr = ssrc; for (size_t i = 0; i < report_count; ++i) { ret.emplace_back(ssrc_ptr); ssrc_ptr += 1; @@ -652,7 +651,7 @@ string RtcpBye::dumpString() const { void RtcpBye::net2Host(size_t size) { static const size_t kMinSize = sizeof(RtcpHeader); CHECK_MIN_SIZE(size, kMinSize); - uint32_t *ssrc_ptr = &ssrc; + auto ssrc_ptr = ssrc; size_t offset = kMinSize; size_t i = 0; for (; i < report_count && offset + sizeof(ssrc) <= size; ++i) { diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index 836699d2..c4055f71 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -622,7 +622,7 @@ class RtcpBye : public RtcpHeader { public: friend class RtcpHeader; /* 变长，根据count决定有多少个ssrc */ - uint32_t ssrc; + uint32_t ssrc[1]; /** 中间可能有若干个 ssrc **/ diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index d0f4fb62..6acb4d3a 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -149,7 +149,7 @@ string FCI_REMB::dumpString() const { for (auto &ssrc : ((FCI_REMB *) this)->getSSRC()) { printer << ssrc << " "; } - return printer; + return std::move(printer); } /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index a0f47bf1..5ea0192d 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -130,7 +130,7 @@ static bool isOneByteExt(){ } template -static void appendExt(map &ret, uint8_t *ptr, const uint8_t *end) { +void appendExt(map &ret, uint8_t *ptr, const uint8_t *end) { while (ptr < end) { auto ext = reinterpret_cast(ptr); if (ext->getId() == (uint8_t) RtpExtType::padding) { @@ -191,7 +191,7 @@ map RtpExt::getExtValue(const RtpHeader *header) XX(encrypt, "urn:ietf:params:rtp-hdrext:encrypt") #define XX(type, url) {RtpExtType::type , url}, -static unordered_map s_type_to_url = {RTP_EXT_MAP(XX)}; +static map s_type_to_url = {RTP_EXT_MAP(XX)}; #undef XX @@ -353,7 +353,7 @@ uint32_t RtpExt::getAbsSendTime() const { uint16_t RtpExt::getTransportCCSeq() const { CHECK(_type == RtpExtType::transport_cc && size() >= 2); uint16_t ret; - ret |= (*this)[0] << 8; + ret = (*this)[0] << 8; ret |= (*this)[1]; return ret; } diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index f2e232d8..38fecd1f 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -667,7 +667,7 @@ static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { auto header = rtp->getHeader(); auto ext_map = RtpExt::getExtValue(header); for (auto &pr : ext_map) { - auto it = map.find((Type::key_type) pr.first); + auto it = map.find((typename Type::key_type) (pr.first)); if (it == map.end()) { WarnL << "未处理的rtp ext, 类型不识别:" << (int) pr.first; pr.second.clearExt(); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 84394534..84fdbde5 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -226,7 +226,7 @@ private: //根据推流端rtcp的ssrc获取相关信息 unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id - unordered_map _rtp_ext_type_to_id; + map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id unordered_map _rtp_ext_id_to_type; }; From 826ec33c2cb41a813e092c18c4ba735546075d32 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 19:45:53 +0800 Subject: [PATCH 164/218] =?UTF-8?q?rtx=20ssrc=E4=B8=8D=E5=BC=BA=E5=88=B6?= =?UTF-8?q?=E6=8C=87=E5=AE=9A?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 35cad68e..8f633566 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1253,11 +1253,15 @@ void RtcMedia::checkValid() const{ CHECK(!plan.empty() || type == TrackApplication ); bool send_rtp = (direction == RtpDirection::sendonly || direction == RtpDirection::sendrecv); CHECK(!rtp_rtx_ssrc.empty() || !send_rtp); + +#if 0 + //todo 发现Firefox(88.0)在mac平台下，开启rtx后没有指定ssrc auto rtx_plan = getPlan("rtx"); if (rtx_plan) { //开启rtx后必须指定rtx_ssrc CHECK(rtp_rtx_ssrc.size() >= 2 || !send_rtp); } +#endif } void RtcSession::checkValid() const{ From 47dc661bb21d80b96f7fb6a8c04862dd315d113d Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 20:27:46 +0800 Subject: [PATCH 165/218] =?UTF-8?q?=E5=88=9D=E6=AD=A5=E5=85=BC=E5=AE=B9sim?= =?UTF-8?q?ulcast?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 83 ++++++++++++++++++++++++++++++++++++++++---------- webrtc/Sdp.h | 12 +++++--- 2 files changed, 74 insertions(+), 21 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 8f633566..02ecef91 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -761,19 +761,35 @@ void SdpAttrSimulcast::parse(const string &str) { SDP_THROW(); } direction = vec[0]; - vec = split(vec[1], ";"); - rid0 = vec[0]; - if (vec.size() > 1) { - rid1 = vec[1]; - if (vec.size() > 2) { - rid2 = vec[2]; - } - } + rids = split(vec[1], ";"); } string SdpAttrSimulcast::toString() const { if (value.empty()) { - value = direction + " " + rid0 + ";" + rid1 + ";" + rid2; + value = direction + " "; + bool first = true; + for (auto &rid : rids) { + if (first) { + first = false; + } else { + value += ';'; + } + value += rid; + } + } + return SdpItem::toString(); +} + +void SdpAttrRid::parse(const string &str) { + auto vec = split(str, " "); + CHECK(vec.size() >= 2); + rid = vec[0]; + direction = vec[1]; +} + +string SdpAttrRid::toString() const { + if (value.empty()) { + value = rid + " " + direction; } return SdpItem::toString(); } @@ -837,7 +853,8 @@ void RtcSession::loadFrom(const string &str, bool check) { rtc_media.rtcp_rsize = media.getItem('a', "rtcp-rsize").operator bool(); map rtc_ssrc_map; - for (auto &ssrc : media.getAllItem('a', "ssrc")) { + auto ssrc_attr = media.getAllItem('a', "ssrc"); + for (auto &ssrc : ssrc_attr) { auto &rtc_ssrc = rtc_ssrc_map[ssrc.ssrc]; rtc_ssrc.ssrc = ssrc.ssrc; if (!strcasecmp(ssrc.attribute.data(), "cname")) { @@ -880,15 +897,24 @@ void RtcSession::loadFrom(const string &str, bool check) { } CHECK(rtc_media.rtp_rtx_ssrc.size() == 2); } else { - //没有指定ssrc-group:FID字段，那么只有1个或0个ssrc - if (rtc_ssrc_map.size() == 1) { - rtc_media.rtp_rtx_ssrc.emplace_back(rtc_ssrc_map.begin()->second); - } else if (rtc_ssrc_map.size() > 1) { - throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却大于1个"); + auto simulcast = media.getItemClass('a', "simulcast"); + if (simulcast.empty()) { + //没有指定ssrc-group:FID字段,也不是simulcast，那么只有1个或0个ssrc + if (rtc_ssrc_map.size() == 1) { + rtc_media.rtp_rtx_ssrc.emplace_back(rtc_ssrc_map.begin()->second); + } else if (rtc_ssrc_map.size() > 1) { + throw std::invalid_argument("sdp中不存在a=ssrc-group:FID字段,但是ssrc却大于1个"); + } + } else { + //开启simulcast + rtc_media.rtp_rids = simulcast.rids; + //simulcast最少要求2种方案 + CHECK(rtc_media.rtp_rids.size() >= 2); } } if (ssrc_group_sim) { + //指定了a=ssrc-group:SIM for (auto ssrc : ssrc_group_sim->ssrcs) { auto it = rtc_ssrc_map.find(ssrc); if (it == rtc_ssrc_map.end()) { @@ -896,6 +922,11 @@ void RtcSession::loadFrom(const string &str, bool check) { } rtc_media.rtp_ssrc_sim.emplace_back(it->second); } + } else if (!rtc_media.rtp_rids.empty()) { + //未指定a=ssrc-group:SIM，但是指定了a=simulcast,且可能指定了ssrc + for (auto &attr : ssrc_attr) { + rtc_media.rtp_ssrc_sim.emplace_back(rtc_ssrc_map[attr.ssrc]); + } } auto rtpmap_arr = media.getAllItem('a', "rtpmap"); @@ -1188,7 +1219,22 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ group->type = "SIM"; sdp_media.items.emplace_back(wrapSdpAttr(std::move(group))); } + + if (m.rtp_rids.size() >= 2) { + auto simulcast = std::make_shared(); + simulcast->direction = "recv"; + simulcast->rids = m.rtp_rids; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(simulcast))); + + for (auto &rid : m.rtp_rids) { + auto attr_rid = std::make_shared(); + attr_rid->rid = rid; + attr_rid->direction = "recv"; + sdp_media.items.emplace_back(wrapSdpAttr(std::move(attr_rid))); + } + } } + } else { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.sctpmap))); sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("sctp-port", to_string(m.sctp_port)))); @@ -1252,7 +1298,10 @@ void RtcMedia::checkValid() const{ CHECK(direction != RtpDirection::invalid || type == TrackApplication); CHECK(!plan.empty() || type == TrackApplication ); bool send_rtp = (direction == RtpDirection::sendonly || direction == RtpDirection::sendrecv); - CHECK(!rtp_rtx_ssrc.empty() || !send_rtp); + if (rtp_rids.empty() && rtp_ssrc_sim.empty()) { + //非simulcast时，检查有没有指定rtp ssrc + CHECK(!rtp_rtx_ssrc.empty() || !send_rtp); + } #if 0 //todo 发现Firefox(88.0)在mac平台下，开启rtx后没有指定ssrc @@ -1539,6 +1588,8 @@ RETRY: answer_media.fingerprint = configure.fingerprint; answer_media.ice_lite = configure.ice_lite; answer_media.candidate = configure.candidate; + answer_media.rtp_rids = offer_media.rtp_rids; + answer_media.rtp_ssrc_sim = offer_media.rtp_ssrc_sim; switch (offer_media.role) { case DtlsRole::actpass : case DtlsRole::active : { diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 0f0a5c1c..3a111649 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -475,17 +475,18 @@ public: const char* getKey() const override { return "simulcast";} void parse(const string &str) override; string toString() const override; - + bool empty() const { return rids.empty(); } string direction; - string rid0; - string rid1; - string rid2; + vector rids; }; class SdpAttrRid : public SdpItem{ public: - //todo + void parse(const string &str) override; + string toString() const override; const char* getKey() const override { return "rid";} + string direction; + string rid; }; class RtcSdpBase { @@ -612,6 +613,7 @@ public: //////// simulcast //////// vector rtp_ssrc_sim; + vector rtp_rids; //////// rtcp //////// bool rtcp_mux{false}; From 60a6d4af0bfc5c320b1bbc3f200eb13925c154ec Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 21:16:51 +0800 Subject: [PATCH 166/218] =?UTF-8?q?=E6=8B=B7=E8=B4=9D=E5=90=8E=E5=86=8D?= =?UTF-8?q?=E4=BF=AE=E6=94=B9rtp=EF=BC=8C=E9=98=B2=E6=AD=A2=E4=BF=AE?= =?UTF-8?q?=E6=94=B9=E5=85=B1=E4=BA=AB=E6=95=B0=E6=8D=AE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/SrtpSession.cpp | 37 ++++---------------------------- webrtc/SrtpSession.hpp | 7 ++----- webrtc/WebRtcTransport.cpp | 43 +++++++++++++++++++++----------------- webrtc/WebRtcTransport.h | 8 +++++-- 4 files changed, 36 insertions(+), 59 deletions(-) diff --git a/webrtc/SrtpSession.cpp b/webrtc/SrtpSession.cpp index 4591e466..7aa109c1 100644 --- a/webrtc/SrtpSession.cpp +++ b/webrtc/SrtpSession.cpp @@ -231,23 +231,11 @@ namespace RTC } } - bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt) + bool SrtpSession::EncryptRtp(uint8_t* data, size_t* len) { MS_TRACE(); - - // Ensure that the resulting SRTP packet fits into the encrypt buffer. - if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) - { - MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len); - - return false; - } - - std::memcpy(EncryptBuffer, *data, *len); - EncryptBuffer[1] = (pt & 0x7F) | (EncryptBuffer[1] & 0x80); - srtp_err_status_t err = - srtp_protect(this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); + srtp_protect(this->session, static_cast(data), reinterpret_cast(len)); if (DepLibSRTP::IsError(err)) { @@ -256,9 +244,6 @@ namespace RTC return false; } - // Update the given data pointer. - *data = (const uint8_t*)EncryptBuffer; - return true; } @@ -279,22 +264,11 @@ namespace RTC return true; } - bool SrtpSession::EncryptRtcp(const uint8_t** data, size_t* len) + bool SrtpSession::EncryptRtcp(uint8_t* data, size_t* len) { MS_TRACE(); - - // Ensure that the resulting SRTCP packet fits into the encrypt buffer. - if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize) - { - MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len); - - return false; - } - - std::memcpy(EncryptBuffer, *data, *len); - srtp_err_status_t err = srtp_protect_rtcp( - this->session, static_cast(EncryptBuffer), reinterpret_cast(len)); + this->session, static_cast(data), reinterpret_cast(len)); if (DepLibSRTP::IsError(err)) { @@ -303,9 +277,6 @@ namespace RTC return false; } - // Update the given data pointer. - *data = (const uint8_t*)EncryptBuffer; - return true; } diff --git a/webrtc/SrtpSession.hpp b/webrtc/SrtpSession.hpp index d9d73fbe..a50684e5 100644 --- a/webrtc/SrtpSession.hpp +++ b/webrtc/SrtpSession.hpp @@ -64,9 +64,9 @@ namespace RTC ~SrtpSession(); public: - bool EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt); + bool EncryptRtp(uint8_t* data, size_t* len); bool DecryptSrtp(uint8_t* data, size_t* len); - bool EncryptRtcp(const uint8_t** data, size_t* len); + bool EncryptRtcp(uint8_t* data, size_t* len); bool DecryptSrtcp(uint8_t* data, size_t* len); void RemoveStream(uint32_t ssrc) { @@ -76,9 +76,6 @@ namespace RTC private: // Allocated by this. srtp_t session{ nullptr }; - //rtp包最大1600 - static constexpr size_t EncryptBufferSize{ 2000 }; - uint8_t EncryptBuffer[EncryptBufferSize]; DepLibSRTP::Ptr _env; }; } // namespace RTC diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 38fecd1f..badd921c 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -266,25 +266,24 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * } } -void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) { - const uint8_t *p = (uint8_t *) buf; - bool ret = false; +void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, TrackType type) { if (_srtp_session_send) { - ret = _srtp_session_send->EncryptRtp(&p, &len, pt); - } - if (ret) { - onSendSockData((char *) p, len, flush); + CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); + memcpy(_srtp_buf, buf, len); + onBeforeEncryptRtp((char *) _srtp_buf, len, type); + if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) { + onSendSockData((char *) _srtp_buf, len, flush); + } } } void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){ - const uint8_t *p = (uint8_t *) buf; - bool ret = false; if (_srtp_session_send) { - ret = _srtp_session_send->EncryptRtcp(&p, &len); - } - if (ret) { - onSendSockData((char *) p, len, flush); + CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); + memcpy(_srtp_buf, buf, len); + if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) { + onSendSockData((char *) _srtp_buf, len, flush); + } } } @@ -611,6 +610,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { + DebugL << "\r\n" << rtcp->dumpString(); break; } default: break; @@ -663,8 +663,7 @@ static void setExtType(RtpExt &ext, RtpExtType tp) { } template -static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { - auto header = rtp->getHeader(); +static void changeRtpExtId(const RtpHeader *header, const Type &map) { auto ext_map = RtpExt::getExtValue(header); for (auto &pr : ext_map) { auto it = map.find((typename Type::key_type) (pr.first)); @@ -675,29 +674,35 @@ static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { } setExtType(pr.second, it->first); setExtType(pr.second, it->second); + DebugL << pr.second.dumpString(); pr.second.setExtId((uint8_t) it->second); } } void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { - changeRtpExtId(rtp, _rtp_ext_id_to_type); + changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type); //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - auto &pt = _send_rtp_pt[rtp->type]; + auto pt = _send_rtp_pt[rtp->type]; if (pt == 0xFF) { //忽略，对方不支持该编码类型 return; } - changeRtpExtId(rtp, _rtp_ext_type_to_id); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type); //统计rtp发送情况，好做sr汇报 _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } +void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) { + auto header = (RtpHeader *)buf; + header->pt = _send_rtp_pt[type]; + changeRtpExtId(header, _rtp_ext_type_to_id); +} + void WebRtcTransportImp::onShutdown(const SockException &ex){ InfoL << ex.what(); _self = nullptr; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 84fdbde5..44fdf973 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -60,9 +60,9 @@ public: * @param buf rtcp内容 * @param len rtcp长度 * @param flush 是否flush socket - * @param pt rtp payload type + * @param type rtp类型 */ - void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt); + void sendRtpPacket(char *buf, size_t len, bool flush, TrackType type); void sendRtcpPacket(char *buf, size_t len, bool flush); const EventPoller::Ptr& getPoller() const; @@ -100,6 +100,7 @@ protected: virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; virtual void onShutdown(const SockException &ex) = 0; + virtual void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) = 0; protected: const RtcSession& getSdp(SdpType type) const; @@ -112,6 +113,7 @@ private: void setRemoteDtlsFingerprint(const RtcSession &remote); private: + uint8_t _srtp_buf[2000]; EventPoller::Ptr _poller; std::shared_ptr _ice_server; std::shared_ptr _dtls_transport; @@ -150,6 +152,8 @@ protected: void onRtp(const char *buf, size_t len) override; void onRtcp(const char *buf, size_t len) override; + void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) override; + void onShutdown(const SockException &ex) override; ///////MediaSourceEvent override/////// From d74f2ff155240569a3effa43580fba61debc2442 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 21:43:27 +0800 Subject: [PATCH 167/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dremb=E5=92=8Ctwcc?= =?UTF-8?q?=E5=BC=80=E5=85=B3=E7=9B=B8=E5=85=B3bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 02ecef91..757aac80 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1349,7 +1349,7 @@ string const SdpConst::kRembRtcpFb = "goog-remb"; void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){ if (!enable) { rtcp_fb.erase(SdpConst::kTWCCRtcpFb); - extmap.erase(RtpExtType::abs_send_time); + extmap.erase(RtpExtType::transport_cc); } else { rtcp_fb.emplace(SdpConst::kTWCCRtcpFb); extmap.emplace(RtpExtType::transport_cc); @@ -1362,7 +1362,7 @@ void RtcConfigure::RtcTrackConfigure::enableREMB(bool enable){ extmap.erase(RtpExtType::abs_send_time); } else { rtcp_fb.emplace(SdpConst::kRembRtcpFb); - extmap.emplace(RtpExtType::transport_cc); + extmap.emplace(RtpExtType::abs_send_time); } } From 783321e74e88a1d3d1d0dd3200926b28b9127187 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 8 May 2021 21:47:17 +0800 Subject: [PATCH 168/218] =?UTF-8?q?=E5=85=B3=E9=97=AD=E8=B0=83=E8=AF=95?= =?UTF-8?q?=E6=97=A5=E5=BF=97?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index badd921c..e32fac33 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -610,7 +610,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - DebugL << "\r\n" << rtcp->dumpString(); +// DebugL << "\r\n" << rtcp->dumpString(); break; } default: break; @@ -674,7 +674,7 @@ static void changeRtpExtId(const RtpHeader *header, const Type &map) { } setExtType(pr.second, it->first); setExtType(pr.second, it->second); - DebugL << pr.second.dumpString(); +// DebugL << pr.second.dumpString(); pr.second.setExtId((uint8_t) it->second); } } From 7f917211e48ac9b2ede2f74175257f0f337bccca Mon Sep 17 00:00:00 2001 From: xgj Date: Sat, 8 May 2021 22:25:56 +0800 Subject: [PATCH 169/218] fix gcc for linux compile error --- webrtc/RtpExt.cpp | 2 +- webrtc/WebRtcTransport.cpp | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 5ea0192d..62cdc830 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -125,7 +125,7 @@ static bool isOneByteExt(){ } template<> -static bool isOneByteExt(){ +bool isOneByteExt(){ return true; } diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index e32fac33..db01b69c 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -576,7 +576,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); } else { - WarnL << "未识别的sr rtcp包:" << sr->ssrc; + WarnL << "未识别的sr rtcp包:" << (uint32_t)sr->ssrc; } break; } @@ -589,7 +589,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); } else { - WarnL << "未识别的rr rtcp包:" << rr->ssrc; + WarnL << "未识别的rr rtcp包:" << (uint32_t)rr->ssrc; } break; } From 45abc23e870ab3f85a8bcd5ff0c6671d3d75d390 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 10:14:00 +0800 Subject: [PATCH 170/218] =?UTF-8?q?=E6=94=BE=E5=AE=BDfci=E9=95=BF=E5=BA=A6?= =?UTF-8?q?=E6=A0=A1=E9=AA=8C?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 6acb4d3a..4164dd11 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -15,7 +15,7 @@ using namespace toolkit; namespace mediakit { void FCI_SLI::check(size_t size){ - CHECK(size == kSize); + CHECK(size >= kSize); } FCI_SLI::FCI_SLI(uint16_t first, uint16_t number, uint8_t pic_id) { @@ -48,7 +48,7 @@ string FCI_SLI::dumpString() const { /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void FCI_FIR::check(size_t size){ - CHECK(size == kSize); + CHECK(size >= kSize); } uint32_t FCI_FIR::getSSRC() const{ @@ -84,7 +84,7 @@ void FCI_REMB::check(size_t size){ CHECK(memcmp(magic, kRembMagic, sizeof(magic)) == 0); auto num_ssrc = bitrate[0]; auto expect_size = kSize + 4 * num_ssrc; - CHECK(size == expect_size); + CHECK(size >= expect_size); } string FCI_REMB::create(const vector &ssrcs, uint32_t bitrate) { @@ -168,7 +168,7 @@ FCI_NACK::FCI_NACK(uint16_t pid_h, const vector &type) { } void FCI_NACK::check(size_t size){ - CHECK(size == kSize); + CHECK(size >= kSize); } uint16_t FCI_NACK::getPid() const { From 985fe310b61d8ef758c8f6181f830e13b8a0df25 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 10:14:42 +0800 Subject: [PATCH 171/218] =?UTF-8?q?extmap=E6=94=B9=E5=9B=9Evector=E5=AD=98?= =?UTF-8?q?=E6=94=BE?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 19 ++++++------------- webrtc/Sdp.h | 2 +- webrtc/WebRtcTransport.cpp | 8 ++++---- 3 files changed, 11 insertions(+), 18 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 757aac80..6f08007e 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -292,7 +292,6 @@ string RtcSessionSdp::toString() const { ////////////////////////////////////////////////////////////////////////////////////////// #define SDP_THROW() throw std::invalid_argument(StrPrinter << "解析sdp " << getKey() << " 字段失败:" << str) -#define SDP_THROW2() throw std::invalid_argument(StrPrinter << "生成sdp " << getKey() << " 字段失败") void SdpTime::parse(const string &str) { if (sscanf(str.data(), "%" SCNu64 " %" SCNu64, &start, &stop) != 2) { @@ -842,13 +841,7 @@ void RtcSession::loadFrom(const string &str, bool check) { } rtc_media.rtcp_addr = media.getItemClass('a', "rtcp"); rtc_media.direction = media.getDirection(); - { - rtc_media.extmap.clear(); - auto arr = media.getAllItem('a', "extmap"); - for (auto &ext : arr) { - rtc_media.extmap.emplace(ext.id, ext); - } - } + rtc_media.extmap = media.getAllItem('a', "extmap"); rtc_media.rtcp_mux = media.getItem('a', "rtcp-mux").operator bool(); rtc_media.rtcp_rsize = media.getItem('a', "rtcp-rsize").operator bool(); @@ -1111,8 +1104,8 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{ if (m.ice_lite) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared("ice-lite"))); } - for (auto &pr : m.extmap) { - sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(pr.second))); + for (auto &ext : m.extmap) { + sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(ext))); } if (m.direction != RtpDirection::invalid) { sdp_media.items.emplace_back(wrapSdpAttr(std::make_shared(m.direction))); @@ -1664,9 +1657,9 @@ RETRY: } //对方和我方都支持的扩展，那么我们才支持 - for (auto &pr : offer_media.extmap) { - if (configure.extmap.find(RtpExt::getExtType(pr.second.ext)) != configure.extmap.end()) { - answer_media.extmap.emplace(pr); + for (auto &ext : offer_media.extmap) { + if (configure.extmap.find(RtpExt::getExtType(ext.ext)) != configure.extmap.end()) { + answer_media.extmap.emplace_back(ext); } } diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 3a111649..34c510ef 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -633,7 +633,7 @@ public: SdpAttrFingerprint fingerprint; //////// extmap //////// - map extmap; + vector extmap; //////// sctp //////////// SdpAttrSctpMap sctpmap; diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index db01b69c..d7aa1a49 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -414,10 +414,10 @@ void WebRtcTransportImp::onStartWebRTC() { } if (m.type != TrackApplication) { //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id - for (auto &pr : m.extmap) { - auto ext_type = RtpExt::getExtType(pr.second.ext); - _rtp_ext_id_to_type.emplace(pr.second.id, ext_type); - _rtp_ext_type_to_id.emplace(ext_type, pr.second.id); + for (auto &ext : m.extmap) { + auto ext_type = RtpExt::getExtType(ext.ext); + _rtp_ext_id_to_type.emplace(ext.id, ext_type); + _rtp_ext_type_to_id.emplace(ext_type, ext.id); } } } From d4ce5b0091412c96f183330b52347b7ca7b6dc4f Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 17:35:22 +0800 Subject: [PATCH 172/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0ssrc=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E7=9A=84=E6=96=B9=E6=B3=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 18 ++++++++++++++++++ webrtc/Sdp.h | 5 +++++ 2 files changed, 23 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 6f08007e..03cb5e9a 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1290,6 +1290,9 @@ void RtcMedia::checkValid() const{ CHECK(!proto.empty()); CHECK(direction != RtpDirection::invalid || type == TrackApplication); CHECK(!plan.empty() || type == TrackApplication ); +} + +void RtcMedia::checkValidSSRC() const { bool send_rtp = (direction == RtpDirection::sendonly || direction == RtpDirection::sendrecv); if (rtp_rids.empty() && rtp_ssrc_sim.empty()) { //非simulcast时，检查有没有指定rtp ssrc @@ -1318,6 +1321,12 @@ void RtcSession::checkValid() const{ } } +void RtcSession::checkValidSSRC() const{ + for (auto &item : media) { + item.checkValidSSRC(); + } +} + const RtcMedia *RtcSession::getMedia(TrackType type) const{ for(auto &m : media){ if(m.type == type){ @@ -1327,6 +1336,15 @@ const RtcMedia *RtcSession::getMedia(TrackType type) const{ return nullptr; } +bool RtcSession::haveSSRC() const { + for (auto &m : media) { + if (!m.rtp_rtx_ssrc.empty()) { + return true; + } + } + return false; +} + bool RtcSession::supportRtcpFb(const string &name, TrackType type) const { auto media = getMedia(type); if (!media) { diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 34c510ef..1ddfce94 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -640,6 +640,8 @@ public: uint32_t sctp_port{0}; void checkValid() const; + //offer sdp,如果指定了发送rtp,那么应该指定ssrc + void checkValidSSRC() const; const RtcCodecPlan *getPlan(uint8_t pt) const; const RtcCodecPlan *getPlan(const char *codec) const; const RtcCodecPlan *getRelatedRtxPlan(uint8_t pt) const; @@ -662,9 +664,12 @@ public: void loadFrom(const string &sdp, bool check = true); void checkValid() const; + //offer sdp,如果指定了发送rtp,那么应该指定ssrc + void checkValidSSRC() const; string toString() const; string toRtspSdp() const; const RtcMedia *getMedia(TrackType type) const; + bool haveSSRC() const; bool supportRtcpFb(const string &name, TrackType type = TrackType::TrackVideo) const; private: From a42e5f6470000c346cf091b2517202abb39aa08c Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 18:11:12 +0800 Subject: [PATCH 173/218] =?UTF-8?q?=E5=AE=8C=E5=96=84ssrc=E5=A4=84?= =?UTF-8?q?=E7=90=86=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 43 ++++++++++++++++++++++++++++---------- webrtc/WebRtcTransport.h | 11 +++++----- 2 files changed, 38 insertions(+), 16 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index d7aa1a49..bec317f5 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -185,6 +185,9 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ if (sdp.group.mids.empty()) { throw std::invalid_argument("只支持group BUNDLE模式"); } + if (type == SdpType::offer) { + sdp.checkValidSSRC(); + } } void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const { @@ -390,6 +393,16 @@ bool WebRtcTransportImp::canRecvRtp() const{ return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly); } +const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{ + auto &offer = getSdp(SdpType::offer); + if (offer.haveSSRC()) { + return offer; + } + auto &answer = getSdp(SdpType::answer); + CHECK(answer.haveSSRC()); + return answer; +} + void WebRtcTransportImp::onStartWebRTC() { //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 for (auto &m : getSdp(SdpType::offer).media) { @@ -398,11 +411,12 @@ void WebRtcTransportImp::onStartWebRTC() { if (!hit_pan) { continue; } + auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); //获取offer端rtp的ssrc和pt相关信息 auto &ref = _rtp_info_pt[plan.pt]; - _rtp_info_ssrc[m.rtp_rtx_ssrc[0].ssrc] = &ref; + _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; ref.plan = &plan; - ref.media = &m; + ref.media = m_with_ssrc; ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); @@ -441,14 +455,16 @@ void WebRtcTransportImp::onStartWebRTC() { RtcSession rtsp_send_sdp; rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); - for (auto &m : getSdp(SdpType::answer).media) { + for (auto &m : getSdp(SdpType::answer).media) { if (m.type == TrackApplication) { continue; } auto rtsp_media = rtsp_send_sdp.getMedia(m.type); if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { - //记录发送rtp时约定的pt，届时发送rtp时需要修改pt - _send_rtp_pt[m.type] = m.plan[0].pt; + auto it = _rtp_info_pt.find(m.plan[0].pt); + CHECK(it != _rtp_info_pt.end()); + //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc + _send_rtp_info[m.type] = &it->second; } } } @@ -457,10 +473,11 @@ void WebRtcTransportImp::onStartWebRTC() { void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer) { + //我们只修改answer sdp return; } - //修改sdp的ip、端口信息 + //修改answer sdp的ip、端口信息 GET_CONFIG(string, extern_ip, RTC::kExternIP); for (auto &m : sdp.media) { m.addr.reset(); @@ -472,7 +489,8 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ sdp.origin.address = m.addr.address; } - if (!canSendRtp()) { + if (!canSendRtp() || getSdp(SdpType::offer).haveSSRC()) { + //offer sdp未包含ssrc相关信息，那么我们才在answer sdp中回复ssrc相关信息 return; } @@ -686,20 +704,23 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp } void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - auto pt = _send_rtp_pt[rtp->type]; - if (pt == 0xFF) { + auto info = _send_rtp_info[rtp->type]; + if (!info) { //忽略，对方不支持该编码类型 return; } _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type); //统计rtp发送情况，好做sr汇报 - _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + _rtp_info_pt[info->plan->pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) { auto header = (RtpHeader *)buf; - header->pt = _send_rtp_pt[type]; + auto info = _send_rtp_info[type]; + //修改目标pt和ssrc + header->pt = info->plan->pt; + header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc); changeRtpExtId(header, _rtp_ext_type_to_id); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 44fdf973..9e6116c1 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -188,6 +188,7 @@ private: SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; + const RtcSession& getSdpWithSSRC() const; class RtpPayloadInfo { public: @@ -215,8 +216,8 @@ private: Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; - //记录协商的rtp的pt类型 - uint8_t _send_rtp_pt[2] = {0xFF, 0xFF}; + //记录协商的发送rtp的pt和ssrc + RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr}; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 @@ -226,9 +227,9 @@ private: //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 - unordered_map _rtp_info_pt; - //根据推流端rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map _rtp_info_pt; + //根据rtcp的ssrc获取相关信息 + unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id From c57f31d3d05941eae26137bac4e6e24cad136d7b Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 18:14:51 +0800 Subject: [PATCH 174/218] =?UTF-8?q?=E4=BB=A3=E7=A0=81=E4=BC=98=E5=8C=96?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index bec317f5..38677d45 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -712,7 +712,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type); //统计rtp发送情况，好做sr汇报 - _rtp_info_pt[info->plan->pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) { From ec8e518180da7bfffa7b4eb6b61ed2c81bb5e36f Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Mon, 10 May 2021 23:27:11 +0800 Subject: [PATCH 175/218] =?UTF-8?q?=E5=AE=8C=E5=96=84fci=E7=9B=B8=E5=85=B3?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/Rtcp.cpp | 44 +++++++++++++++++++++++--------------------- src/Rtcp/Rtcp.h | 24 ++++++++++++++++++++++++ 2 files changed, 47 insertions(+), 21 deletions(-) diff --git a/src/Rtcp/Rtcp.cpp b/src/Rtcp/Rtcp.cpp index 4b7dbd25..c7c5a277 100644 --- a/src/Rtcp/Rtcp.cpp +++ b/src/Rtcp/Rtcp.cpp @@ -517,44 +517,48 @@ std::shared_ptr RtcpFB::create(RTPFBType fmt, const void *fci, size_t fc return RtcpFB::create_l(RtcpType::RTCP_RTPFB, (int)fmt, fci, fci_len); } +const void *RtcpFB::getFciPtr() const { + return (uint8_t *) &ssrc_media + sizeof(ssrc_media); +} + +size_t RtcpFB::getFciSize() const { + auto fci_len = (ssize_t) getSize() - getPaddingSize() - sizeof(RtcpFB); + CHECK(fci_len >= 0); + return fci_len; +} + string RtcpFB::dumpString() const { _StrPrinter printer; printer << RtcpHeader::dumpHeader(); printer << "ssrc:" << ssrc << "\r\n"; printer << "ssrc_media:" << ssrc_media << "\r\n"; - auto fci_data = (uint8_t *)&ssrc_media + sizeof(ssrc_media); - auto fci_len = (ssize_t)getSize() - getPaddingSize() - sizeof(RtcpFB); - CHECK(fci_len >= 0); switch ((RtcpType) pt) { case RtcpType::RTCP_PSFB : { switch ((PSFBType) report_count) { case PSFBType::RTCP_PSFB_SLI : { - FCI_SLI *fci = (FCI_SLI *) fci_data; - fci->check(fci_len); - printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + auto &fci = getFci(); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci.dumpString(); break; } case PSFBType::RTCP_PSFB_PLI : { - CHECK(fci_len == 0); + getFciSize(); printer << "fci:" << psfbTypeToStr((PSFBType) report_count); break; } case PSFBType::RTCP_PSFB_FIR : { - FCI_FIR *fci = (FCI_FIR *) fci_data; - fci->check(fci_len); - printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + auto &fci = getFci(); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci.dumpString(); break; } case PSFBType::RTCP_PSFB_REMB : { - FCI_REMB *fci = (FCI_REMB *) fci_data; - fci->check(fci_len); - printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString(); + auto &fci = getFci(); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci.dumpString(); break; } default:{ - printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << hexdump(fci_data, fci_len); + printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << hexdump(getFciPtr(), getFciSize()); break; } } @@ -563,19 +567,17 @@ string RtcpFB::dumpString() const { case RtcpType::RTCP_RTPFB : { switch ((RTPFBType) report_count) { case RTPFBType::RTCP_RTPFB_NACK : { - FCI_NACK *fci = (FCI_NACK *) fci_data; - fci->check(fci_len); - printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString(); + auto &fci = getFci(); + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci.dumpString(); break; } case RTPFBType::RTCP_RTPFB_TWCC : { - FCI_TWCC *fci = (FCI_TWCC *) fci_data; - fci->check(fci_len); - printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString(fci_len); + auto &fci = getFci(); + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci.dumpString(getFciSize()); break; } default: { - printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << hexdump(fci_data, fci_len); + printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << hexdump(getFciPtr(), getFciSize()); break; } } diff --git a/src/Rtcp/Rtcp.h b/src/Rtcp/Rtcp.h index c4055f71..efb591c1 100644 --- a/src/Rtcp/Rtcp.h +++ b/src/Rtcp/Rtcp.h @@ -586,6 +586,30 @@ public: */ static std::shared_ptr create(RTPFBType fmt, const void *fci = nullptr, size_t fci_len = 0); + /** + * fci转换成某对象指针 + * @tparam Type 对象类型 + * @return 对象指针 + */ + template + const Type& getFci() const{ + auto fci_data = getFciPtr(); + auto fci_len = getFciSize(); + Type *fci = (Type *) fci_data; + fci->check(fci_len); + return *fci; + } + + /** + * 获取fci指针 + */ + const void *getFciPtr() const; + + /** + * 获取fci数据长度 + */ + size_t getFciSize() const; + private: /** * 打印字段详情 From e8d3dec050786e2da568ce2b77dd25a5b2356217 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 11 May 2021 00:34:17 +0800 Subject: [PATCH 176/218] =?UTF-8?q?getBitArray=E6=96=B9=E6=B3=95=E8=BF=94?= =?UTF-8?q?=E5=9B=9Enack=E7=AC=AC=E4=B8=80=E4=B8=AA=E5=8C=85=E7=8A=B6?= =?UTF-8?q?=E6=80=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 4164dd11..67d82ca8 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -181,17 +181,20 @@ uint16_t FCI_NACK::getBlp() const { vector FCI_NACK::getBitArray() const { vector ret; - ret.resize(kBitSize); + ret.resize(kBitSize + 1); + //nack第一个包丢包 + ret[0] = false; + auto blp_h = getBlp(); for (size_t i = 0; i < kBitSize; ++i) { - ret[i] = blp_h & (1 << (kBitSize - i - 1)); + ret[i + 1] = blp_h & (1 << (kBitSize - i - 1)); } return ret; } string FCI_NACK::dumpString() const { _StrPrinter printer; - printer << "pid:" << getPid() << ",blp:"; + printer << "pid:" << getPid() << ",blp:" << getBlp() << ",bit array:"; for (auto flag : getBitArray()) { printer << flag << " "; } From 7ad361b22dd27615d7fc7c3caa1bdce7fcc096f6 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Tue, 11 May 2021 00:54:33 +0800 Subject: [PATCH 177/218] =?UTF-8?q?rtc=E6=92=AD=E6=94=BE=E6=94=AF=E6=8C=81?= =?UTF-8?q?nack=E9=87=8D=E4=BC=A0?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 58 +++++++++++++++++++++----- webrtc/WebRtcTransport.h | 83 ++++++++++++++++++++++++++++++++++---- 2 files changed, 122 insertions(+), 19 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 38677d45..9f9a305a 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -269,21 +269,22 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * } } -void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, TrackType type) { +void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx) { if (_srtp_session_send) { CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); memcpy(_srtp_buf, buf, len); - onBeforeEncryptRtp((char *) _srtp_buf, len, type); + onBeforeEncryptRtp((char *) _srtp_buf, len, ctx); if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) { onSendSockData((char *) _srtp_buf, len, flush); } } } -void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){ +void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx){ if (_srtp_session_send) { CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); memcpy(_srtp_buf, buf, len); + onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx); if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) { onSendSockData((char *) _srtp_buf, len, flush); } @@ -414,10 +415,13 @@ void WebRtcTransportImp::onStartWebRTC() { auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); //获取offer端rtp的ssrc和pt相关信息 auto &ref = _rtp_info_pt[plan.pt]; - _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; ref.plan = &plan; ref.media = m_with_ssrc; ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + if (ref.is_common_rtp) { + //rtp + _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; + } ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { @@ -628,7 +632,30 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { -// DebugL << "\r\n" << rtcp->dumpString(); + RtcpFB *fb = (RtcpFB *) rtcp; + auto it = _rtp_info_ssrc.find(fb->ssrc); + if (it == _rtp_info_ssrc.end()) { + WarnL << "未识别的 rtcp包:" << rtcp->dumpString(); + return; + } + if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) { +// DebugL << "\r\n" << rtcp->dumpString(); + break; + } + //RTPFB + switch ((RTPFBType) rtcp->report_count) { + case RTPFBType::RTCP_RTPFB_NACK : { + auto &fci = fb->getFci(); + it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { + //rtp重传 + onSendRtp(rtp, true, true); + }); + break; + } + default: +// DebugL << "\r\n" << rtcp->dumpString(); + break; + } break; } default: break; @@ -703,27 +730,36 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } -void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ +void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){ auto info = _send_rtp_info[rtp->type]; if (!info) { //忽略，对方不支持该编码类型 return; } + if (!rtx) { + //统计rtp发送情况，好做sr汇报 + info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + info->nack_list.push_back(rtp); + } else { + WarnL << "重传rtp:" << rtp->getSeq(); + } + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type); - //统计rtp发送情况，好做sr汇报 - info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } -void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) { +void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) { + RtpPayloadInfo *info = reinterpret_cast(ctx); auto header = (RtpHeader *)buf; - auto info = _send_rtp_info[type]; //修改目标pt和ssrc header->pt = info->plan->pt; header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc); changeRtpExtId(header, _rtp_ext_type_to_id); } +void WebRtcTransportImp::onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) { + +} + void WebRtcTransportImp::onShutdown(const SockException &ex){ InfoL << ex.what(); _self = nullptr; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 9e6116c1..7b911020 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -21,6 +21,7 @@ #include "Network/Socket.h" #include "Rtsp/RtspMediaSourceImp.h" #include "Rtcp/RtcpContext.h" +#include "Rtcp/RtcpFCI.h" using namespace toolkit; using namespace mediakit; @@ -60,10 +61,10 @@ public: * @param buf rtcp内容 * @param len rtcp长度 * @param flush 是否flush socket - * @param type rtp类型 + * @param ctx 用户指针 */ - void sendRtpPacket(char *buf, size_t len, bool flush, TrackType type); - void sendRtcpPacket(char *buf, size_t len, bool flush); + void sendRtpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr); + void sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr); const EventPoller::Ptr& getPoller() const; @@ -100,7 +101,8 @@ protected: virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; virtual void onShutdown(const SockException &ex) = 0; - virtual void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) = 0; + virtual void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) = 0; + virtual void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) = 0; protected: const RtcSession& getSdp(SdpType type) const; @@ -125,6 +127,69 @@ private: class RtpReceiverImp; +class NackList { +public: + void push_back(RtpPacket::Ptr rtp) { + auto seq = rtp->getSeq(); + nack_cache_seq.emplace_back(seq); + nack_cache_pkt.emplace(seq, std::move(rtp)); + while (get_cache_ms() > kMaxNackMS) { + //需要清除部分nack缓存 + pop_front(); + } + } + + template + void for_each_nack(const FCI_NACK &nack, const FUNC &func) { + auto seq = nack.getPid(); + for (auto bit : nack.getBitArray()) { + if (!bit) { + //丢包 + RtpPacket::Ptr *ptr = get_rtp(seq); + if (ptr) { + func(*ptr); + } + } + ++seq; + } + } + +private: + void pop_front() { + if (nack_cache_seq.empty()) { + return; + } + nack_cache_pkt.erase(nack_cache_seq.front()); + nack_cache_seq.pop_front(); + } + + RtpPacket::Ptr *get_rtp(uint16_t seq) { + auto it = nack_cache_pkt.find(seq); + if (it == nack_cache_pkt.end()) { + return nullptr; + } + return &it->second; + } + + uint32_t get_cache_ms() { + if (nack_cache_seq.size() < 2) { + return 0; + } + uint32_t back = nack_cache_pkt[nack_cache_seq.back()]->getStampMS(); + uint32_t front = nack_cache_pkt[nack_cache_seq.front()]->getStampMS(); + if (back > front) { + return back - front; + } + //很有可能回环了 + return back + (UINT32_MAX - front); + } + +private: + static constexpr uint32_t kMaxNackMS = 10 * 1000; + deque nack_cache_seq; + unordered_map nack_cache_pkt; +}; + class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; @@ -152,7 +217,8 @@ protected: void onRtp(const char *buf, size_t len) override; void onRtcp(const char *buf, size_t len) override; - void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) override; + void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) override; + void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) override; void onShutdown(const SockException &ex) override; @@ -184,7 +250,7 @@ private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onCreate() override; void onDestory() override; - void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); + void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false); SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; @@ -198,10 +264,11 @@ private: std::shared_ptr receiver; RtcpContext::Ptr rtcp_context_recv; RtcpContext::Ptr rtcp_context_send; + NackList nack_list; }; - void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp); - void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); + void onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp); + void onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); private: //用掉的总流量 From cbafdbabc6cc09eaaee0222a1e8f169d4e1728fc Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 11 May 2021 09:09:35 +0800 Subject: [PATCH 178/218] =?UTF-8?q?=E9=98=B2=E6=AD=A2=E6=8B=B7=E8=B4=9D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9f9a305a..f06a20be 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -426,7 +426,7 @@ void WebRtcTransportImp::onStartWebRTC() { ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { onSortedRtp(ref, std::move(rtp)); - }, [ref, this](const RtpPacket::Ptr &rtp) { + }, [&ref, this](const RtpPacket::Ptr &rtp) { onBeforeSortedRtp(ref, rtp); }); } From 5c90a1e137c724a75d50637dfb626c99ded33ba2 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 11 May 2021 11:18:55 +0800 Subject: [PATCH 179/218] =?UTF-8?q?=E5=AE=8C=E5=96=84nack=E4=B8=8E?= =?UTF-8?q?=E4=B8=A2=E5=8C=85=E9=87=8D=E4=BC=A0?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtcp/RtcpFCI.cpp | 2 +- src/Rtcp/RtcpFCI.h | 5 +- src/Rtsp/RtpReceiver.h | 2 +- tests/test_rtcp_nack.cpp | 37 +++++++++++++ webrtc/WebRtcTransport.cpp | 31 ++++++++--- webrtc/WebRtcTransport.h | 104 +++++++++++++++++++++++++++++++------ 6 files changed, 155 insertions(+), 26 deletions(-) create mode 100644 tests/test_rtcp_nack.cpp diff --git a/src/Rtcp/RtcpFCI.cpp b/src/Rtcp/RtcpFCI.cpp index 67d82ca8..dd7f2457 100644 --- a/src/Rtcp/RtcpFCI.cpp +++ b/src/Rtcp/RtcpFCI.cpp @@ -183,7 +183,7 @@ vector FCI_NACK::getBitArray() const { vector ret; ret.resize(kBitSize + 1); //nack第一个包丢包 - ret[0] = false; + ret[0] = true; auto blp_h = getBlp(); for (size_t i = 0; i < kBitSize; ++i) { diff --git a/src/Rtcp/RtcpFCI.h b/src/Rtcp/RtcpFCI.h index a3d5e96e..1b68e457 100644 --- a/src/Rtcp/RtcpFCI.h +++ b/src/Rtcp/RtcpFCI.h @@ -243,18 +243,17 @@ private: class FCI_NACK { public: static constexpr size_t kSize = 4; + static constexpr size_t kBitSize = 16; FCI_NACK(uint16_t pid_h, const vector &type); void check(size_t size); uint16_t getPid() const; uint16_t getBlp() const; + //返回丢包列表，总长度17，第一个包必丢 vector getBitArray() const; string dumpString() const; -private: - static constexpr size_t kBitSize = 16; - private: // The PID field is used to specify a lost packet. The PID field // refers to the RTP sequence number of the lost packet. diff --git a/src/Rtsp/RtpReceiver.h b/src/Rtsp/RtpReceiver.h index a5f1eaa0..24cf8fb8 100644 --- a/src/Rtsp/RtpReceiver.h +++ b/src/Rtsp/RtpReceiver.h @@ -21,7 +21,7 @@ using namespace toolkit; namespace mediakit { -template +template class PacketSortor { public: PacketSortor() = default; diff --git a/tests/test_rtcp_nack.cpp b/tests/test_rtcp_nack.cpp new file mode 100644 index 00000000..7463ce02 --- /dev/null +++ b/tests/test_rtcp_nack.cpp @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2016 The ZLToolKit project authors. All Rights Reserved. + * + * This file is part of ZLToolKit(https://github.com/xia-chu/ZLToolKit). + * + * Use of this source code is governed by MIT license that can be found in the + * LICENSE file in the root of the source tree. All contributing project authors + * may be found in the AUTHORS file in the root of the source tree. + */ + +#include +#include "Util/logger.h" +#include "Rtcp/RtcpFCI.h" +#include "../webrtc/WebRtcTransport.h" +using namespace std; +using namespace toolkit; +using namespace mediakit; + +extern void testFCI(); + +int main() { + Logger::Instance().add(std::make_shared()); + + srand((unsigned) time(NULL)); + + NackContext ctx; + for (int i = 1; i < 1000; ++i) { + if (i % (1 + (rand() % 30)) == 0) { + DebugL << "drop:" << i; + } else { + ctx.received(i); + + } + } + sleep(1); + return 0; +} diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index f06a20be..1cd574c8 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -424,9 +424,9 @@ void WebRtcTransportImp::onStartWebRTC() { } ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); - ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) { + ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) mutable{ onSortedRtp(ref, std::move(rtp)); - }, [&ref, this](const RtpPacket::Ptr &rtp) { + }, [&ref, this](const RtpPacket::Ptr &rtp) mutable { onBeforeSortedRtp(ref, rtp); }); } @@ -674,15 +674,28 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { return; } auto &info = it->second; + +#if 1 + //此处模拟接受丢包 + auto header = (RtpHeader *) buf; + auto seq = ntohs(header->seq); + if (seq % 10 == 0) { + //丢包 + return; + } else { + info.nack_ctx.received(seq); + } +#endif + //解析并排序rtp info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); } /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) { - if(!info.is_common_rtp){ - //todo rtx/red/ulpfec类型的rtp先未处理 +void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { + if (!info.is_common_rtp) { + WarnL; return; } if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { @@ -724,7 +737,7 @@ static void changeRtpExtId(const RtpHeader *header, const Type &map) { } } -void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { +void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type); //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); @@ -740,6 +753,12 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r //统计rtp发送情况，好做sr汇报 info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); info->nack_list.push_back(rtp); +#if 0 + //此处模拟发送丢包 + if(rtp->getSeq() % 10 == 0){ + return; + } +#endif } else { WarnL << "重传rtp:" << rtp->getSeq(); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 7b911020..caeb80a6 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -131,8 +131,8 @@ class NackList { public: void push_back(RtpPacket::Ptr rtp) { auto seq = rtp->getSeq(); - nack_cache_seq.emplace_back(seq); - nack_cache_pkt.emplace(seq, std::move(rtp)); + _nack_cache_seq.emplace_back(seq); + _nack_cache_pkt.emplace(seq, std::move(rtp)); while (get_cache_ms() > kMaxNackMS) { //需要清除部分nack缓存 pop_front(); @@ -143,7 +143,7 @@ public: void for_each_nack(const FCI_NACK &nack, const FUNC &func) { auto seq = nack.getPid(); for (auto bit : nack.getBitArray()) { - if (!bit) { + if (bit) { //丢包 RtpPacket::Ptr *ptr = get_rtp(seq); if (ptr) { @@ -156,27 +156,27 @@ public: private: void pop_front() { - if (nack_cache_seq.empty()) { + if (_nack_cache_seq.empty()) { return; } - nack_cache_pkt.erase(nack_cache_seq.front()); - nack_cache_seq.pop_front(); + _nack_cache_pkt.erase(_nack_cache_seq.front()); + _nack_cache_seq.pop_front(); } RtpPacket::Ptr *get_rtp(uint16_t seq) { - auto it = nack_cache_pkt.find(seq); - if (it == nack_cache_pkt.end()) { + auto it = _nack_cache_pkt.find(seq); + if (it == _nack_cache_pkt.end()) { return nullptr; } return &it->second; } uint32_t get_cache_ms() { - if (nack_cache_seq.size() < 2) { + if (_nack_cache_seq.size() < 2) { return 0; } - uint32_t back = nack_cache_pkt[nack_cache_seq.back()]->getStampMS(); - uint32_t front = nack_cache_pkt[nack_cache_seq.front()]->getStampMS(); + uint32_t back = _nack_cache_pkt[_nack_cache_seq.back()]->getStampMS(); + uint32_t front = _nack_cache_pkt[_nack_cache_seq.front()]->getStampMS(); if (back > front) { return back - front; } @@ -186,8 +186,81 @@ private: private: static constexpr uint32_t kMaxNackMS = 10 * 1000; - deque nack_cache_seq; - unordered_map nack_cache_pkt; + deque _nack_cache_seq; + unordered_map _nack_cache_pkt; +}; + +class NackContext { +public: + void received(uint16_t seq) { + if (!_last_max_seq && _seq.empty()) { + _last_max_seq = seq - 1; + } + _seq.emplace(seq); + auto max_seq = *_seq.rbegin(); + auto min_seq = *_seq.begin(); + auto diff = max_seq - min_seq; + if (!diff) { + return; + } + + if (diff > UINT32_MAX / 2) { + //回环 + _seq.clear(); + _last_max_seq = min_seq; + return; + } + + if (_seq.size() == diff + 1 && _last_max_seq + 1 == min_seq) { + //都是连续的seq，未丢包 + _seq.clear(); + _last_max_seq = max_seq; + } else { + //seq不连续，有丢包 + if (min_seq == _last_max_seq + 1) { + //前面部分seq是连续的，未丢包，移除之 + eraseFrontSeq(); + } + + //有丢包，丢包从_last_max_seq开始 + if (max_seq - _last_max_seq > FCI_NACK::kBitSize) { + vector vec; + vec.resize(FCI_NACK::kBitSize); + for (auto i = 0; i < FCI_NACK::kBitSize; ++i) { + vec[i] = _seq.find(_last_max_seq + i + 2) == _seq.end(); + } + onNack(FCI_NACK(_last_max_seq + 1, vec)); + _last_max_seq += FCI_NACK::kBitSize + 1; + if (_last_max_seq >= max_seq) { + _seq.clear(); + } else { + auto it = _seq.emplace_hint(_seq.begin(), _last_max_seq); + _seq.erase(_seq.begin(), it); + } + } + } + } + + void onNack(const FCI_NACK &nack) { + InfoL << nack.dumpString() << " " << _seq.size(); + } + +private: + void eraseFrontSeq(){ + //前面部分seq是连续的，未丢包，移除之 + for (auto it = _seq.begin(); it != _seq.end();) { + if (*it != _last_max_seq + 1) { + //seq不连续，丢包了 + break; + } + _last_max_seq = *it; + it = _seq.erase(it); + } + } + +private: + set _seq; + uint16_t _last_max_seq = 0; }; class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this{ @@ -265,10 +338,11 @@ private: RtcpContext::Ptr rtcp_context_recv; RtcpContext::Ptr rtcp_context_send; NackList nack_list; + NackContext nack_ctx; }; - void onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp); - void onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); + void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); + void onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); private: //用掉的总流量 From 0d61de758e5620a1e4a5cc89e20ec1814f9176a0 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 11 May 2021 12:12:28 +0800 Subject: [PATCH 180/218] =?UTF-8?q?=E5=AE=8C=E5=96=84nack=E4=B8=8E?= =?UTF-8?q?=E4=B8=A2=E5=8C=85=E9=87=8D=E4=BC=A02?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 31 +++++++++++++++++++++++++------ webrtc/WebRtcTransport.h | 16 +++++++++++++--- 2 files changed, 38 insertions(+), 9 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 1cd574c8..0b1f4372 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -429,6 +429,9 @@ void WebRtcTransportImp::onStartWebRTC() { }, [&ref, this](const RtpPacket::Ptr &rtp) mutable { onBeforeSortedRtp(ref, rtp); }); + ref.nack_ctx.setOnNack([&ref, this](const FCI_NACK &nack) mutable{ + onNack(ref, nack); + }); } if (m.type != TrackApplication) { //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id @@ -676,14 +679,21 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { auto &info = it->second; #if 1 - //此处模拟接受丢包 auto header = (RtpHeader *) buf; auto seq = ntohs(header->seq); - if (seq % 10 == 0) { - //丢包 - return; + if (info.is_common_rtp) { + //此处模拟接受丢包 + if (info.media->type == TrackVideo && seq % 10 == 0) { + //丢包 + DebugL << "模拟接受丢包:" << seq; + return; + } else { + info.nack_ctx.received(seq); + } } else { - info.nack_ctx.received(seq); + //收到重传包 + header->ssrc = info.media->rtp_rtx_ssrc[0].ssrc; + InfoL << "收到重传包:" << seq; } #endif @@ -691,6 +701,14 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); } +void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { + auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize); + rtcp->ssrc = htons(0); + rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc); + InfoL << rtcp->RtcpHeader::dumpString(); + sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true); +} + /////////////////////////////////////////////////////////////////// void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { @@ -756,11 +774,12 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r #if 0 //此处模拟发送丢包 if(rtp->getSeq() % 10 == 0){ + DebugL << "模拟发送丢包:" << rtp->getSeq(); return; } #endif } else { - WarnL << "重传rtp:" << rtp->getSeq(); + WarnL << "rtp发送重传:" << rtp->getSeq(); } sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index caeb80a6..beebd16c 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -192,6 +192,8 @@ private: class NackContext { public: + using onNack = function; + void received(uint16_t seq) { if (!_last_max_seq && _seq.empty()) { _last_max_seq = seq - 1; @@ -229,7 +231,7 @@ public: for (auto i = 0; i < FCI_NACK::kBitSize; ++i) { vec[i] = _seq.find(_last_max_seq + i + 2) == _seq.end(); } - onNack(FCI_NACK(_last_max_seq + 1, vec)); + doNack(FCI_NACK(_last_max_seq + 1, vec)); _last_max_seq += FCI_NACK::kBitSize + 1; if (_last_max_seq >= max_seq) { _seq.clear(); @@ -241,11 +243,17 @@ public: } } - void onNack(const FCI_NACK &nack) { - InfoL << nack.dumpString() << " " << _seq.size(); + void setOnNack(onNack cb) { + _cb = std::move(cb); } private: + void doNack(const FCI_NACK &nack) { + if (_cb) { + _cb(nack); + } + } + void eraseFrontSeq(){ //前面部分seq是连续的，未丢包，移除之 for (auto it = _seq.begin(); it != _seq.end();) { @@ -259,6 +267,7 @@ private: } private: + onNack _cb; set _seq; uint16_t _last_max_seq = 0; }; @@ -343,6 +352,7 @@ private: void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); void onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); + void onNack(RtpPayloadInfo &info, const FCI_NACK &nack); private: //用掉的总流量 From cb280e1eebb5e39df0eb28cdbfa629c378b58712 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Tue, 11 May 2021 15:48:01 +0800 Subject: [PATCH 181/218] =?UTF-8?q?rtc=E6=8E=A8=E6=B5=81=E6=94=AF=E6=8C=81?= =?UTF-8?q?nack/rtx=E9=87=8D=E4=BC=A0?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 163 ++++++++++++++++++++----------------- webrtc/WebRtcTransport.h | 3 + 2 files changed, 93 insertions(+), 73 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 0b1f4372..36738ebb 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -421,6 +421,10 @@ void WebRtcTransportImp::onStartWebRTC() { if (ref.is_common_rtp) { //rtp _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; + } else { + //rtx + auto apt = atoi(plan.getFmtp("apt").data()); + ref.plan_apt = m_with_ssrc->getPlan(apt); } ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); @@ -642,7 +646,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { return; } if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) { -// DebugL << "\r\n" << rtcp->dumpString(); break; } //RTPFB @@ -655,9 +658,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { }); break; } - default: -// DebugL << "\r\n" << rtcp->dumpString(); - break; + default: break; } break; } @@ -666,73 +667,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } } -void WebRtcTransportImp::onRtp(const char *buf, size_t len) { - _bytes_usage += len; - _alive_ticker.resetTime(); - RtpHeader *rtp = (RtpHeader *) buf; - //根据接收到的rtp的pt信息，找到该流的信息 - auto it = _rtp_info_pt.find(rtp->pt); - if (it == _rtp_info_pt.end()) { - WarnL; - return; - } - auto &info = it->second; - -#if 1 - auto header = (RtpHeader *) buf; - auto seq = ntohs(header->seq); - if (info.is_common_rtp) { - //此处模拟接受丢包 - if (info.media->type == TrackVideo && seq % 10 == 0) { - //丢包 - DebugL << "模拟接受丢包:" << seq; - return; - } else { - info.nack_ctx.received(seq); - } - } else { - //收到重传包 - header->ssrc = info.media->rtp_rtx_ssrc[0].ssrc; - InfoL << "收到重传包:" << seq; - } -#endif - - //解析并排序rtp - info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); -} - -void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { - auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize); - rtcp->ssrc = htons(0); - rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc); - InfoL << rtcp->RtcpHeader::dumpString(); - sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true); -} - /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { - if (!info.is_common_rtp) { - WarnL; - return; - } - if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { - //定期发送pli请求关键帧，方便非rtc等协议 - _pli_ticker.resetTime(); - sendRtcpPli(rtp->getSSRC()); - - //开启remb，则发送remb包调节比特率 - GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); - if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) { - sendRtcpRemb(rtp->getSSRC(), remb_bit_rate); - } - } - - if (_push_src) { - _push_src->onWrite(std::move(rtp), false); - } -} - static void setExtType(RtpExt &ext, uint8_t tp) {} static void setExtType(RtpExt &ext, RtpExtType tp) { ext.setType(tp); @@ -750,17 +686,98 @@ static void changeRtpExtId(const RtpHeader *header, const Type &map) { } setExtType(pr.second, it->first); setExtType(pr.second, it->second); -// DebugL << pr.second.dumpString(); pr.second.setExtId((uint8_t) it->second); } } +void WebRtcTransportImp::onRtp(const char *buf, size_t len) { + onRtp_l(buf, len, false); +} + +void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { + _bytes_usage += len; + _alive_ticker.resetTime(); + RtpHeader *rtp = (RtpHeader *) buf; + //根据接收到的rtp的pt信息，找到该流的信息 + auto it = _rtp_info_pt.find(rtp->pt); + if (it == _rtp_info_pt.end()) { + WarnL; + return; + } + auto &info = it->second; + if (info.is_common_rtp) { + //这是普通的rtp数据 + auto seq = ntohs(rtp->seq); +#if 0 + if (!rtx && info.media->type == TrackVideo && seq % 100 == 0) { + //此处模拟接受丢包 + DebugL << "recv dropped:" << seq; + return; + } +#endif + if (!rtx) { + //统计rtp接受情况，便于生成nack rtcp包 + info.nack_ctx.received(seq); + } + //解析并排序rtp + info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); + return; + } + + //这里是rtx重传包 + //https://datatracker.ietf.org/doc/html/rfc4588#section-4 + auto payload = rtp->getPayloadData(); + auto size = rtp->getPayloadSize(len); + if (size < 2) { + return; + } + //前两个字节是原始的rtp的seq + auto origin_seq = payload[0] << 8 | payload[1]; + InfoL << "received rtx rtp: " << origin_seq; + rtp->seq = htons(origin_seq); + rtp->ssrc = htonl(info.media->rtp_rtx_ssrc[0].ssrc); + rtp->pt = info.plan_apt->pt; + memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); + buf += 2; + len -= 2; + onRtp_l(buf, len, true); +} + +void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { + auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize); + rtcp->ssrc = htons(0); + rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc); + sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true); +} + +/////////////////////////////////////////////////////////////////// + void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type); //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } +void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { + if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { + //定期发送pli请求关键帧，方便非rtc等协议 + _pli_ticker.resetTime(); + sendRtcpPli(rtp->getSSRC()); + + //开启remb，则发送remb包调节比特率 + GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); + if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) { + sendRtcpRemb(rtp->getSSRC(), remb_bit_rate); + } + } + + if (_push_src) { + _push_src->onWrite(std::move(rtp), false); + } +} + +/////////////////////////////////////////////////////////////////// + void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){ auto info = _send_rtp_info[rtp->type]; if (!info) { @@ -773,13 +790,13 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r info->nack_list.push_back(rtp); #if 0 //此处模拟发送丢包 - if(rtp->getSeq() % 10 == 0){ - DebugL << "模拟发送丢包:" << rtp->getSeq(); + if(rtp->getSeq() % 100 == 0){ + DebugL << "send droped:" << rtp->getSeq(); return; } #endif } else { - WarnL << "rtp发送重传:" << rtp->getSeq(); + WarnL << "send rtx rtp:" << rtp->getSeq(); } sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index beebd16c..35046f30 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -298,6 +298,8 @@ protected: void onRtcConfigure(RtcConfigure &configure) const override; void onRtp(const char *buf, size_t len) override; + void onRtp_l(const char *buf, size_t len, bool rtx); + void onRtcp(const char *buf, size_t len) override; void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) override; void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) override; @@ -342,6 +344,7 @@ private: public: bool is_common_rtp; const RtcCodecPlan *plan; + const RtcCodecPlan *plan_apt; const RtcMedia *media; std::shared_ptr receiver; RtcpContext::Ptr rtcp_context_recv; From 12be56493e48710f6f5c4e647e7ced991016cab8 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 12 May 2021 15:40:40 +0800 Subject: [PATCH 182/218] =?UTF-8?q?=E4=BC=98=E5=8C=96=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 76 ++++++++++++++++++-------------------- webrtc/WebRtcTransport.h | 9 +++-- 2 files changed, 40 insertions(+), 45 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 36738ebb..50cf0c64 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -414,27 +414,26 @@ void WebRtcTransportImp::onStartWebRTC() { } auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); //获取offer端rtp的ssrc和pt相关信息 - auto &ref = _rtp_info_pt[plan.pt]; - ref.plan = &plan; - ref.media = m_with_ssrc; - ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; - if (ref.is_common_rtp) { + auto info = std::make_shared(); + _rtp_info_pt.emplace(plan.pt, info); + info->plan = &plan; + info->media = m_with_ssrc; + info->is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + if (info->is_common_rtp) { //rtp - _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; + _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = info; } else { //rtx auto apt = atoi(plan.getFmtp("apt").data()); - ref.plan_apt = m_with_ssrc->getPlan(apt); + info->plan_apt = m.getPlan(apt); } - ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); - ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); - ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) mutable{ - onSortedRtp(ref, std::move(rtp)); - }, [&ref, this](const RtpPacket::Ptr &rtp) mutable { - onBeforeSortedRtp(ref, rtp); + info->rtcp_context_recv = std::make_shared(info->plan->sample_rate, true); + info->rtcp_context_send = std::make_shared(info->plan->sample_rate, false); + info->receiver = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { + onSortedRtp(*info, std::move(rtp)); }); - ref.nack_ctx.setOnNack([&ref, this](const FCI_NACK &nack) mutable{ - onNack(ref, nack); + info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable { + onNack(*info, nack); }); } if (m.type != TrackApplication) { @@ -475,7 +474,7 @@ void WebRtcTransportImp::onStartWebRTC() { auto it = _rtp_info_pt.find(m.plan[0].pt); CHECK(it != _rtp_info_pt.end()); //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc - _send_rtp_info[m.type] = &it->second; + _send_rtp_info[m.type] = it->second; } } } @@ -564,9 +563,8 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ class RtpReceiverImp : public RtpReceiver { public: - RtpReceiverImp( function cb, function cb_before = nullptr){ + RtpReceiverImp( function cb){ _on_sort = std::move(cb); - _on_before_sort = std::move(cb_before); } ~RtpReceiverImp() override = default; @@ -580,15 +578,8 @@ protected: _on_sort(std::move(rtp)); } - void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override { - if (_on_before_sort) { - _on_before_sort(rtp); - } - } - private: function _on_sort; - function _on_before_sort; }; void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { @@ -695,8 +686,11 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { } void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { - _bytes_usage += len; - _alive_ticker.resetTime(); + if (!rtx) { + _bytes_usage += len; + _alive_ticker.resetTime(); + } + RtpHeader *rtp = (RtpHeader *) buf; //根据接收到的rtp的pt信息，找到该流的信息 auto it = _rtp_info_pt.find(rtp->pt); @@ -705,11 +699,11 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { return; } auto &info = it->second; - if (info.is_common_rtp) { + if (info->is_common_rtp) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); #if 0 - if (!rtx && info.media->type == TrackVideo && seq % 100 == 0) { + if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) { //此处模拟接受丢包 DebugL << "recv dropped:" << seq; return; @@ -717,10 +711,16 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { #endif if (!rtx) { //统计rtp接受情况，便于生成nack rtcp包 - info.nack_ctx.received(seq); + info->nack_ctx.received(seq); + //修改ext id至统一 + changeRtpExtId(rtp, _rtp_ext_id_to_type); + //时间戳转换成毫秒 + auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan->sample_rate; + //统计rtp收到的情况，好做rr汇报 + info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } //解析并排序rtp - info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); + info->receiver->inputRtp(info->media->type, info->plan->sample_rate, (uint8_t *) buf, len); return; } @@ -735,8 +735,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { auto origin_seq = payload[0] << 8 | payload[1]; InfoL << "received rtx rtp: " << origin_seq; rtp->seq = htons(origin_seq); - rtp->ssrc = htonl(info.media->rtp_rtx_ssrc[0].ssrc); - rtp->pt = info.plan_apt->pt; + rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc); + rtp->pt = info->plan_apt->pt; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; len -= 2; @@ -752,12 +752,6 @@ void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { - changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type); - //统计rtp收到的情况，好做rr汇报 - info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); -} - void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { //定期发送pli请求关键帧，方便非rtc等协议 @@ -779,7 +773,7 @@ void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { /////////////////////////////////////////////////////////////////// void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){ - auto info = _send_rtp_info[rtp->type]; + auto &info = _send_rtp_info[rtp->type]; if (!info) { //忽略，对方不支持该编码类型 return; @@ -798,7 +792,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r } else { WarnL << "send rtx rtp:" << rtp->getSeq(); } - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info); + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info.get()); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 35046f30..8fc1ddca 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -342,6 +342,8 @@ private: class RtpPayloadInfo { public: + using Ptr = std::shared_ptr; + bool is_common_rtp; const RtcCodecPlan *plan; const RtcCodecPlan *plan_apt; @@ -354,7 +356,6 @@ private: }; void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); - void onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); void onNack(RtpPayloadInfo &info, const FCI_NACK &nack); private: @@ -371,7 +372,7 @@ private: //pli rtcp计时器 Ticker _pli_ticker; //记录协商的发送rtp的pt和ssrc - RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr}; + RtpPayloadInfo::Ptr _send_rtp_info[2]; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 @@ -381,9 +382,9 @@ private: //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 - unordered_map _rtp_info_pt; + unordered_map _rtp_info_pt; //根据rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id From 8fdfc14f6feb2d9b3837a1281974155f450149ea Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 12 May 2021 20:51:51 +0800 Subject: [PATCH 183/218] =?UTF-8?q?=E6=95=B4=E7=90=86=20nack/rtx/ssrc?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 131 ++++++++++++++++++++++--------------- webrtc/WebRtcTransport.h | 12 ++-- 2 files changed, 85 insertions(+), 58 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 50cf0c64..b599fd5a 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -271,7 +271,8 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx) { if (_srtp_session_send) { - CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); + //预留rtx加入的两个字节 + CHECK(len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf)); memcpy(_srtp_buf, buf, len); onBeforeEncryptRtp((char *) _srtp_buf, len, ctx); if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) { @@ -395,40 +396,40 @@ bool WebRtcTransportImp::canRecvRtp() const{ } const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{ - auto &offer = getSdp(SdpType::offer); + auto &offer = getSdp(SdpType::answer); if (offer.haveSSRC()) { return offer; } - auto &answer = getSdp(SdpType::answer); + auto &answer = getSdp(SdpType::offer); CHECK(answer.haveSSRC()); return answer; } void WebRtcTransportImp::onStartWebRTC() { //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 - for (auto &m : getSdp(SdpType::offer).media) { - for (auto &plan : m.plan) { - auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); - if (!hit_pan) { - continue; - } - auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); + for (auto &m_answer : getSdp(SdpType::answer).media) { + auto m_with_ssrc = getSdpWithSSRC().getMedia(m_answer.type); + for (auto &plan_answer : m_answer.plan) { //获取offer端rtp的ssrc和pt相关信息 auto info = std::make_shared(); - _rtp_info_pt.emplace(plan.pt, info); - info->plan = &plan; + _rtp_info_pt.emplace(plan_answer.pt, info); info->media = m_with_ssrc; - info->is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + info->is_common_rtp = getCodecId(plan_answer.codec) != CodecInvalid; if (info->is_common_rtp) { //rtp - _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = info; + _rtp_info_ssrc[info->media->rtp_rtx_ssrc[0].ssrc] = info; + info->plan_rtp = &plan_answer; + info->plan_rtx = m_answer.getRelatedRtxPlan(plan_answer.pt); } else { //rtx - auto apt = atoi(plan.getFmtp("apt").data()); - info->plan_apt = m.getPlan(apt); + if (info->media->rtp_rtx_ssrc.size() > 1) { + _rtp_info_ssrc[info->media->rtp_rtx_ssrc[1].ssrc] = info; + } + info->plan_rtp = m_answer.getPlan(atoi(plan_answer.getFmtp("apt").data())); + info->plan_rtx = &plan_answer; } - info->rtcp_context_recv = std::make_shared(info->plan->sample_rate, true); - info->rtcp_context_send = std::make_shared(info->plan->sample_rate, false); + info->rtcp_context_recv = std::make_shared(info->plan_rtp->sample_rate, true); + info->rtcp_context_send = std::make_shared(info->plan_rtp->sample_rate, false); info->receiver = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { onSortedRtp(*info, std::move(rtp)); }); @@ -436,9 +437,9 @@ void WebRtcTransportImp::onStartWebRTC() { onNack(*info, nack); }); } - if (m.type != TrackApplication) { + if (m_answer.type != TrackApplication) { //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id - for (auto &ext : m.extmap) { + for (auto &ext : m_answer.extmap) { auto ext_type = RtpExt::getExtType(ext.ext); _rtp_ext_id_to_type.emplace(ext.id, ext_type); _rtp_ext_type_to_id.emplace(ext_type, ext.id); @@ -499,8 +500,8 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ sdp.origin.address = m.addr.address; } - if (!canSendRtp() || getSdp(SdpType::offer).haveSSRC()) { - //offer sdp未包含ssrc相关信息，那么我们才在answer sdp中回复ssrc相关信息 + if (!canSendRtp()) { + //设置我们发送的rtp的ssrc return; } @@ -596,7 +597,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); } else { - WarnL << "未识别的sr rtcp包:" << (uint32_t)sr->ssrc; + WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); } break; } @@ -604,12 +605,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { _alive_ticker.resetTime(); //对方汇报rtp接收情况 RtcpRR *rr = (RtcpRR *) rtcp; - auto it = _rtp_info_ssrc.find(rr->ssrc); - if (it != _rtp_info_ssrc.end()) { - auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); - sendRtcpPacket(sr->data(), sr->size(), true); - } else { - WarnL << "未识别的rr rtcp包:" << (uint32_t)rr->ssrc; + for (auto item : rr->getItemList()) { + auto it = _rtp_info_ssrc.find(item->ssrc); + if (it != _rtp_info_ssrc.end()) { + auto sr = it->second->rtcp_context_send->createRtcpSR(item->ssrc); + sendRtcpPacket(sr->data(), sr->size(), true); + } else { + WarnL << "未识别的rr rtcp包:" << rtcp->dumpString(); + } } break; } @@ -619,10 +622,13 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { for (auto ssrc : bye->getSSRC()) { auto it = _rtp_info_ssrc.find(*ssrc); if (it == _rtp_info_ssrc.end()) { - WarnL << "未识别的bye rtcp包:" << *ssrc; + WarnL << "未识别的bye rtcp包:" << rtcp->dumpString(); continue; } - _rtp_info_pt.erase(it->second->plan->pt); + _rtp_info_pt.erase(it->second->plan_rtp->pt); + if (it->second->plan_rtx) { + _rtp_info_pt.erase(it->second->plan_rtx->pt); + } _rtp_info_ssrc.erase(it); } onShutdown(SockException(Err_eof, "rtcp bye message received")); @@ -630,18 +636,18 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - RtcpFB *fb = (RtcpFB *) rtcp; - auto it = _rtp_info_ssrc.find(fb->ssrc); - if (it == _rtp_info_ssrc.end()) { - WarnL << "未识别的 rtcp包:" << rtcp->dumpString(); - return; - } if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) { break; } //RTPFB switch ((RTPFBType) rtcp->report_count) { case RTPFBType::RTCP_RTPFB_NACK : { + RtcpFB *fb = (RtcpFB *) rtcp; + auto it = _rtp_info_ssrc.find(fb->ssrc_media); + if (it == _rtp_info_ssrc.end()) { + WarnL << "未识别的 rtcp包:" << rtcp->dumpString(); + return; + } auto &fci = fb->getFci(); it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { //rtp重传 @@ -702,7 +708,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { if (info->is_common_rtp) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); -#if 0 +#if 1 if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) { //此处模拟接受丢包 DebugL << "recv dropped:" << seq; @@ -715,12 +721,12 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { //修改ext id至统一 changeRtpExtId(rtp, _rtp_ext_id_to_type); //时间戳转换成毫秒 - auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan->sample_rate; + auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate; //统计rtp收到的情况，好做rr汇报 info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } //解析并排序rtp - info->receiver->inputRtp(info->media->type, info->plan->sample_rate, (uint8_t *) buf, len); + info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; } @@ -736,7 +742,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { InfoL << "received rtx rtp: " << origin_seq; rtp->seq = htons(origin_seq); rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc); - rtp->pt = info->plan_apt->pt; + rtp->pt = info->plan_rtp->pt; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; len -= 2; @@ -782,31 +788,52 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r //统计rtp发送情况，好做sr汇报 info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); info->nack_list.push_back(rtp); -#if 0 +#if 1 //此处模拟发送丢包 if(rtp->getSeq() % 100 == 0){ - DebugL << "send droped:" << rtp->getSeq(); + DebugL << "send dropped:" << rtp->getSeq(); return; } #endif } else { WarnL << "send rtx rtp:" << rtp->getSeq(); } - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info.get()); + pair ctx{rtx, info.get()}; + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; } -void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) { - RtpPayloadInfo *info = reinterpret_cast(ctx); - auto header = (RtpHeader *)buf; - //修改目标pt和ssrc - header->pt = info->plan->pt; - header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc); +void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) { + auto pr = (pair *) ctx; + auto header = (RtpHeader *) buf; changeRtpExtId(header, _rtp_ext_type_to_id); -} -void WebRtcTransportImp::onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) { + if (!pr->first || !pr->second->plan_rtx) { + //普通的rtp,或者不支持rtx, 修改目标pt和ssrc + header->pt = pr->second->plan_rtp->pt; + header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[0].ssrc); + } else { + //重传的rtp, rtx + header->pt = pr->second->plan_rtx->pt; + if (pr->second->media->rtp_rtx_ssrc.size() > 1) { + //有rtx单独的ssrc + header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[1].ssrc); + } + auto origin_seq = ntohs(header->seq); + //seq跟原来的不一样 + header->seq = htons(origin_seq + 100); + auto payload = header->getPayloadData(); + auto payload_size = header->getPayloadSize(len); + if (payload_size) { + //rtp负载后移两个字节，这两个字节用于存放osn + //https://datatracker.ietf.org/doc/html/rfc4588#section-4 + memmove(payload + 2, payload, payload_size); + } + payload[0] = origin_seq >> 8; + payload[1] = origin_seq & 0xFF; + len += 2; + } } void WebRtcTransportImp::onShutdown(const SockException &ex){ diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 8fc1ddca..f9ca0d31 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -101,8 +101,8 @@ protected: virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; virtual void onShutdown(const SockException &ex) = 0; - virtual void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) = 0; - virtual void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) = 0; + virtual void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) = 0; + virtual void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) = 0; protected: const RtcSession& getSdp(SdpType type) const; @@ -301,8 +301,8 @@ protected: void onRtp_l(const char *buf, size_t len, bool rtx); void onRtcp(const char *buf, size_t len) override; - void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) override; - void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) override; + void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) override; + void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) override {}; void onShutdown(const SockException &ex) override; @@ -345,8 +345,8 @@ private: using Ptr = std::shared_ptr; bool is_common_rtp; - const RtcCodecPlan *plan; - const RtcCodecPlan *plan_apt; + const RtcCodecPlan *plan_rtp; + const RtcCodecPlan *plan_rtx; const RtcMedia *media; std::shared_ptr receiver; RtcpContext::Ptr rtcp_context_recv; From 5f13335166f618fa5294a653c47b206fb8cc7505 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 12 May 2021 20:53:37 +0800 Subject: [PATCH 184/218] =?UTF-8?q?=E5=85=B3=E9=97=AD=E4=B8=A2=E5=8C=85?= =?UTF-8?q?=E6=B5=8B=E8=AF=95=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index b599fd5a..9c4972e2 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -708,7 +708,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { if (info->is_common_rtp) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); -#if 1 +#if 0 if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) { //此处模拟接受丢包 DebugL << "recv dropped:" << seq; @@ -788,7 +788,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r //统计rtp发送情况，好做sr汇报 info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); info->nack_list.push_back(rtp); -#if 1 +#if 0 //此处模拟发送丢包 if(rtp->getSeq() % 100 == 0){ DebugL << "send dropped:" << rtp->getSeq(); From 0e968a2d5efc2a6b93217eabeee0170a94dd0778 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Wed, 12 May 2021 23:48:13 +0800 Subject: [PATCH 185/218] =?UTF-8?q?=E6=9B=B4=E6=96=B0ZLToolKit?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- 3rdpart/ZLToolKit | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/3rdpart/ZLToolKit b/3rdpart/ZLToolKit index d5de1d4f..b678e474 160000 --- a/3rdpart/ZLToolKit +++ b/3rdpart/ZLToolKit @@ -1 +1 @@ -Subproject commit d5de1d4f0c86139fe8f0b6d956e45dee3b151783 +Subproject commit b678e474f8ba8c9f3e6efd9cec6e4c517e80209d From a7cd4d0719c2e69b6679a6ce51290d5d87894ea3 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Thu, 13 May 2021 01:29:55 +0800 Subject: [PATCH 186/218] =?UTF-8?q?rtp=20ext=E4=B8=8D=E5=81=9A=E5=86=85?= =?UTF-8?q?=E5=AD=98=E6=8B=B7=E8=B4=9D?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 37 ++++++++++++++++++++++++++++--------- webrtc/RtpExt.h | 14 ++++++++++---- 2 files changed, 38 insertions(+), 13 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index 62cdc830..a3e2ec3a 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -147,9 +147,28 @@ void appendExt(map &ret, uint8_t *ptr, const uint8_t *end) { } } -RtpExt::RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size) : std::string(str, size) { - _ptr = ptr; +RtpExt::RtpExt(void *ext, bool one_byte_ext, const char *str, size_t size) { + _ext = ext; _one_byte_ext = one_byte_ext; + _data = str; + _size = size; +} + +const char *RtpExt::data() const { + return _data; +} + +size_t RtpExt::size() const { + return _size; +} + +const char& RtpExt::operator[](size_t pos) const{ + CHECK(pos < _size); + return _data[pos]; +} + +RtpExt::operator std::string() const{ + return string(_data, _size); } map RtpExt::getExtValue(const RtpHeader *header) { @@ -364,7 +383,7 @@ uint16_t RtpExt::getTransportCCSeq() const { // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // | ID | len | SDES Item text value ... | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -const string &RtpExt::getSdesMid() const { +string RtpExt::getSdesMid() const { CHECK(_type == RtpExtType::sdes_mid && size() >= 1); return *this; } @@ -515,23 +534,23 @@ uint8_t RtpExt::getFramemarkingTID() const { } void RtpExt::setExtId(uint8_t ext_id) { - assert(ext_id > (int) RtpExtType::padding && ext_id <= (int) RtpExtType::reserved && _ptr); + assert(ext_id > (int) RtpExtType::padding && ext_id <= (int) RtpExtType::reserved && _ext); if (_one_byte_ext) { - auto ptr = reinterpret_cast(_ptr); + auto ptr = reinterpret_cast(_ext); ptr->setId(ext_id); } else { - auto ptr = reinterpret_cast(_ptr); + auto ptr = reinterpret_cast(_ext); ptr->setId(ext_id); } } void RtpExt::clearExt(){ - assert(_ptr); + assert(_ext); if (_one_byte_ext) { - auto ptr = reinterpret_cast(_ptr); + auto ptr = reinterpret_cast(_ext); memset(ptr, (int) RtpExtType::padding, RtpExtOneByte::kMinSize + ptr->getSize()); } else { - auto ptr = reinterpret_cast(_ptr); + auto ptr = reinterpret_cast(_ext); memset(ptr, (int) RtpExtType::padding, RtpExtTwoByte::kMinSize + ptr->getSize()); } } diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 3d916def..9c59129c 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -45,7 +45,8 @@ enum class RtpExtType : uint8_t { class RtcMedia; -class RtpExt : public std::string { +//使用次对象的方法前需保证RtpHeader内存未释放 +class RtpExt { public: template friend void appendExt(map &ret, uint8_t *ptr, const uint8_t *end); @@ -63,7 +64,7 @@ public: uint8_t getAudioLevel(bool *vad) const; uint32_t getAbsSendTime() const; uint16_t getTransportCCSeq() const; - const string& getSdesMid() const; + string getSdesMid() const; string getRtpStreamId() const; string getRepairedRtpStreamId() const; @@ -88,15 +89,20 @@ public: uint8_t getFramemarkingTID() const; - //危险函数，必须保证关联的RtpHeader对象有效 void setExtId(uint8_t ext_id); void clearExt(); private: RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size); + const char *data() const; + size_t size() const; + const char& operator[](size_t pos) const; + operator std::string() const; private: - void *_ptr = nullptr; + void *_ext = nullptr; + const char *_data; + size_t _size; bool _one_byte_ext = true; RtpExtType _type = RtpExtType::padding; }; From d395d23eeb066598d50bae0b878ae5a34175b5f6 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 15 May 2021 10:05:47 +0800 Subject: [PATCH 187/218] =?UTF-8?q?=E7=94=B1=E4=BA=8E=E5=9C=A8RtspSession?= =?UTF-8?q?=E7=B1=BB=E7=A1=AE=E4=BF=9D=E4=BA=86=E5=87=BD=E6=95=B0=E6=89=A7?= =?UTF-8?q?=E8=A1=8C=E9=A1=BA=E5=BA=8F=EF=BC=8C=E6=89=80=E4=BB=A5=E5=9B=9E?= =?UTF-8?q?=E6=BB=9A=E5=A4=9A=E4=BD=99=E4=BB=A3=E7=A0=81=E4=B9=9F=E8=83=BD?= =?UTF-8?q?=E7=A1=AE=E4=BF=9D=E5=8D=95track=E6=B5=81=E8=BF=85=E9=80=9F?= =?UTF-8?q?=E6=B3=A8=E5=86=8C?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtsp/RtspMediaSourceImp.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/src/Rtsp/RtspMediaSourceImp.h b/src/Rtsp/RtspMediaSourceImp.h index 53e9d4d0..37731f1e 100644 --- a/src/Rtsp/RtspMediaSourceImp.h +++ b/src/Rtsp/RtspMediaSourceImp.h @@ -89,9 +89,6 @@ public: _muxer->addTrack(track); track->addDelegate(_muxer); } - if (_all_demuxer_track_ready) { - this->addTrackCompleted(); - } } /** @@ -111,7 +108,6 @@ public: if (_muxer) { _muxer->addTrackCompleted(); } - _all_demuxer_track_ready = true; } void resetTracks() override { @@ -145,7 +141,6 @@ private: RtspDemuxer::Ptr _demuxer; MultiMediaSourceMuxer::Ptr _muxer; bool _all_track_ready = false; - bool _all_demuxer_track_ready = false; }; } /* namespace mediakit */ From f6eb84b4130f159f8d30ecda4e877612d466d093 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 12:28:50 +0800 Subject: [PATCH 188/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0ssrc=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E6=96=B9=E6=B3=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 14 ++++++++++++++ webrtc/Sdp.h | 2 ++ 2 files changed, 16 insertions(+) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 03cb5e9a..299655e4 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1284,6 +1284,20 @@ const RtcCodecPlan *RtcMedia::getRelatedRtxPlan(uint8_t pt) const{ return nullptr; } +uint32_t RtcMedia::getRtpSSRC() const { + if (rtp_rtx_ssrc.size()) { + return rtp_rtx_ssrc[0].ssrc; + } + return 0; +} + +uint32_t RtcMedia::getRtxSSRC() const { + if (rtp_rtx_ssrc.size() > 1) { + return rtp_rtx_ssrc[1].ssrc; + } + return 0; +} + void RtcMedia::checkValid() const{ CHECK(type != TrackInvalid); CHECK(!mid.empty()); diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h index 1ddfce94..6ed2710a 100644 --- a/webrtc/Sdp.h +++ b/webrtc/Sdp.h @@ -645,6 +645,8 @@ public: const RtcCodecPlan *getPlan(uint8_t pt) const; const RtcCodecPlan *getPlan(const char *codec) const; const RtcCodecPlan *getRelatedRtxPlan(uint8_t pt) const; + uint32_t getRtpSSRC() const; + uint32_t getRtxSSRC() const; }; class RtcSession{ From d4ff84e447f8fcacf1d75671479d06c4c81c2936 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 16:12:10 +0800 Subject: [PATCH 189/218] =?UTF-8?q?=E5=AE=8C=E5=96=84ssrc=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E5=A4=84=E7=90=86?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 134 +++++++++++++++++++------------------ webrtc/WebRtcTransport.h | 45 ++++--------- 2 files changed, 82 insertions(+), 97 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9c4972e2..3ff65047 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -395,51 +395,45 @@ bool WebRtcTransportImp::canRecvRtp() const{ return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly); } -const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{ - auto &offer = getSdp(SdpType::answer); - if (offer.haveSSRC()) { - return offer; - } - auto &answer = getSdp(SdpType::offer); - CHECK(answer.haveSSRC()); - return answer; -} - void WebRtcTransportImp::onStartWebRTC() { //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 for (auto &m_answer : getSdp(SdpType::answer).media) { - auto m_with_ssrc = getSdpWithSSRC().getMedia(m_answer.type); - for (auto &plan_answer : m_answer.plan) { - //获取offer端rtp的ssrc和pt相关信息 - auto info = std::make_shared(); - _rtp_info_pt.emplace(plan_answer.pt, info); - info->media = m_with_ssrc; - info->is_common_rtp = getCodecId(plan_answer.codec) != CodecInvalid; - if (info->is_common_rtp) { - //rtp - _rtp_info_ssrc[info->media->rtp_rtx_ssrc[0].ssrc] = info; - info->plan_rtp = &plan_answer; - info->plan_rtx = m_answer.getRelatedRtxPlan(plan_answer.pt); - } else { - //rtx - if (info->media->rtp_rtx_ssrc.size() > 1) { - _rtp_info_ssrc[info->media->rtp_rtx_ssrc[1].ssrc] = info; - } - info->plan_rtp = m_answer.getPlan(atoi(plan_answer.getFmtp("apt").data())); - info->plan_rtx = &plan_answer; - } - info->rtcp_context_recv = std::make_shared(info->plan_rtp->sample_rate, true); - info->rtcp_context_send = std::make_shared(info->plan_rtp->sample_rate, false); - info->receiver = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { - onSortedRtp(*info, std::move(rtp)); - }); - info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable { - onNack(*info, nack); - }); + auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type); + auto info = std::make_shared(); + + info->media = &m_answer; + info->answer_ssrc_rtp = m_answer.getRtpSSRC(); + info->answer_ssrc_rtx = m_answer.getRtxSSRC(); + info->offer_ssrc_rtp = m_offer->getRtpSSRC(); + info->offer_ssrc_rtx = m_offer->getRtxSSRC(); + info->plan_rtp = &m_answer.plan[0];; + info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt); + info->rtcp_context_recv = std::make_shared(info->plan_rtp->sample_rate, true); + info->rtcp_context_send = std::make_shared(info->plan_rtp->sample_rate, false); + info->receiver = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { + onSortedRtp(*info, std::move(rtp)); + }); + info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable { + onSendNack(*info, nack); + }); + + //send ssrc --> RtpPayloadInfo + _rtp_info_ssrc[info->answer_ssrc_rtp] = std::make_pair(false, info); + _rtp_info_ssrc[info->answer_ssrc_rtx] = std::make_pair(true, info); + + //recv ssrc --> RtpPayloadInfo + _rtp_info_ssrc[info->offer_ssrc_rtp] = std::make_pair(false, info);; + _rtp_info_ssrc[info->offer_ssrc_rtx] = std::make_pair(true, info);; + + //rtp pt --> RtpPayloadInfo + _rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info)); + if (info->plan_rtx) { + //rtx pt --> RtpPayloadInfo + _rtp_info_pt.emplace(info->plan_rtx->pt, std::make_pair(true, info)); } - if (m_answer.type != TrackApplication) { + if (m_offer->type != TrackApplication) { //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id - for (auto &ext : m_answer.extmap) { + for (auto &ext : m_offer->extmap) { auto ext_type = RtpExt::getExtType(ext.ext); _rtp_ext_id_to_type.emplace(ext.id, ext_type); _rtp_ext_type_to_id.emplace(ext_type, ext.id); @@ -475,7 +469,7 @@ void WebRtcTransportImp::onStartWebRTC() { auto it = _rtp_info_pt.find(m.plan[0].pt); CHECK(it != _rtp_info_pt.end()); //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc - _send_rtp_info[m.type] = it->second; + _send_rtp_info[m.type] = it->second.second; } } } @@ -593,9 +587,13 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { RtcpSR *sr = (RtcpSR *) rtcp; auto it = _rtp_info_ssrc.find(sr->ssrc); if (it != _rtp_info_ssrc.end()) { - it->second->rtcp_context_recv->onRtcp(sr); - auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); - sendRtcpPacket(rr->data(), rr->size(), true); + auto rtx = it->second.first; + if (!rtx) { + auto &info = it->second.second; + info->rtcp_context_recv->onRtcp(sr); + auto rr = info->rtcp_context_recv->createRtcpRR(info->answer_ssrc_rtp, info->offer_ssrc_rtp); + sendRtcpPacket(rr->data(), rr->size(), true); + } } else { WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); } @@ -608,8 +606,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { for (auto item : rr->getItemList()) { auto it = _rtp_info_ssrc.find(item->ssrc); if (it != _rtp_info_ssrc.end()) { - auto sr = it->second->rtcp_context_send->createRtcpSR(item->ssrc); - sendRtcpPacket(sr->data(), sr->size(), true); + auto rtx = it->second.first; + if (!rtx) { + auto &info = it->second.second; + auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp); + sendRtcpPacket(sr->data(), sr->size(), true); + } } else { WarnL << "未识别的rr rtcp包:" << rtcp->dumpString(); } @@ -625,10 +627,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { WarnL << "未识别的bye rtcp包:" << rtcp->dumpString(); continue; } - _rtp_info_pt.erase(it->second->plan_rtp->pt); - if (it->second->plan_rtx) { - _rtp_info_pt.erase(it->second->plan_rtx->pt); - } _rtp_info_ssrc.erase(it); } onShutdown(SockException(Err_eof, "rtcp bye message received")); @@ -648,11 +646,15 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { WarnL << "未识别的 rtcp包:" << rtcp->dumpString(); return; } - auto &fci = fb->getFci(); - it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { - //rtp重传 - onSendRtp(rtp, true, true); - }); + auto rtx = it->second.first; + if (!rtx) { + auto &info = it->second.second; + auto &fci = fb->getFci(); + info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { + //rtp重传 + onSendRtp(rtp, true, true); + }); + } break; } default: break; @@ -704,8 +706,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { WarnL; return; } - auto &info = it->second; - if (info->is_common_rtp) { + auto &info = it->second.second; + if (!it->second.first) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); #if 0 @@ -741,7 +743,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { auto origin_seq = payload[0] << 8 | payload[1]; InfoL << "received rtx rtp: " << origin_seq; rtp->seq = htons(origin_seq); - rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc); + rtp->ssrc = htonl(info->offer_ssrc_rtp); rtp->pt = info->plan_rtp->pt; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; @@ -749,10 +751,10 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { onRtp_l(buf, len, true); } -void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { +void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack) { auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize); - rtcp->ssrc = htons(0); - rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc); + rtcp->ssrc = htons(info.answer_ssrc_rtp); + rtcp->ssrc_media = htonl(info.offer_ssrc_rtp); sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true); } @@ -790,7 +792,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r info->nack_list.push_back(rtp); #if 0 //此处模拟发送丢包 - if(rtp->getSeq() % 100 == 0){ + if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) { DebugL << "send dropped:" << rtp->getSeq(); return; } @@ -811,13 +813,13 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void * if (!pr->first || !pr->second->plan_rtx) { //普通的rtp,或者不支持rtx, 修改目标pt和ssrc header->pt = pr->second->plan_rtp->pt; - header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[0].ssrc); + header->ssrc = htonl(pr->second->answer_ssrc_rtp); } else { //重传的rtp, rtx header->pt = pr->second->plan_rtx->pt; - if (pr->second->media->rtp_rtx_ssrc.size() > 1) { - //有rtx单独的ssrc - header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[1].ssrc); + if (pr->second->answer_ssrc_rtx) { + //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc + header->ssrc = htonl(pr->second->answer_ssrc_rtx); } auto origin_seq = ntohs(header->seq); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index f9ca0d31..21b09372 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -338,25 +338,26 @@ private: SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; - const RtcSession& getSdpWithSSRC() const; class RtpPayloadInfo { public: using Ptr = std::shared_ptr; - - bool is_common_rtp; const RtcCodecPlan *plan_rtp; const RtcCodecPlan *plan_rtx; + uint32_t offer_ssrc_rtp = 0; + uint32_t offer_ssrc_rtx = 0; + uint32_t answer_ssrc_rtp = 0; + uint32_t answer_ssrc_rtx = 0; const RtcMedia *media; - std::shared_ptr receiver; - RtcpContext::Ptr rtcp_context_recv; - RtcpContext::Ptr rtcp_context_send; NackList nack_list; NackContext nack_ctx; + RtcpContext::Ptr rtcp_context_recv; + RtcpContext::Ptr rtcp_context_send; + std::shared_ptr receiver; }; void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); - void onNack(RtpPayloadInfo &info, const FCI_NACK &nack); + void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack); private: //用掉的总流量 @@ -371,8 +372,6 @@ private: Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; - //记录协商的发送rtp的pt和ssrc - RtpPayloadInfo::Ptr _send_rtp_info[2]; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 @@ -381,30 +380,14 @@ private: RtspMediaSource::Ptr _play_src; //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; - //根据rtp的pt获取相关信息 - unordered_map _rtp_info_pt; + //根据发送rtp的track类型获取相关信息 + RtpPayloadInfo::Ptr _send_rtp_info[2]; + //根据接收rtp的pt获取相关信息 + unordered_map > _rtp_info_pt; //根据rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map > _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id unordered_map _rtp_ext_id_to_type; -}; - - - - - - - - - - - - - - - - - - +}; \ No newline at end of file From 7cdd5ed9dfb123b2b4db7bbf6ed539f435a22b3e Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 16:59:12 +0800 Subject: [PATCH 190/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Drtx=E5=8C=85=E6=9C=AA?= =?UTF-8?q?=E4=BF=AE=E6=94=B9rtp=20ext=20id=E7=9A=84=E9=97=AE=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 8 ++++---- webrtc/WebRtcTransport.h | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 3ff65047..9a911cde 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -269,7 +269,7 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * } } -void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx) { +void WebRtcTransport::sendRtpPacket(const char *buf, size_t len, bool flush, void *ctx) { if (_srtp_session_send) { //预留rtx加入的两个字节 CHECK(len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf)); @@ -281,7 +281,7 @@ void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx } } -void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx){ +void WebRtcTransport::sendRtcpPacket(const char *buf, size_t len, bool flush, void *ctx){ if (_srtp_session_send) { CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf)); memcpy(_srtp_buf, buf, len); @@ -720,13 +720,13 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { if (!rtx) { //统计rtp接受情况，便于生成nack rtcp包 info->nack_ctx.received(seq); - //修改ext id至统一 - changeRtpExtId(rtp, _rtp_ext_id_to_type); //时间戳转换成毫秒 auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate; //统计rtp收到的情况，好做rr汇报 info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } + //修改ext id至统一 + changeRtpExtId(rtp, _rtp_ext_id_to_type); //解析并排序rtp info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 21b09372..512db38b 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -63,8 +63,8 @@ public: * @param flush 是否flush socket * @param ctx 用户指针 */ - void sendRtpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr); - void sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr); + void sendRtpPacket(const char *buf, size_t len, bool flush, void *ctx = nullptr); + void sendRtcpPacket(const char *buf, size_t len, bool flush, void *ctx = nullptr); const EventPoller::Ptr& getPoller() const; From bda378c340f52727d5b18a9c8575376f3c81044a Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 18:06:34 +0800 Subject: [PATCH 191/218] =?UTF-8?q?=E5=AE=8C=E5=96=84=E5=8F=91=E9=80=81rtx?= =?UTF-8?q?=20rtp=E5=A4=84=E7=90=86?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 5 ++++- webrtc/WebRtcTransport.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9a911cde..d898c31a 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -820,11 +820,14 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void * if (pr->second->answer_ssrc_rtx) { //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc header->ssrc = htonl(pr->second->answer_ssrc_rtx); + } else { + //未单独指定rtx的ssrc，那么使用rtp的ssrc + header->ssrc = htonl(pr->second->answer_ssrc_rtp); } auto origin_seq = ntohs(header->seq); //seq跟原来的不一样 - header->seq = htons(origin_seq + 100); + header->seq = htons(_rtx_seq[pr->second->media->type]++); auto payload = header->getPayloadData(); auto payload_size = header->getPayloadSize(len); if (payload_size) { diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 512db38b..11544eb6 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -360,6 +360,7 @@ private: void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack); private: + uint16_t _rtx_seq[2] = {0, 0}; //用掉的总流量 uint64_t _bytes_usage = 0; //媒体相关元数据 From a84bcec4aa3a960f8bc4a7197a27fae2d1f803fa Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 20:56:03 +0800 Subject: [PATCH 192/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtp=20ext=E5=A4=84?= =?UTF-8?q?=E7=90=86=E7=9B=B8=E5=85=B3=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/Sdp.cpp | 6 ++++-- webrtc/WebRtcTransport.cpp | 41 ++++++++++++++++++++++---------------- webrtc/WebRtcTransport.h | 1 + 3 files changed, 29 insertions(+), 19 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 299655e4..5bd993b3 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1412,7 +1412,8 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ RtpExtType::csrc_audio_level, RtpExtType::abs_send_time, RtpExtType::transport_cc, - RtpExtType::sdes_mid, + //rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放 + //RtpExtType::sdes_mid, RtpExtType::sdes_rtp_stream_id, RtpExtType::sdes_repaired_rtp_stream_id }; @@ -1425,7 +1426,8 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ extmap = { RtpExtType::abs_send_time, RtpExtType::transport_cc, - RtpExtType::sdes_mid, + //rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放 + //RtpExtType::sdes_mid, RtpExtType::sdes_rtp_stream_id, RtpExtType::sdes_repaired_rtp_stream_id, RtpExtType::video_timing, diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index d898c31a..5d7b0cdc 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -668,24 +668,30 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { /////////////////////////////////////////////////////////////////// -static void setExtType(RtpExt &ext, uint8_t tp) {} -static void setExtType(RtpExt &ext, RtpExtType tp) { - ext.setType(tp); -} - -template -static void changeRtpExtId(const RtpHeader *header, const Type &map) { +void WebRtcTransportImp::changeRtpExtId(const RtpHeader *header, bool is_recv, bool is_rtx) const{ auto ext_map = RtpExt::getExtValue(header); for (auto &pr : ext_map) { - auto it = map.find((typename Type::key_type) (pr.first)); - if (it == map.end()) { - WarnL << "未处理的rtp ext, 类型不识别:" << (int) pr.first; - pr.second.clearExt(); - continue; + if (is_recv) { + auto it = _rtp_ext_id_to_type.find(pr.first); + if (it == _rtp_ext_id_to_type.end()) { + WarnL << "接收rtp时,忽略不识别的rtp ext, id=" << (int) pr.first; + pr.second.clearExt(); + continue; + } + pr.second.setType(it->second); + //重新赋值ext id为 ext type，作为后面处理ext的统一中间类型 + pr.second.setExtId((uint8_t) it->second); + } else { + pr.second.setType((RtpExtType) pr.first); + auto it = _rtp_ext_type_to_id.find((RtpExtType) pr.first); + if (it == _rtp_ext_type_to_id.end()) { + WarnL << "发送rtp时, 忽略不被客户端支持rtp ext:" << pr.second.dumpString(); + pr.second.clearExt(); + continue; + } + //重新赋值ext id为客户端sdp声明的类型 + pr.second.setExtId(it->second); } - setExtType(pr.second, it->first); - setExtType(pr.second, it->second); - pr.second.setExtId((uint8_t) it->second); } } @@ -726,7 +732,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } //修改ext id至统一 - changeRtpExtId(rtp, _rtp_ext_id_to_type); + changeRtpExtId(rtp, true); //解析并排序rtp info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; @@ -808,14 +814,15 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) { auto pr = (pair *) ctx; auto header = (RtpHeader *) buf; - changeRtpExtId(header, _rtp_ext_type_to_id); if (!pr->first || !pr->second->plan_rtx) { //普通的rtp,或者不支持rtx, 修改目标pt和ssrc + changeRtpExtId(header, false, false); header->pt = pr->second->plan_rtp->pt; header->ssrc = htonl(pr->second->answer_ssrc_rtp); } else { //重传的rtp, rtx + changeRtpExtId(header, false, true); header->pt = pr->second->plan_rtx->pt; if (pr->second->answer_ssrc_rtx) { //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 11544eb6..7ed7b9dc 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -338,6 +338,7 @@ private: SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; + void changeRtpExtId(const RtpHeader *header, bool is_recv, bool is_rtx = false) const; class RtpPayloadInfo { public: From 5f05df98b0cf44fc0904cbd686f687d93fd8c62b Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 21:17:55 +0800 Subject: [PATCH 193/218] =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E8=8E=B7=E5=8F=96?= =?UTF-8?q?=E7=B1=BB=E5=9E=8B=E6=96=B9=E6=B3=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 4 ++++ webrtc/RtpExt.h | 1 + 2 files changed, 5 insertions(+) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index a3e2ec3a..b74ef0c8 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -558,3 +558,7 @@ void RtpExt::clearExt(){ void RtpExt::setType(RtpExtType type) { _type = type; } + +RtpExtType RtpExt::getType() const { + return _type; +} \ No newline at end of file diff --git a/webrtc/RtpExt.h b/webrtc/RtpExt.h index 9c59129c..9520bac6 100644 --- a/webrtc/RtpExt.h +++ b/webrtc/RtpExt.h @@ -59,6 +59,7 @@ public: static const char *getExtName(RtpExtType type); void setType(RtpExtType type); + RtpExtType getType() const; string dumpString() const; uint8_t getAudioLevel(bool *vad) const; From c6d2b41d977410fe57ee9aa053919b9c16c413d6 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 21:49:46 +0800 Subject: [PATCH 194/218] =?UTF-8?q?=E5=AE=8C=E5=96=84rtp=20ext=E7=9B=B8?= =?UTF-8?q?=E5=85=B3=E5=A4=84=E7=90=86=E6=96=B9=E6=B3=95?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 8 ++++---- webrtc/WebRtcTransport.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 5d7b0cdc..0d8fb0d8 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -668,7 +668,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::changeRtpExtId(const RtpHeader *header, bool is_recv, bool is_rtx) const{ +void WebRtcTransportImp::changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx) const{ auto ext_map = RtpExt::getExtValue(header); for (auto &pr : ext_map) { if (is_recv) { @@ -732,7 +732,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } //修改ext id至统一 - changeRtpExtId(rtp, true); + changeRtpExtId(info.get(), rtp, true, rtx); //解析并排序rtp info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; @@ -817,12 +817,12 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void * if (!pr->first || !pr->second->plan_rtx) { //普通的rtp,或者不支持rtx, 修改目标pt和ssrc - changeRtpExtId(header, false, false); + changeRtpExtId(pr->second, header, false, false); header->pt = pr->second->plan_rtp->pt; header->ssrc = htonl(pr->second->answer_ssrc_rtp); } else { //重传的rtp, rtx - changeRtpExtId(header, false, true); + changeRtpExtId(pr->second, header, false, true); header->pt = pr->second->plan_rtx->pt; if (pr->second->answer_ssrc_rtx) { //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 7ed7b9dc..7f0dbb6b 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -338,7 +338,6 @@ private: SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; - void changeRtpExtId(const RtpHeader *header, bool is_recv, bool is_rtx = false) const; class RtpPayloadInfo { public: @@ -359,6 +358,7 @@ private: void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack); + void changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx = false) const; private: uint16_t _rtx_seq[2] = {0, 0}; From 44b0e13ba7675b91258cc54f58e142d7c747cc4e Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 22:12:26 +0800 Subject: [PATCH 195/218] =?UTF-8?q?=E5=AE=8C=E5=96=84=E6=89=93=E5=8D=B0svc?= =?UTF-8?q?=20tid?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/RtpExt.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index b74ef0c8..ecc39c5d 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -305,7 +305,7 @@ string RtpExt::dumpString() const { break; } case RtpExtType::framemarking : { - printer << "framemarking tid:" << getFramemarkingTID(); + printer << "framemarking tid:" << (int)getFramemarkingTID(); break; } default: { From 5b57b9d6aeef33d30a2c0b01bc697298f55e1f12 Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sun, 16 May 2021 22:13:08 +0800 Subject: [PATCH 196/218] =?UTF-8?q?=E7=A1=AE=E4=BF=9Drtc=E6=92=AD=E6=94=BE?= =?UTF-8?q?=E4=B8=8D=E5=BC=BA=E5=BC=95=E7=94=A8rtsp=E7=9B=B4=E6=92=AD?= =?UTF-8?q?=E6=BA=90?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 0d8fb0d8..15ffe15e 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -344,7 +344,7 @@ void WebRtcTransportImp::onDestory() { //流量统计事件广播 GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold); - if (_play_src) { + if (_reader) { WarnL << "RTC播放器(" << _media_info._vhost << "/" << _media_info._app << "/" @@ -457,6 +457,13 @@ void WebRtcTransportImp::onStartWebRTC() { strongSelf->onSendRtp(rtp, ++i == pkt->size()); }); }); + _reader->setDetachCB([weak_self](){ + auto strongSelf = weak_self.lock(); + if (!strongSelf) { + return; + } + strongSelf->onShutdown(SockException(Err_eof, "rtsp ring buffer detached")); + }); RtcSession rtsp_send_sdp; rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); @@ -473,6 +480,8 @@ void WebRtcTransportImp::onStartWebRTC() { } } } + //使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体 + _play_src = nullptr; } void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ @@ -849,7 +858,7 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void * } void WebRtcTransportImp::onShutdown(const SockException &ex){ - InfoL << ex.what(); + WarnL << ex.what(); _self = nullptr; } From d74589945accc129032eefc8c29abdfcd8b89e72 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 19 May 2021 18:39:23 +0800 Subject: [PATCH 197/218] =?UTF-8?q?=E8=BF=98=E5=8E=9F?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/readme.md | 48 +++++++++++++++++++++++++++++++++++++----------- 1 file changed, 37 insertions(+), 11 deletions(-) diff --git a/webrtc/readme.md b/webrtc/readme.md index 2884df14..ff832278 100644 --- a/webrtc/readme.md +++ b/webrtc/readme.md @@ -1,11 +1,37 @@ -14 urn:ietf:params:rtp-hdrext:toffset -2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time -13 urn:3gpp:video-orientation -3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 -12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay -11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type -7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing -8 http://www.webrtc.org/experiments/rtp-hdrext/color-space -4 urn:ietf:params:rtp-hdrext:sdes:mid -5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id -6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id \ No newline at end of file +# 致谢与声明 +本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为： + +- ice相关功能： + - IceServer.cpp + - IceServer.hpp + - StunPacket.cpp + - StunPacket.hpp + - Utils.hpp + +- dtls相关功能： + - DtlsTransport.cpp + - DtlsTransport.hpp + +- srtp相关功能： + - SrtpSession.cpp + - SrtpSession.hpp + + +以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者， +用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。 + +同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者， +在集成MediaSoup相关代码前期，主要参考这个项目。 + +另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)， +其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js + +# 现状与规划 +ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有： + +- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。 +- 2、实现rtp重传等相关功能。 +- 3、实现simulecast相关功能。 +- 4、fec、rtp扩展等其他功能。 +- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。 + From 08d4edddb0573b4a4d062f1e8fc772c0ef2c0ebc Mon Sep 17 00:00:00 2001 From: xiongziliang <771730766@qq.com> Date: Sat, 22 May 2021 09:13:00 +0800 Subject: [PATCH 198/218] =?UTF-8?q?=E4=BF=AE=E5=A4=8Dhttp=E6=96=87?= =?UTF-8?q?=E4=BB=B6=E6=9C=8D=E5=8A=A1=E5=99=A8=E7=9B=B8=E5=85=B3bug?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Http/HttpFileManager.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/Http/HttpFileManager.cpp b/src/Http/HttpFileManager.cpp index 735c053c..2d156e44 100644 --- a/src/Http/HttpFileManager.cpp +++ b/src/Http/HttpFileManager.cpp @@ -577,7 +577,7 @@ void HttpResponseInvokerImp::responseFile(const StrCaseMap &requestHeader, //分节下载 code = 206; iRangeStart = atoll(FindField(strRange.data(), "bytes=", "-").data()); - iRangeEnd = atoll(FindField(strRange.data(), "-", "\r\n").data()); + iRangeEnd = atoll(FindField(strRange.data(), "-", nullptr).data()); if (iRangeEnd == 0) { iRangeEnd = fileSize - 1; } From d74779dc021bf0bedf893f8d46b06eb1756b803a Mon Sep 17 00:00:00 2001 From: xgj Date: Fri, 4 Jun 2021 18:06:40 +0800 Subject: [PATCH 199/218] set server config can dynamic add config for add multi ffmpeg cmd template --- server/WebApi.cpp | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 1d6535a4..7ef77828 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -401,7 +401,10 @@ void installWebApi() { for (auto &pr : allArgs) { if (ini.find(pr.first) == ini.end()) { //没有这个key - continue; + //continue; + // 新增配置选项,为了动态添加多个ffmpeg cmd 模板 + ini[pr.first] = pr.second; + continue;// 防止changed变化 } if (ini[pr.first] == pr.second) { continue; From 0f3a8f4a9db5cb77148d5308f4352da16946d32d Mon Sep 17 00:00:00 2001 From: xgj Date: Mon, 7 Jun 2021 17:32:28 +0800 Subject: [PATCH 200/218] fix webrtc to rtmp multi slice on frame error --- src/Extension/H264Rtmp.cpp | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/Extension/H264Rtmp.cpp b/src/Extension/H264Rtmp.cpp index ca58472a..b196cb6e 100644 --- a/src/Extension/H264Rtmp.cpp +++ b/src/Extension/H264Rtmp.cpp @@ -183,7 +183,7 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) { } } - if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || type == H264Frame::NAL_B_P)) { + if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || pcData[1]&0x80 != 0)) { RtmpCodec::inputRtmp(_lastPacket); _lastPacket = nullptr; } @@ -214,10 +214,12 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) { _lastPacket->buffer.append((char *) &size, 4); _lastPacket->buffer.append(pcData, iLen); _lastPacket->body_size = _lastPacket->buffer.size(); + /* if (type == H264Frame::NAL_B_P) { RtmpCodec::inputRtmp(_lastPacket); _lastPacket = nullptr; } + */ } void H264RtmpEncoder::makeVideoConfigPkt() { From bb37bce3aca7bc267d537070e0cb84d287ce18d9 Mon Sep 17 00:00:00 2001 From: xgj Date: Mon, 7 Jun 2021 18:40:47 +0800 Subject: [PATCH 201/218] update zlmrtcclient.js --- www/webrtc/ZLMRTCClient.js | 219 ++++++++++++++++++++++++++++++--- www/webrtc/ZLMRTCClient.js.map | 2 +- www/webrtc/index.html | 192 +++++++++++++++++++++++++++++ www/webrtc/index_play.html | 97 --------------- www/webrtc/index_push.html | 97 --------------- 5 files changed, 393 insertions(+), 214 deletions(-) create mode 100644 www/webrtc/index.html delete mode 100644 www/webrtc/index_play.html delete mode 100644 www/webrtc/index_push.html diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js index 969178b6..c495dc1a 100644 --- a/www/webrtc/ZLMRTCClient.js +++ b/www/webrtc/ZLMRTCClient.js @@ -4,13 +4,14 @@ var ZLMRTCClient = (function (exports) { const Events$1 = { WEBRTC_NOT_SUPPORT: 'WEBRTC_NOT_SUPPORT', WEBRTC_ICE_CANDIDATE_ERROR: 'WEBRTC_ICE_CANDIDATE_ERROR', - WEBRTC_OFFER_answer_EXCHANGE_FAILED: 'WEBRTC_OFFER_answer_EXCHANGE_FAILED', + WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED: 'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED', WEBRTC_ON_REMOTE_STREAMS: 'WEBRTC_ON_REMOTE_STREAMS', - WEBRTC_ON_LOCAL_STREAM: 'WEBRTC_ON_LOCAL_STREAM' + WEBRTC_ON_LOCAL_STREAM: 'WEBRTC_ON_LOCAL_STREAM', + CAPTURE_STREAM_FAILED: 'CAPTURE_STREAM_FAILED' }; const VERSION = '1.0.1'; - const BUILD_DATE = 'Wed Mar 31 2021 14:49:19 GMT+0800 (China Standard Time)'; + const BUILD_DATE = 'Mon Jun 07 2021 18:09:53 GMT+0800 (China Standard Time)'; // Copyright (C) <2018> Intel Corporation // @@ -7284,7 +7285,15 @@ var ZLMRTCClient = (function (exports) { debug: false, // if output debug log zlmsdpUrl: '', - simulecast: false + simulecast: false, + useCamera: true, + audioEnable: true, + videoEnable: true, + recvOnly: false, + resolution: { + w: 0, + h: 0 + } }; this.options = Object.assign({}, defaults, options); @@ -7303,13 +7312,80 @@ var ZLMRTCClient = (function (exports) { this.pc.onicecandidate = this.e.onicecandidate; this.pc.onicecandidateerror = this.e.onicecandidateerror; this.pc.ontrack = this.e.ontrack; - this.start(); + if (!this.options.recvOnly && (this.options.audioEnable || this.options.videoEnable)) this.start();else this.receive(); + } + + receive() { + + const AudioTransceiverInit = { + direction: 'recvonly', + sendEncodings: [] + }; + const VideoTransceiverInit = { + direction: 'recvonly', + sendEncodings: [] + }; + this.pc.addTransceiver('audio', AudioTransceiverInit); + this.pc.addTransceiver('video', VideoTransceiverInit); + this.pc.createOffer().then(desc => { + log(this.TAG, 'offer:', desc.sdp); + this.pc.setLocalDescription(desc).then(() => { + axios({ + method: 'post', + url: this.options.zlmsdpUrl, + responseType: 'json', + data: desc.sdp, + headers: { + 'Content-Type': 'text/plain;charset=utf-8' + } + }).then(response => { + let ret = response.data; //JSON.parse(response.data); + + if (ret.code != 0) { + // mean failed for offer/anwser exchange + this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); + return; + } + + let anwser = {}; + anwser.sdp = ret.sdp; + anwser.type = 'answer'; + log(this.TAG, 'answer:', ret.sdp); + this.pc.setRemoteDescription(anwser).then(() => { + log(this.TAG, 'set remote sucess'); + }).catch(e => { + error(this.TAG, e); + }); + }); + }); + }).catch(e => { + error(this.TAG, e); + }); } start() { - //todo 此处修改捕获桌面或摄像头 - let audioConstraints = new AudioTrackConstraints(AudioSourceInfo.MIC); - let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.CAMERA); + let videoConstraints = false; + let audioConstraints = false; + + if (this.options.useCamera) { + if (this.options.videoEnable) videoConstraints = new VideoTrackConstraints(VideoSourceInfo.CAMERA); + if (this.options.audioEnable) audioConstraints = new AudioTrackConstraints(AudioSourceInfo.MIC); + } else { + if (this.options.videoEnable) { + videoConstraints = new VideoTrackConstraints(VideoSourceInfo.SCREENCAST); + if (this.options.audioEnable) audioConstraints = new AudioTrackConstraints(AudioSourceInfo.SCREENCAST); + } else { + if (this.options.audioEnable) audioConstraints = new AudioTrackConstraints(AudioSourceInfo.MIC);else { + // error shared display media not only audio + error(this.TAG, 'error paramter'); + } + } + } + + if (this.options.resolution.w != 0 && this.options.resolution.h != 0 && typeof videoConstraints == 'object') { + videoConstraints.resolution = new Resolution(this.options.resolution.w, this.options.resolution.h); + } + MediaStreamFactory.createMediaStream(new StreamConstraints(audioConstraints, videoConstraints)).then(stream => { this._localStream = stream; this.dispatch(Events$1.WEBRTC_ON_LOCAL_STREAM, stream); @@ -7322,7 +7398,7 @@ var ZLMRTCClient = (function (exports) { sendEncodings: [] }; - if (this.options.simulecast) { + if (this.options.simulecast && stream.getVideoTracks().length > 0) { VideoTransceiverInit.sendEncodings = [{ rid: 'q', active: true, @@ -7337,8 +7413,19 @@ var ZLMRTCClient = (function (exports) { }]; } - this.pc.addTransceiver(stream.getAudioTracks()[0], AudioTransceiverInit); - this.pc.addTransceiver(stream.getVideoTracks()[0], VideoTransceiverInit); + if (stream.getAudioTracks().length > 0) { + this.pc.addTransceiver(stream.getAudioTracks()[0], AudioTransceiverInit); + } else { + AudioTransceiverInit.direction = 'recvonly'; + this.pc.addTransceiver('audio', AudioTransceiverInit); + } + + if (stream.getVideoTracks().length > 0) { + this.pc.addTransceiver(stream.getVideoTracks()[0], VideoTransceiverInit); + } else { + VideoTransceiverInit.direction = 'recvonly'; + this.pc.addTransceiver('video', VideoTransceiverInit); + } /* stream.getTracks().forEach((track,idx)=>{ debug.log(this.TAG,track); @@ -7346,6 +7433,7 @@ var ZLMRTCClient = (function (exports) { }); */ + this.pc.createOffer().then(desc => { log(this.TAG, 'offer:', desc.sdp); this.pc.setLocalDescription(desc).then(() => { @@ -7358,19 +7446,19 @@ var ZLMRTCClient = (function (exports) { 'Content-Type': 'text/plain;charset=utf-8' } }).then(response => { - let ret = response.data; + let ret = response.data; //JSON.parse(response.data); if (ret.code != 0) { - // mean failed for offer/answer exchange - this.dispatch(Events$1.WEBRTC_OFFER_answer_EXCHANGE_FAILED, ret); + // mean failed for offer/anwser exchange + this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); return; } - let answer = {}; - answer.sdp = ret.sdp; - answer.type = 'answer'; + let anwser = {}; + anwser.sdp = ret.sdp; + anwser.type = 'answer'; log(this.TAG, 'answer:', ret.sdp); - this.pc.setRemoteDescription(answer).then(() => { + this.pc.setRemoteDescription(anwser).then(() => { log(this.TAG, 'set remote sucess'); }).catch(e => { error(this.TAG, e); @@ -7381,7 +7469,7 @@ var ZLMRTCClient = (function (exports) { error(this.TAG, e); }); }).catch(e => { - error(this.TAG, e); + this.dispatch(Events$1.CAPTURE_STREAM_FAILED); //debug.error(this.TAG,e); }); //const offerOptions = {}; /* @@ -7449,15 +7537,108 @@ var ZLMRTCClient = (function (exports) { } + const quickScan = [{ + 'label': '4K(UHD)', + 'width': 3840, + 'height': 2160 + }, { + 'label': '1080p(FHD)', + 'width': 1920, + 'height': 1080 + }, { + 'label': 'UXGA', + 'width': 1600, + 'height': 1200, + 'ratio': '4:3' + }, { + 'label': '720p(HD)', + 'width': 1280, + 'height': 720 + }, { + 'label': 'SVGA', + 'width': 800, + 'height': 600 + }, { + 'label': 'VGA', + 'width': 640, + 'height': 480 + }, { + 'label': '360p(nHD)', + 'width': 640, + 'height': 360 + }, { + 'label': 'CIF', + 'width': 352, + 'height': 288 + }, { + 'label': 'QVGA', + 'width': 320, + 'height': 240 + }, { + 'label': 'QCIF', + 'width': 176, + 'height': 144 + }, { + 'label': 'QQVGA', + 'width': 160, + 'height': 120 + }]; + function GetSupportCameraResolutions$1() { + return new Promise(function (resolve, reject) { + let resolutions = []; + let ok = 0; + let err = 0; + + for (let i = 0; i < quickScan.length; ++i) { + let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.CAMERA); + videoConstraints.resolution = new Resolution(quickScan[i].width, quickScan[i].height); + MediaStreamFactory.createMediaStream(new StreamConstraints(false, videoConstraints)).then(stream => { + resolutions.push(quickScan[i]); + ok++; + + if (ok + err == quickScan.length) { + resolve(resolutions); + } + }).catch(e => { + err++; + + if (ok + err == quickScan.length) { + resolve(resolutions); + } + }); + } + }); + } + function GetAllScanResolution$1() { + return quickScan; + } + function isSupportResolution$1(w, h) { + return new Promise(function (resolve, reject) { + let videoConstraints = new VideoTrackConstraints(VideoSourceInfo.CAMERA); + videoConstraints.resolution = new Resolution(w, h); + MediaStreamFactory.createMediaStream(new StreamConstraints(false, videoConstraints)).then(stream => { + resolve(); + }).catch(e => { + reject(e); + }); + }); + } + console.log('build date:', BUILD_DATE); console.log('version:', VERSION); const Events = Events$1; const Media = media; const Endpoint = RTCEndpoint; + const GetSupportCameraResolutions = GetSupportCameraResolutions$1; + const GetAllScanResolution = GetAllScanResolution$1; + const isSupportResolution = isSupportResolution$1; exports.Endpoint = Endpoint; exports.Events = Events; + exports.GetAllScanResolution = GetAllScanResolution; + exports.GetSupportCameraResolutions = GetSupportCameraResolutions; exports.Media = Media; + exports.isSupportResolution = isSupportResolution; Object.defineProperty(exports, '__esModule', { value: true }); diff --git a/www/webrtc/ZLMRTCClient.js.map b/www/webrtc/ZLMRTCClient.js.map index 7f4bbb98..28184d8d 100644 --- a/www/webrtc/ZLMRTCClient.js.map +++ b/www/webrtc/ZLMRTCClient.js.map @@ -1 +1 @@ -{"version":3,"file":"ZLMRTCClient.js","sources":["../src/base/event.js","../src/ulity/version.js","../src/base/utils.js","../src/base/mediaformat.js","../node_modules/webrtc-adapter/src/js/utils.js","../node_modules/webrtc-adapter/src/js/chrome/getusermedia.js","../node_modules/webrtc-adapter/src/js/chrome/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/chrome/chrome_shim.js","../node_modules/webrtc-adapter/src/js/edge/filtericeservers.js","../node_modules/sdp/sdp.js","../node_modules/rtcpeerconnection-shim/rtcpeerconnection.js","../node_modules/webrtc-adapter/src/js/edge/getusermedia.js","../node_modules/webrtc-adapter/src/js/edge/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/edge/edge_shim.js","../node_modules/webrtc-adapter/src/js/firefox/getusermedia.js","../node_modules/webrtc-adapter/src/js/firefox/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/firefox/firefox_shim.js","../node_modules/webrtc-adapter/src/js/safari/safari_shim.js","../node_modules/webrtc-adapter/src/js/common_shim.js","../node_modules/webrtc-adapter/src/js/adapter_factory.js","../node_modules/webrtc-adapter/src/js/adapter_core.js","../src/base/mediastream-factory.js","../src/base/export.js","../src/ulity/debug.js","../src/ulity/event.js","../node_modules/axios/lib/helpers/bind.js","../node_modules/axios/lib/utils.js","../node_modules/axios/lib/helpers/buildURL.js","../node_modules/axios/lib/core/InterceptorManager.js","../node_modules/axios/lib/core/transformData.js","../node_modules/axios/lib/cancel/isCancel.js","../node_modules/axios/lib/helpers/normalizeHeaderName.js","../node_modules/axios/lib/core/enhanceError.js","../node_modules/axios/lib/core/createError.js","../node_modules/axios/lib/core/settle.js","../node_modules/axios/lib/helpers/cookies.js","../node_modules/axios/lib/helpers/isAbsoluteURL.js","../node_modules/axios/lib/helpers/combineURLs.js","../node_modules/axios/lib/core/buildFullPath.js","../node_modules/axios/lib/helpers/parseHeaders.js","../node_modules/axios/lib/helpers/isURLSameOrigin.js","../node_modules/axios/lib/adapters/xhr.js","../node_modules/axios/lib/defaults.js","../node_modules/axios/lib/core/dispatchRequest.js","../node_modules/axios/lib/core/mergeConfig.js","../node_modules/axios/lib/core/Axios.js","../node_modules/axios/lib/cancel/Cancel.js","../node_modules/axios/lib/cancel/CancelToken.js","../node_modules/axios/lib/helpers/spread.js","../node_modules/axios/lib/helpers/isAxiosError.js","../node_modules/axios/lib/axios.js","../node_modules/axios/index.js","../src/endpoint/endpoint.js","../src/export.js"],"sourcesContent":["const Events = {\n\tWEBRTC_NOT_SUPPORT : 'WEBRTC_NOT_SUPPORT',\n\tWEBRTC_ICE_CANDIDATE_ERROR : 'WEBRTC_ICE_CANDIDATE_ERROR',\n\tWEBRTC_OFFER_ANWSER_EXCHANGE_FAILED:'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED',\n\tWEBRTC_ON_REMOTE_STREAMS:'WEBRTC_ON_REMOTE_STREAMS',\n\tWEBRTC_ON_LOCAL_STREAM:'WEBRTC_ON_LOCAL_STREAM'\n};\n\nexport default Events;","export const VERSION = '__VERSION__';\nexport const BUILD_DATE = '__BUILD_DATE__';","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n\n// eslint-disable-next-line require-jsdoc\nexport function isFirefox() {\n return window.navigator.userAgent.match('Firefox') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isChrome() {\n return window.navigator.userAgent.match('Chrome') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isSafari() {\n return /^((?!chrome|android).)*safari/i.test(window.navigator.userAgent);\n}\n// eslint-disable-next-line require-jsdoc\nexport function isEdge() {\n return window.navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/) !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function createUuid() {\n return 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx'.replace(/[xy]/g, function(c) {\n const r = Math.random() * 16 | 0;\n const v = c === 'x' ? r : (r & 0x3 | 0x8);\n return v.toString(16);\n });\n}","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n/**\n * @class AudioSourceInfo\n * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const AudioSourceInfo = {\n MIC: 'mic',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class VideoSourceInfo\n * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const VideoSourceInfo = {\n CAMERA: 'camera',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class TrackKind\n * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const TrackKind = {\n /**\n * Audio tracks.\n * @type string\n */\n AUDIO: 'audio',\n /**\n * Video tracks.\n * @type string\n */\n VIDEO: 'video',\n /**\n * Both audio and video tracks.\n * @type string\n */\n AUDIO_AND_VIDEO: 'av',\n};\n/**\n * @class Resolution\n * @memberOf Owt.Base\n * @classDesc The Resolution defines the size of a rectangle.\n * @constructor\n * @param {number} width\n * @param {number} height\n */\nexport class Resolution {\n // eslint-disable-next-line require-jsdoc\n constructor(width, height) {\n /**\n * @member {number} width\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.width = width;\n /**\n * @member {number} height\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.height = height;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nlet logDisabled_ = true;\nlet deprecationWarnings_ = true;\n\n/**\n * Extract browser version out of the provided user agent string.\n *\n * @param {!string} uastring userAgent string.\n * @param {!string} expr Regular expression used as match criteria.\n * @param {!number} pos position in the version string to be returned.\n * @return {!number} browser version.\n */\nexport function extractVersion(uastring, expr, pos) {\n const match = uastring.match(expr);\n return match && match.length >= pos && parseInt(match[pos], 10);\n}\n\n// Wraps the peerconnection event eventNameToWrap in a function\n// which returns the modified event object (or false to prevent\n// the event).\nexport function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) {\n if (!window.RTCPeerConnection) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n const nativeAddEventListener = proto.addEventListener;\n proto.addEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap) {\n return nativeAddEventListener.apply(this, arguments);\n }\n const wrappedCallback = (e) => {\n const modifiedEvent = wrapper(e);\n if (modifiedEvent) {\n if (cb.handleEvent) {\n cb.handleEvent(modifiedEvent);\n } else {\n cb(modifiedEvent);\n }\n }\n };\n this._eventMap = this._eventMap || {};\n if (!this._eventMap[eventNameToWrap]) {\n this._eventMap[eventNameToWrap] = new Map();\n }\n this._eventMap[eventNameToWrap].set(cb, wrappedCallback);\n return nativeAddEventListener.apply(this, [nativeEventName,\n wrappedCallback]);\n };\n\n const nativeRemoveEventListener = proto.removeEventListener;\n proto.removeEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap || !this._eventMap\n || !this._eventMap[eventNameToWrap]) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n if (!this._eventMap[eventNameToWrap].has(cb)) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n const unwrappedCb = this._eventMap[eventNameToWrap].get(cb);\n this._eventMap[eventNameToWrap].delete(cb);\n if (this._eventMap[eventNameToWrap].size === 0) {\n delete this._eventMap[eventNameToWrap];\n }\n if (Object.keys(this._eventMap).length === 0) {\n delete this._eventMap;\n }\n return nativeRemoveEventListener.apply(this, [nativeEventName,\n unwrappedCb]);\n };\n\n Object.defineProperty(proto, 'on' + eventNameToWrap, {\n get() {\n return this['_on' + eventNameToWrap];\n },\n set(cb) {\n if (this['_on' + eventNameToWrap]) {\n this.removeEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap]);\n delete this['_on' + eventNameToWrap];\n }\n if (cb) {\n this.addEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap] = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n}\n\nexport function disableLog(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n logDisabled_ = bool;\n return (bool) ? 'adapter.js logging disabled' :\n 'adapter.js logging enabled';\n}\n\n/**\n * Disable or enable deprecation warnings\n * @param {!boolean} bool set to true to disable warnings.\n */\nexport function disableWarnings(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n deprecationWarnings_ = !bool;\n return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled');\n}\n\nexport function log() {\n if (typeof window === 'object') {\n if (logDisabled_) {\n return;\n }\n if (typeof console !== 'undefined' && typeof console.log === 'function') {\n console.log.apply(console, arguments);\n }\n }\n}\n\n/**\n * Shows a deprecation warning suggesting the modern and spec-compatible API.\n */\nexport function deprecated(oldMethod, newMethod) {\n if (!deprecationWarnings_) {\n return;\n }\n console.warn(oldMethod + ' is deprecated, please use ' + newMethod +\n ' instead.');\n}\n\n/**\n * Browser detector.\n *\n * @return {object} result containing browser and version\n * properties.\n */\nexport function detectBrowser(window) {\n // Returned result object.\n const result = {browser: null, version: null};\n\n // Fail early if it's not a browser\n if (typeof window === 'undefined' || !window.navigator) {\n result.browser = 'Not a browser.';\n return result;\n }\n\n const {navigator} = window;\n\n if (navigator.mozGetUserMedia) { // Firefox.\n result.browser = 'firefox';\n result.version = extractVersion(navigator.userAgent,\n /Firefox\\/(\\d+)\\./, 1);\n } else if (navigator.webkitGetUserMedia ||\n (window.isSecureContext === false && window.webkitRTCPeerConnection &&\n !window.RTCIceGatherer)) {\n // Chrome, Chromium, Webview, Opera.\n // Version matches Chrome/WebRTC version.\n // Chrome 74 removed webkitGetUserMedia on http as well so we need the\n // more complicated fallback to webkitRTCPeerConnection.\n result.browser = 'chrome';\n result.version = extractVersion(navigator.userAgent,\n /Chrom(e|ium)\\/(\\d+)\\./, 2);\n } else if (navigator.mediaDevices &&\n navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/)) { // Edge.\n result.browser = 'edge';\n result.version = extractVersion(navigator.userAgent,\n /Edge\\/(\\d+).(\\d+)$/, 2);\n } else if (window.RTCPeerConnection &&\n navigator.userAgent.match(/AppleWebKit\\/(\\d+)\\./)) { // Safari.\n result.browser = 'safari';\n result.version = extractVersion(navigator.userAgent,\n /AppleWebKit\\/(\\d+)\\./, 1);\n result.supportsUnifiedPlan = window.RTCRtpTransceiver &&\n 'currentDirection' in window.RTCRtpTransceiver.prototype;\n } else { // Default fallthrough: not supported.\n result.browser = 'Not a supported browser.';\n return result;\n }\n\n return result;\n}\n\n/**\n * Checks if something is an object.\n *\n * @param {*} val The something you want to check.\n * @return true if val is an object, false otherwise.\n */\nfunction isObject(val) {\n return Object.prototype.toString.call(val) === '[object Object]';\n}\n\n/**\n * Remove all empty objects and undefined values\n * from a nested object -- an enhanced and vanilla version\n * of Lodash's `compact`.\n */\nexport function compactObject(data) {\n if (!isObject(data)) {\n return data;\n }\n\n return Object.keys(data).reduce(function(accumulator, key) {\n const isObj = isObject(data[key]);\n const value = isObj ? compactObject(data[key]) : data[key];\n const isEmptyObject = isObj && !Object.keys(value).length;\n if (value === undefined || isEmptyObject) {\n return accumulator;\n }\n return Object.assign(accumulator, {[key]: value});\n }, {});\n}\n\n/* iterates the stats graph recursively. */\nexport function walkStats(stats, base, resultSet) {\n if (!base || resultSet.has(base.id)) {\n return;\n }\n resultSet.set(base.id, base);\n Object.keys(base).forEach(name => {\n if (name.endsWith('Id')) {\n walkStats(stats, stats.get(base[name]), resultSet);\n } else if (name.endsWith('Ids')) {\n base[name].forEach(id => {\n walkStats(stats, stats.get(id), resultSet);\n });\n }\n });\n}\n\n/* filter getStats for a sender/receiver track. */\nexport function filterStats(result, track, outbound) {\n const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp';\n const filteredResult = new Map();\n if (track === null) {\n return filteredResult;\n }\n const trackStats = [];\n result.forEach(value => {\n if (value.type === 'track' &&\n value.trackIdentifier === track.id) {\n trackStats.push(value);\n }\n });\n trackStats.forEach(trackStat => {\n result.forEach(stats => {\n if (stats.type === streamStatsType && stats.trackId === trackStat.id) {\n walkStats(result, stats, filteredResult);\n }\n });\n });\n return filteredResult;\n}\n\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\nconst logging = utils.log;\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n\n if (!navigator.mediaDevices) {\n return;\n }\n\n const constraintsToChrome_ = function(c) {\n if (typeof c !== 'object' || c.mandatory || c.optional) {\n return c;\n }\n const cc = {};\n Object.keys(c).forEach(key => {\n if (key === 'require' || key === 'advanced' || key === 'mediaSource') {\n return;\n }\n const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]};\n if (r.exact !== undefined && typeof r.exact === 'number') {\n r.min = r.max = r.exact;\n }\n const oldname_ = function(prefix, name) {\n if (prefix) {\n return prefix + name.charAt(0).toUpperCase() + name.slice(1);\n }\n return (name === 'deviceId') ? 'sourceId' : name;\n };\n if (r.ideal !== undefined) {\n cc.optional = cc.optional || [];\n let oc = {};\n if (typeof r.ideal === 'number') {\n oc[oldname_('min', key)] = r.ideal;\n cc.optional.push(oc);\n oc = {};\n oc[oldname_('max', key)] = r.ideal;\n cc.optional.push(oc);\n } else {\n oc[oldname_('', key)] = r.ideal;\n cc.optional.push(oc);\n }\n }\n if (r.exact !== undefined && typeof r.exact !== 'number') {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_('', key)] = r.exact;\n } else {\n ['min', 'max'].forEach(mix => {\n if (r[mix] !== undefined) {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_(mix, key)] = r[mix];\n }\n });\n }\n });\n if (c.advanced) {\n cc.optional = (cc.optional || []).concat(c.advanced);\n }\n return cc;\n };\n\n const shimConstraints_ = function(constraints, func) {\n if (browserDetails.version >= 61) {\n return func(constraints);\n }\n constraints = JSON.parse(JSON.stringify(constraints));\n if (constraints && typeof constraints.audio === 'object') {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n constraints = JSON.parse(JSON.stringify(constraints));\n remap(constraints.audio, 'autoGainControl', 'googAutoGainControl');\n remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression');\n constraints.audio = constraintsToChrome_(constraints.audio);\n }\n if (constraints && typeof constraints.video === 'object') {\n // Shim facingMode for mobile & surface pro.\n let face = constraints.video.facingMode;\n face = face && ((typeof face === 'object') ? face : {ideal: face});\n const getSupportedFacingModeLies = browserDetails.version < 66;\n\n if ((face && (face.exact === 'user' || face.exact === 'environment' ||\n face.ideal === 'user' || face.ideal === 'environment')) &&\n !(navigator.mediaDevices.getSupportedConstraints &&\n navigator.mediaDevices.getSupportedConstraints().facingMode &&\n !getSupportedFacingModeLies)) {\n delete constraints.video.facingMode;\n let matches;\n if (face.exact === 'environment' || face.ideal === 'environment') {\n matches = ['back', 'rear'];\n } else if (face.exact === 'user' || face.ideal === 'user') {\n matches = ['front'];\n }\n if (matches) {\n // Look for matches in label, or use last cam for back (typical).\n return navigator.mediaDevices.enumerateDevices()\n .then(devices => {\n devices = devices.filter(d => d.kind === 'videoinput');\n let dev = devices.find(d => matches.some(match =>\n d.label.toLowerCase().includes(match)));\n if (!dev && devices.length && matches.includes('back')) {\n dev = devices[devices.length - 1]; // more likely the back cam\n }\n if (dev) {\n constraints.video.deviceId = face.exact ? {exact: dev.deviceId} :\n {ideal: dev.deviceId};\n }\n constraints.video = constraintsToChrome_(constraints.video);\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n });\n }\n }\n constraints.video = constraintsToChrome_(constraints.video);\n }\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n };\n\n const shimError_ = function(e) {\n if (browserDetails.version >= 64) {\n return e;\n }\n return {\n name: {\n PermissionDeniedError: 'NotAllowedError',\n PermissionDismissedError: 'NotAllowedError',\n InvalidStateError: 'NotAllowedError',\n DevicesNotFoundError: 'NotFoundError',\n ConstraintNotSatisfiedError: 'OverconstrainedError',\n TrackStartError: 'NotReadableError',\n MediaDeviceFailedDueToShutdown: 'NotAllowedError',\n MediaDeviceKillSwitchOn: 'NotAllowedError',\n TabCaptureError: 'AbortError',\n ScreenCaptureError: 'AbortError',\n DeviceCaptureError: 'AbortError'\n }[e.name] || e.name,\n message: e.message,\n constraint: e.constraint || e.constraintName,\n toString() {\n return this.name + (this.message && ': ') + this.message;\n }\n };\n };\n\n const getUserMedia_ = function(constraints, onSuccess, onError) {\n shimConstraints_(constraints, c => {\n navigator.webkitGetUserMedia(c, onSuccess, e => {\n if (onError) {\n onError(shimError_(e));\n }\n });\n });\n };\n navigator.getUserMedia = getUserMedia_.bind(navigator);\n\n // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia\n // function which returns a Promise, it does not accept spec-style\n // constraints.\n if (navigator.mediaDevices.getUserMedia) {\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(cs) {\n return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => {\n if (c.audio && !stream.getAudioTracks().length ||\n c.video && !stream.getVideoTracks().length) {\n stream.getTracks().forEach(track => {\n track.stop();\n });\n throw new DOMException('', 'NotFoundError');\n }\n return stream;\n }, e => Promise.reject(shimError_(e))));\n };\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nexport function shimGetDisplayMedia(window, getSourceId) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n // getSourceId is a function that returns a promise resolving with\n // the sourceId of the screen/window/tab to be shared.\n if (typeof getSourceId !== 'function') {\n console.error('shimGetDisplayMedia: getSourceId argument is not ' +\n 'a function');\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n return getSourceId(constraints)\n .then(sourceId => {\n const widthSpecified = constraints.video && constraints.video.width;\n const heightSpecified = constraints.video &&\n constraints.video.height;\n const frameRateSpecified = constraints.video &&\n constraints.video.frameRate;\n constraints.video = {\n mandatory: {\n chromeMediaSource: 'desktop',\n chromeMediaSourceId: sourceId,\n maxFrameRate: frameRateSpecified || 3\n }\n };\n if (widthSpecified) {\n constraints.video.mandatory.maxWidth = widthSpecified;\n }\n if (heightSpecified) {\n constraints.video.mandatory.maxHeight = heightSpecified;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n });\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimMediaStream(window) {\n window.MediaStream = window.MediaStream || window.webkitMediaStream;\n}\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in\n window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', {\n get() {\n return this._ontrack;\n },\n set(f) {\n if (this._ontrack) {\n this.removeEventListener('track', this._ontrack);\n }\n this.addEventListener('track', this._ontrack = f);\n },\n enumerable: true,\n configurable: true\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n if (!this._ontrackpoly) {\n this._ontrackpoly = (e) => {\n // onaddstream does not fire when a track is added to an existing\n // stream. But stream.onaddtrack is implemented so we use that.\n e.stream.addEventListener('addtrack', te => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === te.track.id);\n } else {\n receiver = {track: te.track};\n }\n\n const event = new Event('track');\n event.track = te.track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n e.stream.getTracks().forEach(track => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === track.id);\n } else {\n receiver = {track};\n }\n const event = new Event('track');\n event.track = track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n };\n this.addEventListener('addstream', this._ontrackpoly);\n }\n return origSetRemoteDescription.apply(this, arguments);\n };\n } else {\n // even if RTCRtpTransceiver is in window, it is only used and\n // emitted in unified-plan. Unfortunately this means we need\n // to unconditionally wrap the event.\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n if (!e.transceiver) {\n Object.defineProperty(e, 'transceiver',\n {value: {receiver: e.receiver}});\n }\n return e;\n });\n }\n}\n\nexport function shimGetSendersWithDtmf(window) {\n // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack.\n if (typeof window === 'object' && window.RTCPeerConnection &&\n !('getSenders' in window.RTCPeerConnection.prototype) &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype) {\n const shimSenderWithDtmf = function(pc, track) {\n return {\n track,\n get dtmf() {\n if (this._dtmf === undefined) {\n if (track.kind === 'audio') {\n this._dtmf = pc.createDTMFSender(track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n },\n _pc: pc\n };\n };\n\n // augment addTrack when getSenders is not available.\n if (!window.RTCPeerConnection.prototype.getSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n this._senders = this._senders || [];\n return this._senders.slice(); // return a copy of the internal state.\n };\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n let sender = origAddTrack.apply(this, arguments);\n if (!sender) {\n sender = shimSenderWithDtmf(this, track);\n this._senders.push(sender);\n }\n return sender;\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n origRemoveTrack.apply(this, arguments);\n const idx = this._senders.indexOf(sender);\n if (idx !== -1) {\n this._senders.splice(idx, 1);\n }\n };\n }\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._senders = this._senders || [];\n origAddStream.apply(this, [stream]);\n stream.getTracks().forEach(track => {\n this._senders.push(shimSenderWithDtmf(this, track));\n });\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._senders = this._senders || [];\n origRemoveStream.apply(this, [stream]);\n\n stream.getTracks().forEach(track => {\n const sender = this._senders.find(s => s.track === track);\n if (sender) { // remove sender\n this._senders.splice(this._senders.indexOf(sender), 1);\n }\n });\n };\n } else if (typeof window === 'object' && window.RTCPeerConnection &&\n 'getSenders' in window.RTCPeerConnection.prototype &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype &&\n window.RTCRtpSender &&\n !('dtmf' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = this._pc.createDTMFSender(this.track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n}\n\nexport function shimGetStats(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n\n // If selector is a function then we are in the old style stats so just\n // pass back the original getStats format to avoid breaking old users.\n if (arguments.length > 0 && typeof selector === 'function') {\n return origGetStats.apply(this, arguments);\n }\n\n // When spec-style getStats is supported, return those when called with\n // either no arguments or the selector argument is null.\n if (origGetStats.length === 0 && (arguments.length === 0 ||\n typeof selector !== 'function')) {\n return origGetStats.apply(this, []);\n }\n\n const fixChromeStats_ = function(response) {\n const standardReport = {};\n const reports = response.result();\n reports.forEach(report => {\n const standardStats = {\n id: report.id,\n timestamp: report.timestamp,\n type: {\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[report.type] || report.type\n };\n report.names().forEach(name => {\n standardStats[name] = report.stat(name);\n });\n standardReport[standardStats.id] = standardStats;\n });\n\n return standardReport;\n };\n\n // shim getStats with maplike support\n const makeMapStats = function(stats) {\n return new Map(Object.keys(stats).map(key => [key, stats[key]]));\n };\n\n if (arguments.length >= 2) {\n const successCallbackWrapper_ = function(response) {\n onSucc(makeMapStats(fixChromeStats_(response)));\n };\n\n return origGetStats.apply(this, [successCallbackWrapper_,\n selector]);\n }\n\n // promise-support\n return new Promise((resolve, reject) => {\n origGetStats.apply(this, [\n function(response) {\n resolve(makeMapStats(fixChromeStats_(response)));\n }, reject]);\n }).then(onSucc, onErr);\n };\n}\n\nexport function shimSenderReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender && window.RTCRtpReceiver)) {\n return;\n }\n\n // shim sender stats.\n if (!('getStats' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n const sender = this;\n return this._pc.getStats().then(result =>\n /* Note: this will include stats of all senders that\n * send a track with the same id as sender.track as\n * it is not possible to identify the RTCRtpSender.\n */\n utils.filterStats(result, sender.track, true));\n };\n }\n\n // shim receiver stats.\n if (!('getStats' in window.RTCRtpReceiver.prototype)) {\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers =\n function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n const receiver = this;\n return this._pc.getStats().then(result =>\n utils.filterStats(result, receiver.track, false));\n };\n }\n\n if (!('getStats' in window.RTCRtpSender.prototype &&\n 'getStats' in window.RTCRtpReceiver.prototype)) {\n return;\n }\n\n // shim RTCPeerConnection.getStats(track).\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n if (arguments.length > 0 &&\n arguments[0] instanceof window.MediaStreamTrack) {\n const track = arguments[0];\n let sender;\n let receiver;\n let err;\n this.getSenders().forEach(s => {\n if (s.track === track) {\n if (sender) {\n err = true;\n } else {\n sender = s;\n }\n }\n });\n this.getReceivers().forEach(r => {\n if (r.track === track) {\n if (receiver) {\n err = true;\n } else {\n receiver = r;\n }\n }\n return r.track === track;\n });\n if (err || (sender && receiver)) {\n return Promise.reject(new DOMException(\n 'There are more than one sender or receiver for the track.',\n 'InvalidAccessError'));\n } else if (sender) {\n return sender.getStats();\n } else if (receiver) {\n return receiver.getStats();\n }\n return Promise.reject(new DOMException(\n 'There is no sender or receiver for the track.',\n 'InvalidAccessError'));\n }\n return origGetStats.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrackWithNative(window) {\n // shim addTrack/removeTrack with native variants in order to make\n // the interactions with legacy getLocalStreams behave as in other browsers.\n // Keeps a mapping stream.id => [stream, rtpsenders...]\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n return Object.keys(this._shimmedLocalStreams)\n .map(streamId => this._shimmedLocalStreams[streamId][0]);\n };\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (!stream) {\n return origAddTrack.apply(this, arguments);\n }\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n const sender = origAddTrack.apply(this, arguments);\n if (!this._shimmedLocalStreams[stream.id]) {\n this._shimmedLocalStreams[stream.id] = [stream, sender];\n } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) {\n this._shimmedLocalStreams[stream.id].push(sender);\n }\n return sender;\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n const existingSenders = this.getSenders();\n origAddStream.apply(this, arguments);\n const newSenders = this.getSenders()\n .filter(newSender => existingSenders.indexOf(newSender) === -1);\n this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n delete this._shimmedLocalStreams[stream.id];\n return origRemoveStream.apply(this, arguments);\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n if (sender) {\n Object.keys(this._shimmedLocalStreams).forEach(streamId => {\n const idx = this._shimmedLocalStreams[streamId].indexOf(sender);\n if (idx !== -1) {\n this._shimmedLocalStreams[streamId].splice(idx, 1);\n }\n if (this._shimmedLocalStreams[streamId].length === 1) {\n delete this._shimmedLocalStreams[streamId];\n }\n });\n }\n return origRemoveTrack.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrack(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // shim addTrack and removeTrack.\n if (window.RTCPeerConnection.prototype.addTrack &&\n browserDetails.version >= 65) {\n return shimAddTrackRemoveTrackWithNative(window);\n }\n\n // also shim pc.getLocalStreams when addTrack is shimmed\n // to return the original streams.\n const origGetLocalStreams = window.RTCPeerConnection.prototype\n .getLocalStreams;\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n const nativeStreams = origGetLocalStreams.apply(this);\n this._reverseStreams = this._reverseStreams || {};\n return nativeStreams.map(stream => this._reverseStreams[stream.id]);\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n // Add identity mapping for consistency with addTrack.\n // Unless this is being used with a stream from addTrack.\n if (!this._reverseStreams[stream.id]) {\n const newStream = new window.MediaStream(stream.getTracks());\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n stream = newStream;\n }\n origAddStream.apply(this, [stream]);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]);\n delete this._reverseStreams[(this._streams[stream.id] ?\n this._streams[stream.id].id : stream.id)];\n delete this._streams[stream.id];\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n const streams = [].slice.call(arguments, 1);\n if (streams.length !== 1 ||\n !streams[0].getTracks().find(t => t === track)) {\n // this is not fully correct but all we can manage without\n // [[associated MediaStreams]] internal slot.\n throw new DOMException(\n 'The adapter.js addTrack polyfill only supports a single ' +\n ' stream which is associated with the specified track.',\n 'NotSupportedError');\n }\n\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n const oldStream = this._streams[stream.id];\n if (oldStream) {\n // this is using odd Chrome behaviour, use with caution:\n // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815\n // Note: we rely on the high-level addTrack/dtmf shim to\n // create the sender with a dtmf sender.\n oldStream.addTrack(track);\n\n // Trigger ONN async.\n Promise.resolve().then(() => {\n this.dispatchEvent(new Event('negotiationneeded'));\n });\n } else {\n const newStream = new window.MediaStream([track]);\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n this.addStream(newStream);\n }\n return this.getSenders().find(s => s.track === track);\n };\n\n // replace the internal stream id with the external one and\n // vice versa.\n function replaceInternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(internalStream.id, 'g'),\n externalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n function replaceExternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(externalStream.id, 'g'),\n internalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n ['createOffer', 'createAnswer'].forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n const args = arguments;\n const isLegacyCall = arguments.length &&\n typeof arguments[0] === 'function';\n if (isLegacyCall) {\n return nativeMethod.apply(this, [\n (description) => {\n const desc = replaceInternalStreamId(this, description);\n args[0].apply(null, [desc]);\n },\n (err) => {\n if (args[1]) {\n args[1].apply(null, err);\n }\n }, arguments[2]\n ]);\n }\n return nativeMethod.apply(this, arguments)\n .then(description => replaceInternalStreamId(this, description));\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n\n const origSetLocalDescription =\n window.RTCPeerConnection.prototype.setLocalDescription;\n window.RTCPeerConnection.prototype.setLocalDescription =\n function setLocalDescription() {\n if (!arguments.length || !arguments[0].type) {\n return origSetLocalDescription.apply(this, arguments);\n }\n arguments[0] = replaceExternalStreamId(this, arguments[0]);\n return origSetLocalDescription.apply(this, arguments);\n };\n\n // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier\n\n const origLocalDescription = Object.getOwnPropertyDescriptor(\n window.RTCPeerConnection.prototype, 'localDescription');\n Object.defineProperty(window.RTCPeerConnection.prototype,\n 'localDescription', {\n get() {\n const description = origLocalDescription.get.apply(this);\n if (description.type === '') {\n return description;\n }\n return replaceInternalStreamId(this, description);\n }\n });\n\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n // We can not yet check for sender instanceof RTCRtpSender\n // since we shim RTPSender. So we check if sender._pc is set.\n if (!sender._pc) {\n throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.', 'TypeError');\n }\n const isLocal = sender._pc === this;\n if (!isLocal) {\n throw new DOMException('Sender was not created by this connection.',\n 'InvalidAccessError');\n }\n\n // Search for the native stream the senders track belongs to.\n this._streams = this._streams || {};\n let stream;\n Object.keys(this._streams).forEach(streamid => {\n const hasTrack = this._streams[streamid].getTracks()\n .find(track => sender.track === track);\n if (hasTrack) {\n stream = this._streams[streamid];\n }\n });\n\n if (stream) {\n if (stream.getTracks().length === 1) {\n // if this is the last track of the stream, remove the stream. This\n // takes care of any shimmed _senders.\n this.removeStream(this._reverseStreams[stream.id]);\n } else {\n // relying on the same odd chrome behaviour as above.\n stream.removeTrack(sender.track);\n }\n this.dispatchEvent(new Event('negotiationneeded'));\n }\n };\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.webkitRTCPeerConnection;\n }\n if (!window.RTCPeerConnection) {\n return;\n }\n\n // shim implicit creation of RTCSessionDescription/RTCIceCandidate\n if (browserDetails.version < 53) {\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n}\n\n// Attempt to fix ONN in plan-b mode.\nexport function fixNegotiationNeeded(window, browserDetails) {\n utils.wrapPeerConnectionEvent(window, 'negotiationneeded', e => {\n const pc = e.target;\n if (browserDetails.version < 72 || (pc.getConfiguration &&\n pc.getConfiguration().sdpSemantics === 'plan-b')) {\n if (pc.signalingState !== 'stable') {\n return;\n }\n }\n return e;\n });\n}\n","/*\n * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nexport function filterIceServers(iceServers, edgeVersion) {\n let hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(server => {\n if (server && (server.urls || server.url)) {\n let urls = server.urls || server.url;\n if (server.url && !server.urls) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n }\n const isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(url => {\n // filter STUN unconditionally.\n if (url.indexOf('stun:') === 0) {\n return false;\n }\n\n const validTurn = url.startsWith('turn') &&\n !url.startsWith('turn:[') &&\n url.includes('transport=udp');\n if (validTurn && !hasTurn) {\n hasTurn = true;\n return true;\n }\n return validTurn && !hasTurn;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n","/* eslint-env node */\n'use strict';\n\n// SDP helpers.\nvar SDPUtils = {};\n\n// Generate an alphanumeric identifier for cname or mids.\n// TODO: use UUIDs instead? https://gist.github.com/jed/982883\nSDPUtils.generateIdentifier = function() {\n return Math.random().toString(36).substr(2, 10);\n};\n\n// The RTCP CNAME used by all peerconnections from the same JS.\nSDPUtils.localCName = SDPUtils.generateIdentifier();\n\n// Splits SDP into lines, dealing with both CRLF and LF.\nSDPUtils.splitLines = function(blob) {\n return blob.trim().split('\\n').map(function(line) {\n return line.trim();\n });\n};\n// Splits SDP into sessionpart and mediasections. Ensures CRLF.\nSDPUtils.splitSections = function(blob) {\n var parts = blob.split('\\nm=');\n return parts.map(function(part, index) {\n return (index > 0 ? 'm=' + part : part).trim() + '\\r\\n';\n });\n};\n\n// returns the session description.\nSDPUtils.getDescription = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n return sections && sections[0];\n};\n\n// returns the individual media sections.\nSDPUtils.getMediaSections = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n sections.shift();\n return sections;\n};\n\n// Returns lines that start with a certain prefix.\nSDPUtils.matchPrefix = function(blob, prefix) {\n return SDPUtils.splitLines(blob).filter(function(line) {\n return line.indexOf(prefix) === 0;\n });\n};\n\n// Parses an ICE candidate line. Sample input:\n// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8\n// rport 55996\"\nSDPUtils.parseCandidate = function(line) {\n var parts;\n // Parse both variants.\n if (line.indexOf('a=candidate:') === 0) {\n parts = line.substring(12).split(' ');\n } else {\n parts = line.substring(10).split(' ');\n }\n\n var candidate = {\n foundation: parts[0],\n component: parseInt(parts[1], 10),\n protocol: parts[2].toLowerCase(),\n priority: parseInt(parts[3], 10),\n ip: parts[4],\n address: parts[4], // address is an alias for ip.\n port: parseInt(parts[5], 10),\n // skip parts[6] == 'typ'\n type: parts[7]\n };\n\n for (var i = 8; i < parts.length; i += 2) {\n switch (parts[i]) {\n case 'raddr':\n candidate.relatedAddress = parts[i + 1];\n break;\n case 'rport':\n candidate.relatedPort = parseInt(parts[i + 1], 10);\n break;\n case 'tcptype':\n candidate.tcpType = parts[i + 1];\n break;\n case 'ufrag':\n candidate.ufrag = parts[i + 1]; // for backward compability.\n candidate.usernameFragment = parts[i + 1];\n break;\n default: // extension handling, in particular ufrag\n candidate[parts[i]] = parts[i + 1];\n break;\n }\n }\n return candidate;\n};\n\n// Translates a candidate object into SDP candidate attribute.\nSDPUtils.writeCandidate = function(candidate) {\n var sdp = [];\n sdp.push(candidate.foundation);\n sdp.push(candidate.component);\n sdp.push(candidate.protocol.toUpperCase());\n sdp.push(candidate.priority);\n sdp.push(candidate.address || candidate.ip);\n sdp.push(candidate.port);\n\n var type = candidate.type;\n sdp.push('typ');\n sdp.push(type);\n if (type !== 'host' && candidate.relatedAddress &&\n candidate.relatedPort) {\n sdp.push('raddr');\n sdp.push(candidate.relatedAddress);\n sdp.push('rport');\n sdp.push(candidate.relatedPort);\n }\n if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') {\n sdp.push('tcptype');\n sdp.push(candidate.tcpType);\n }\n if (candidate.usernameFragment || candidate.ufrag) {\n sdp.push('ufrag');\n sdp.push(candidate.usernameFragment || candidate.ufrag);\n }\n return 'candidate:' + sdp.join(' ');\n};\n\n// Parses an ice-options line, returns an array of option tags.\n// a=ice-options:foo bar\nSDPUtils.parseIceOptions = function(line) {\n return line.substr(14).split(' ');\n};\n\n// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:\n// a=rtpmap:111 opus/48000/2\nSDPUtils.parseRtpMap = function(line) {\n var parts = line.substr(9).split(' ');\n var parsed = {\n payloadType: parseInt(parts.shift(), 10) // was: id\n };\n\n parts = parts[0].split('/');\n\n parsed.name = parts[0];\n parsed.clockRate = parseInt(parts[1], 10); // was: clockrate\n parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;\n // legacy alias, got renamed back to channels in ORTC.\n parsed.numChannels = parsed.channels;\n return parsed;\n};\n\n// Generate an a=rtpmap line from RTCRtpCodecCapability or\n// RTCRtpCodecParameters.\nSDPUtils.writeRtpMap = function(codec) {\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n var channels = codec.channels || codec.numChannels || 1;\n return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate +\n (channels !== 1 ? '/' + channels : '') + '\\r\\n';\n};\n\n// Parses an a=extmap line (headerextension from RFC 5285). Sample input:\n// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset\nSDPUtils.parseExtmap = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n id: parseInt(parts[0], 10),\n direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv',\n uri: parts[1]\n };\n};\n\n// Generates a=extmap line from RTCRtpHeaderExtensionParameters or\n// RTCRtpHeaderExtension.\nSDPUtils.writeExtmap = function(headerExtension) {\n return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) +\n (headerExtension.direction && headerExtension.direction !== 'sendrecv'\n ? '/' + headerExtension.direction\n : '') +\n ' ' + headerExtension.uri + '\\r\\n';\n};\n\n// Parses an ftmp line, returns dictionary. Sample input:\n// a=fmtp:96 vbr=on;cng=on\n// Also deals with vbr=on; cng=on\nSDPUtils.parseFmtp = function(line) {\n var parsed = {};\n var kv;\n var parts = line.substr(line.indexOf(' ') + 1).split(';');\n for (var j = 0; j < parts.length; j++) {\n kv = parts[j].trim().split('=');\n parsed[kv[0].trim()] = kv[1];\n }\n return parsed;\n};\n\n// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeFmtp = function(codec) {\n var line = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.parameters && Object.keys(codec.parameters).length) {\n var params = [];\n Object.keys(codec.parameters).forEach(function(param) {\n if (codec.parameters[param]) {\n params.push(param + '=' + codec.parameters[param]);\n } else {\n params.push(param);\n }\n });\n line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\\r\\n';\n }\n return line;\n};\n\n// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:\n// a=rtcp-fb:98 nack rpsi\nSDPUtils.parseRtcpFb = function(line) {\n var parts = line.substr(line.indexOf(' ') + 1).split(' ');\n return {\n type: parts.shift(),\n parameter: parts.join(' ')\n };\n};\n// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeRtcpFb = function(codec) {\n var lines = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.rtcpFeedback && codec.rtcpFeedback.length) {\n // FIXME: special handling for trr-int?\n codec.rtcpFeedback.forEach(function(fb) {\n lines += 'a=rtcp-fb:' + pt + ' ' + fb.type +\n (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') +\n '\\r\\n';\n });\n }\n return lines;\n};\n\n// Parses an RFC 5576 ssrc media attribute. Sample input:\n// a=ssrc:3735928559 cname:something\nSDPUtils.parseSsrcMedia = function(line) {\n var sp = line.indexOf(' ');\n var parts = {\n ssrc: parseInt(line.substr(7, sp - 7), 10)\n };\n var colon = line.indexOf(':', sp);\n if (colon > -1) {\n parts.attribute = line.substr(sp + 1, colon - sp - 1);\n parts.value = line.substr(colon + 1);\n } else {\n parts.attribute = line.substr(sp + 1);\n }\n return parts;\n};\n\nSDPUtils.parseSsrcGroup = function(line) {\n var parts = line.substr(13).split(' ');\n return {\n semantics: parts.shift(),\n ssrcs: parts.map(function(ssrc) {\n return parseInt(ssrc, 10);\n })\n };\n};\n\n// Extracts the MID (RFC 5888) from a media section.\n// returns the MID or undefined if no mid line was found.\nSDPUtils.getMid = function(mediaSection) {\n var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0];\n if (mid) {\n return mid.substr(6);\n }\n};\n\nSDPUtils.parseFingerprint = function(line) {\n var parts = line.substr(14).split(' ');\n return {\n algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.\n value: parts[1]\n };\n};\n\n// Extracts DTLS parameters from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the fingerprint line as input. See also getIceParameters.\nSDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=fingerprint:');\n // Note: a=setup line is ignored since we use the 'auto' role.\n // Note2: 'algorithm' is not case sensitive except in Edge.\n return {\n role: 'auto',\n fingerprints: lines.map(SDPUtils.parseFingerprint)\n };\n};\n\n// Serializes DTLS parameters to SDP.\nSDPUtils.writeDtlsParameters = function(params, setupType) {\n var sdp = 'a=setup:' + setupType + '\\r\\n';\n params.fingerprints.forEach(function(fp) {\n sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\\r\\n';\n });\n return sdp;\n};\n\n// Parses a=crypto lines into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members\nSDPUtils.parseCryptoLine = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n tag: parseInt(parts[0], 10),\n cryptoSuite: parts[1],\n keyParams: parts[2],\n sessionParams: parts.slice(3),\n };\n};\n\nSDPUtils.writeCryptoLine = function(parameters) {\n return 'a=crypto:' + parameters.tag + ' ' +\n parameters.cryptoSuite + ' ' +\n (typeof parameters.keyParams === 'object'\n ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)\n : parameters.keyParams) +\n (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') +\n '\\r\\n';\n};\n\n// Parses the crypto key parameters into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*\nSDPUtils.parseCryptoKeyParams = function(keyParams) {\n if (keyParams.indexOf('inline:') !== 0) {\n return null;\n }\n var parts = keyParams.substr(7).split('|');\n return {\n keyMethod: 'inline',\n keySalt: parts[0],\n lifeTime: parts[1],\n mkiValue: parts[2] ? parts[2].split(':')[0] : undefined,\n mkiLength: parts[2] ? parts[2].split(':')[1] : undefined,\n };\n};\n\nSDPUtils.writeCryptoKeyParams = function(keyParams) {\n return keyParams.keyMethod + ':'\n + keyParams.keySalt +\n (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') +\n (keyParams.mkiValue && keyParams.mkiLength\n ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength\n : '');\n};\n\n// Extracts all SDES paramters.\nSDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=crypto:');\n return lines.map(SDPUtils.parseCryptoLine);\n};\n\n// Parses ICE information from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the ice-ufrag and ice-pwd lines as input.\nSDPUtils.getIceParameters = function(mediaSection, sessionpart) {\n var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-ufrag:')[0];\n var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-pwd:')[0];\n if (!(ufrag && pwd)) {\n return null;\n }\n return {\n usernameFragment: ufrag.substr(12),\n password: pwd.substr(10),\n };\n};\n\n// Serializes ICE parameters to SDP.\nSDPUtils.writeIceParameters = function(params) {\n return 'a=ice-ufrag:' + params.usernameFragment + '\\r\\n' +\n 'a=ice-pwd:' + params.password + '\\r\\n';\n};\n\n// Parses the SDP media section and returns RTCRtpParameters.\nSDPUtils.parseRtpParameters = function(mediaSection) {\n var description = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: [],\n rtcp: []\n };\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..]\n var pt = mline[i];\n var rtpmapline = SDPUtils.matchPrefix(\n mediaSection, 'a=rtpmap:' + pt + ' ')[0];\n if (rtpmapline) {\n var codec = SDPUtils.parseRtpMap(rtpmapline);\n var fmtps = SDPUtils.matchPrefix(\n mediaSection, 'a=fmtp:' + pt + ' ');\n // Only the first a=fmtp: is considered.\n codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};\n codec.rtcpFeedback = SDPUtils.matchPrefix(\n mediaSection, 'a=rtcp-fb:' + pt + ' ')\n .map(SDPUtils.parseRtcpFb);\n description.codecs.push(codec);\n // parse FEC mechanisms from rtpmap lines.\n switch (codec.name.toUpperCase()) {\n case 'RED':\n case 'ULPFEC':\n description.fecMechanisms.push(codec.name.toUpperCase());\n break;\n default: // only RED and ULPFEC are recognized as FEC mechanisms.\n break;\n }\n }\n }\n SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) {\n description.headerExtensions.push(SDPUtils.parseExtmap(line));\n });\n // FIXME: parse rtcp.\n return description;\n};\n\n// Generates parts of the SDP media section describing the capabilities /\n// parameters.\nSDPUtils.writeRtpDescription = function(kind, caps) {\n var sdp = '';\n\n // Build the mline.\n sdp += 'm=' + kind + ' ';\n sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs.\n sdp += ' UDP/TLS/RTP/SAVPF ';\n sdp += caps.codecs.map(function(codec) {\n if (codec.preferredPayloadType !== undefined) {\n return codec.preferredPayloadType;\n }\n return codec.payloadType;\n }).join(' ') + '\\r\\n';\n\n sdp += 'c=IN IP4 0.0.0.0\\r\\n';\n sdp += 'a=rtcp:9 IN IP4 0.0.0.0\\r\\n';\n\n // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.\n caps.codecs.forEach(function(codec) {\n sdp += SDPUtils.writeRtpMap(codec);\n sdp += SDPUtils.writeFmtp(codec);\n sdp += SDPUtils.writeRtcpFb(codec);\n });\n var maxptime = 0;\n caps.codecs.forEach(function(codec) {\n if (codec.maxptime > maxptime) {\n maxptime = codec.maxptime;\n }\n });\n if (maxptime > 0) {\n sdp += 'a=maxptime:' + maxptime + '\\r\\n';\n }\n sdp += 'a=rtcp-mux\\r\\n';\n\n if (caps.headerExtensions) {\n caps.headerExtensions.forEach(function(extension) {\n sdp += SDPUtils.writeExtmap(extension);\n });\n }\n // FIXME: write fecMechanisms.\n return sdp;\n};\n\n// Parses the SDP media section and returns an array of\n// RTCRtpEncodingParameters.\nSDPUtils.parseRtpEncodingParameters = function(mediaSection) {\n var encodingParameters = [];\n var description = SDPUtils.parseRtpParameters(mediaSection);\n var hasRed = description.fecMechanisms.indexOf('RED') !== -1;\n var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1;\n\n // filter a=ssrc:... cname:, ignore PlanB-msid\n var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(parts) {\n return parts.attribute === 'cname';\n });\n var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;\n var secondarySsrc;\n\n var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID')\n .map(function(line) {\n var parts = line.substr(17).split(' ');\n return parts.map(function(part) {\n return parseInt(part, 10);\n });\n });\n if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {\n secondarySsrc = flows[0][1];\n }\n\n description.codecs.forEach(function(codec) {\n if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) {\n var encParam = {\n ssrc: primarySsrc,\n codecPayloadType: parseInt(codec.parameters.apt, 10)\n };\n if (primarySsrc && secondarySsrc) {\n encParam.rtx = {ssrc: secondarySsrc};\n }\n encodingParameters.push(encParam);\n if (hasRed) {\n encParam = JSON.parse(JSON.stringify(encParam));\n encParam.fec = {\n ssrc: primarySsrc,\n mechanism: hasUlpfec ? 'red+ulpfec' : 'red'\n };\n encodingParameters.push(encParam);\n }\n }\n });\n if (encodingParameters.length === 0 && primarySsrc) {\n encodingParameters.push({\n ssrc: primarySsrc\n });\n }\n\n // we support both b=AS and b=TIAS but interpret AS as TIAS.\n var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b=');\n if (bandwidth.length) {\n if (bandwidth[0].indexOf('b=TIAS:') === 0) {\n bandwidth = parseInt(bandwidth[0].substr(7), 10);\n } else if (bandwidth[0].indexOf('b=AS:') === 0) {\n // use formula from JSEP to convert b=AS to TIAS value.\n bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95\n - (50 * 40 * 8);\n } else {\n bandwidth = undefined;\n }\n encodingParameters.forEach(function(params) {\n params.maxBitrate = bandwidth;\n });\n }\n return encodingParameters;\n};\n\n// parses http://draft.ortc.org/#rtcrtcpparameters*\nSDPUtils.parseRtcpParameters = function(mediaSection) {\n var rtcpParameters = {};\n\n // Gets the first SSRC. Note tha with RTX there might be multiple\n // SSRCs.\n var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(obj) {\n return obj.attribute === 'cname';\n })[0];\n if (remoteSsrc) {\n rtcpParameters.cname = remoteSsrc.value;\n rtcpParameters.ssrc = remoteSsrc.ssrc;\n }\n\n // Edge uses the compound attribute instead of reducedSize\n // compound is !reducedSize\n var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize');\n rtcpParameters.reducedSize = rsize.length > 0;\n rtcpParameters.compound = rsize.length === 0;\n\n // parses the rtcp-mux attrіbute.\n // Note that Edge does not support unmuxed RTCP.\n var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux');\n rtcpParameters.mux = mux.length > 0;\n\n return rtcpParameters;\n};\n\n// parses either a=msid: or a=ssrc:... msid lines and returns\n// the id of the MediaStream and MediaStreamTrack.\nSDPUtils.parseMsid = function(mediaSection) {\n var parts;\n var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:');\n if (spec.length === 1) {\n parts = spec[0].substr(7).split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(msidParts) {\n return msidParts.attribute === 'msid';\n });\n if (planB.length > 0) {\n parts = planB[0].value.split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n};\n\n// SCTP\n// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back\n// to draft-ietf-mmusic-sctp-sdp-05\nSDPUtils.parseSctpDescription = function(mediaSection) {\n var mline = SDPUtils.parseMLine(mediaSection);\n var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:');\n var maxMessageSize;\n if (maxSizeLine.length > 0) {\n maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);\n }\n if (isNaN(maxMessageSize)) {\n maxMessageSize = 65536;\n }\n var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:');\n if (sctpPort.length > 0) {\n return {\n port: parseInt(sctpPort[0].substr(12), 10),\n protocol: mline.fmt,\n maxMessageSize: maxMessageSize\n };\n }\n var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:');\n if (sctpMapLines.length > 0) {\n var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0]\n .substr(10)\n .split(' ');\n return {\n port: parseInt(parts[0], 10),\n protocol: parts[1],\n maxMessageSize: maxMessageSize\n };\n }\n};\n\n// SCTP\n// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers\n// support by now receiving in this format, unless we originally parsed\n// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line\n// protocol of DTLS/SCTP -- without UDP/ or TCP/)\nSDPUtils.writeSctpDescription = function(media, sctp) {\n var output = [];\n if (media.protocol !== 'DTLS/SCTP') {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctp-port:' + sctp.port + '\\r\\n'\n ];\n } else {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\\r\\n'\n ];\n }\n if (sctp.maxMessageSize !== undefined) {\n output.push('a=max-message-size:' + sctp.maxMessageSize + '\\r\\n');\n }\n return output.join('');\n};\n\n// Generate a session ID for SDP.\n// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1\n// recommends using a cryptographically random +ve 64-bit value\n// but right now this should be acceptable and within the right range\nSDPUtils.generateSessionId = function() {\n return Math.random().toString().substr(2, 21);\n};\n\n// Write boilder plate for start of SDP\n// sessId argument is optional - if not supplied it will\n// be generated randomly\n// sessVersion is optional and defaults to 2\n// sessUser is optional and defaults to 'thisisadapterortc'\nSDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {\n var sessionId;\n var version = sessVer !== undefined ? sessVer : 2;\n if (sessId) {\n sessionId = sessId;\n } else {\n sessionId = SDPUtils.generateSessionId();\n }\n var user = sessUser || 'thisisadapterortc';\n // FIXME: sess-id should be an NTP timestamp.\n return 'v=0\\r\\n' +\n 'o=' + user + ' ' + sessionId + ' ' + version +\n ' IN IP4 127.0.0.1\\r\\n' +\n 's=-\\r\\n' +\n 't=0 0\\r\\n';\n};\n\nSDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.direction) {\n sdp += 'a=' + transceiver.direction + '\\r\\n';\n } else if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n // spec.\n var msid = 'msid:' + stream.id + ' ' +\n transceiver.rtpSender.track.id + '\\r\\n';\n sdp += 'a=' + msid;\n\n // for Chrome.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n};\n\n// Gets the direction from the mediaSection or the sessionpart.\nSDPUtils.getDirection = function(mediaSection, sessionpart) {\n // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.\n var lines = SDPUtils.splitLines(mediaSection);\n for (var i = 0; i < lines.length; i++) {\n switch (lines[i]) {\n case 'a=sendrecv':\n case 'a=sendonly':\n case 'a=recvonly':\n case 'a=inactive':\n return lines[i].substr(2);\n default:\n // FIXME: What should happen here?\n }\n }\n if (sessionpart) {\n return SDPUtils.getDirection(sessionpart);\n }\n return 'sendrecv';\n};\n\nSDPUtils.getKind = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n return mline[0].substr(2);\n};\n\nSDPUtils.isRejected = function(mediaSection) {\n return mediaSection.split(' ', 2)[1] === '0';\n};\n\nSDPUtils.parseMLine = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var parts = lines[0].substr(2).split(' ');\n return {\n kind: parts[0],\n port: parseInt(parts[1], 10),\n protocol: parts[2],\n fmt: parts.slice(3).join(' ')\n };\n};\n\nSDPUtils.parseOLine = function(mediaSection) {\n var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0];\n var parts = line.substr(2).split(' ');\n return {\n username: parts[0],\n sessionId: parts[1],\n sessionVersion: parseInt(parts[2], 10),\n netType: parts[3],\n addressType: parts[4],\n address: parts[5]\n };\n};\n\n// a very naive interpretation of a valid SDP.\nSDPUtils.isValidSDP = function(blob) {\n if (typeof blob !== 'string' || blob.length === 0) {\n return false;\n }\n var lines = SDPUtils.splitLines(blob);\n for (var i = 0; i < lines.length; i++) {\n if (lines[i].length < 2 || lines[i].charAt(1) !== '=') {\n return false;\n }\n // TODO: check the modifier a bit more.\n }\n return true;\n};\n\n// Expose public methods.\nif (typeof module === 'object') {\n module.exports = SDPUtils;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nvar SDPUtils = require('sdp');\n\nfunction fixStatsType(stat) {\n return {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[stat.type] || stat.type;\n}\n\nfunction writeMediaSection(transceiver, caps, type, stream, dtlsRole) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : dtlsRole || 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n var trackId = transceiver.rtpSender._initialTrackId ||\n transceiver.rtpSender.track.id;\n transceiver.rtpSender._initialTrackId = trackId;\n // spec.\n var msid = 'msid:' + (stream ? stream.id : '-') + ' ' +\n trackId + '\\r\\n';\n sdp += 'a=' + msid;\n // for Chrome. Legacy should no longer be required.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n\n // RTX\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n}\n\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nfunction filterIceServers(iceServers, edgeVersion) {\n var hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(function(server) {\n if (server && (server.urls || server.url)) {\n var urls = server.urls || server.url;\n if (server.url && !server.urls) {\n console.warn('RTCIceServer.url is deprecated! Use urls instead.');\n }\n var isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(function(url) {\n var validTurn = url.indexOf('turn:') === 0 &&\n url.indexOf('transport=udp') !== -1 &&\n url.indexOf('turn:[') === -1 &&\n !hasTurn;\n\n if (validTurn) {\n hasTurn = true;\n return true;\n }\n return url.indexOf('stun:') === 0 && edgeVersion >= 14393 &&\n url.indexOf('?transport=udp') === -1;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n\n// Determines the intersection of local and remote capabilities.\nfunction getCommonCapabilities(localCapabilities, remoteCapabilities) {\n var commonCapabilities = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: []\n };\n\n var findCodecByPayloadType = function(pt, codecs) {\n pt = parseInt(pt, 10);\n for (var i = 0; i < codecs.length; i++) {\n if (codecs[i].payloadType === pt ||\n codecs[i].preferredPayloadType === pt) {\n return codecs[i];\n }\n }\n };\n\n var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) {\n var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs);\n var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs);\n return lCodec && rCodec &&\n lCodec.name.toLowerCase() === rCodec.name.toLowerCase();\n };\n\n localCapabilities.codecs.forEach(function(lCodec) {\n for (var i = 0; i < remoteCapabilities.codecs.length; i++) {\n var rCodec = remoteCapabilities.codecs[i];\n if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() &&\n lCodec.clockRate === rCodec.clockRate) {\n if (lCodec.name.toLowerCase() === 'rtx' &&\n lCodec.parameters && rCodec.parameters.apt) {\n // for RTX we need to find the local rtx that has a apt\n // which points to the same local codec as the remote one.\n if (!rtxCapabilityMatches(lCodec, rCodec,\n localCapabilities.codecs, remoteCapabilities.codecs)) {\n continue;\n }\n }\n rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy\n // number of channels is the highest common number of channels\n rCodec.numChannels = Math.min(lCodec.numChannels,\n rCodec.numChannels);\n // push rCodec so we reply with offerer payload type\n commonCapabilities.codecs.push(rCodec);\n\n // determine common feedback mechanisms\n rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) {\n for (var j = 0; j < lCodec.rtcpFeedback.length; j++) {\n if (lCodec.rtcpFeedback[j].type === fb.type &&\n lCodec.rtcpFeedback[j].parameter === fb.parameter) {\n return true;\n }\n }\n return false;\n });\n // FIXME: also need to determine .parameters\n // see https://github.com/openpeer/ortc/issues/569\n break;\n }\n }\n });\n\n localCapabilities.headerExtensions.forEach(function(lHeaderExtension) {\n for (var i = 0; i < remoteCapabilities.headerExtensions.length;\n i++) {\n var rHeaderExtension = remoteCapabilities.headerExtensions[i];\n if (lHeaderExtension.uri === rHeaderExtension.uri) {\n commonCapabilities.headerExtensions.push(rHeaderExtension);\n break;\n }\n }\n });\n\n // FIXME: fecMechanisms\n return commonCapabilities;\n}\n\n// is action=setLocalDescription with type allowed in signalingState\nfunction isActionAllowedInSignalingState(action, type, signalingState) {\n return {\n offer: {\n setLocalDescription: ['stable', 'have-local-offer'],\n setRemoteDescription: ['stable', 'have-remote-offer']\n },\n answer: {\n setLocalDescription: ['have-remote-offer', 'have-local-pranswer'],\n setRemoteDescription: ['have-local-offer', 'have-remote-pranswer']\n }\n }[type][action].indexOf(signalingState) !== -1;\n}\n\nfunction maybeAddCandidate(iceTransport, candidate) {\n // Edge's internal representation adds some fields therefore\n // not all fieldѕ are taken into account.\n var alreadyAdded = iceTransport.getRemoteCandidates()\n .find(function(remoteCandidate) {\n return candidate.foundation === remoteCandidate.foundation &&\n candidate.ip === remoteCandidate.ip &&\n candidate.port === remoteCandidate.port &&\n candidate.priority === remoteCandidate.priority &&\n candidate.protocol === remoteCandidate.protocol &&\n candidate.type === remoteCandidate.type;\n });\n if (!alreadyAdded) {\n iceTransport.addRemoteCandidate(candidate);\n }\n return !alreadyAdded;\n}\n\n\nfunction makeError(name, description) {\n var e = new Error(description);\n e.name = name;\n // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names\n e.code = {\n NotSupportedError: 9,\n InvalidStateError: 11,\n InvalidAccessError: 15,\n TypeError: undefined,\n OperationError: undefined\n }[name];\n return e;\n}\n\nmodule.exports = function(window, edgeVersion) {\n // https://w3c.github.io/mediacapture-main/#mediastream\n // Helper function to add the track to the stream and\n // dispatch the event ourselves.\n function addTrackToStreamAndFireEvent(track, stream) {\n stream.addTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack',\n {track: track}));\n }\n\n function removeTrackFromStreamAndFireEvent(track, stream) {\n stream.removeTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack',\n {track: track}));\n }\n\n function fireAddTrack(pc, track, receiver, streams) {\n var trackEvent = new Event('track');\n trackEvent.track = track;\n trackEvent.receiver = receiver;\n trackEvent.transceiver = {receiver: receiver};\n trackEvent.streams = streams;\n window.setTimeout(function() {\n pc._dispatchEvent('track', trackEvent);\n });\n }\n\n var RTCPeerConnection = function(config) {\n var pc = this;\n\n var _eventTarget = document.createDocumentFragment();\n ['addEventListener', 'removeEventListener', 'dispatchEvent']\n .forEach(function(method) {\n pc[method] = _eventTarget[method].bind(_eventTarget);\n });\n\n this.canTrickleIceCandidates = null;\n\n this.needNegotiation = false;\n\n this.localStreams = [];\n this.remoteStreams = [];\n\n this._localDescription = null;\n this._remoteDescription = null;\n\n this.signalingState = 'stable';\n this.iceConnectionState = 'new';\n this.connectionState = 'new';\n this.iceGatheringState = 'new';\n\n config = JSON.parse(JSON.stringify(config || {}));\n\n this.usingBundle = config.bundlePolicy === 'max-bundle';\n if (config.rtcpMuxPolicy === 'negotiate') {\n throw(makeError('NotSupportedError',\n 'rtcpMuxPolicy \\'negotiate\\' is not supported'));\n } else if (!config.rtcpMuxPolicy) {\n config.rtcpMuxPolicy = 'require';\n }\n\n switch (config.iceTransportPolicy) {\n case 'all':\n case 'relay':\n break;\n default:\n config.iceTransportPolicy = 'all';\n break;\n }\n\n switch (config.bundlePolicy) {\n case 'balanced':\n case 'max-compat':\n case 'max-bundle':\n break;\n default:\n config.bundlePolicy = 'balanced';\n break;\n }\n\n config.iceServers = filterIceServers(config.iceServers || [], edgeVersion);\n\n this._iceGatherers = [];\n if (config.iceCandidatePoolSize) {\n for (var i = config.iceCandidatePoolSize; i > 0; i--) {\n this._iceGatherers.push(new window.RTCIceGatherer({\n iceServers: config.iceServers,\n gatherPolicy: config.iceTransportPolicy\n }));\n }\n } else {\n config.iceCandidatePoolSize = 0;\n }\n\n this._config = config;\n\n // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ...\n // everything that is needed to describe a SDP m-line.\n this.transceivers = [];\n\n this._sdpSessionId = SDPUtils.generateSessionId();\n this._sdpSessionVersion = 0;\n\n this._dtlsRole = undefined; // role for a=setup to use in answers.\n\n this._isClosed = false;\n };\n\n Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', {\n configurable: true,\n get: function() {\n return this._localDescription;\n }\n });\n Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', {\n configurable: true,\n get: function() {\n return this._remoteDescription;\n }\n });\n\n // set up event handlers on prototype\n RTCPeerConnection.prototype.onicecandidate = null;\n RTCPeerConnection.prototype.onaddstream = null;\n RTCPeerConnection.prototype.ontrack = null;\n RTCPeerConnection.prototype.onremovestream = null;\n RTCPeerConnection.prototype.onsignalingstatechange = null;\n RTCPeerConnection.prototype.oniceconnectionstatechange = null;\n RTCPeerConnection.prototype.onconnectionstatechange = null;\n RTCPeerConnection.prototype.onicegatheringstatechange = null;\n RTCPeerConnection.prototype.onnegotiationneeded = null;\n RTCPeerConnection.prototype.ondatachannel = null;\n\n RTCPeerConnection.prototype._dispatchEvent = function(name, event) {\n if (this._isClosed) {\n return;\n }\n this.dispatchEvent(event);\n if (typeof this['on' + name] === 'function') {\n this['on' + name](event);\n }\n };\n\n RTCPeerConnection.prototype._emitGatheringStateChange = function() {\n var event = new Event('icegatheringstatechange');\n this._dispatchEvent('icegatheringstatechange', event);\n };\n\n RTCPeerConnection.prototype.getConfiguration = function() {\n return this._config;\n };\n\n RTCPeerConnection.prototype.getLocalStreams = function() {\n return this.localStreams;\n };\n\n RTCPeerConnection.prototype.getRemoteStreams = function() {\n return this.remoteStreams;\n };\n\n // internal helper to create a transceiver object.\n // (which is not yet the same as the WebRTC 1.0 transceiver)\n RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) {\n var hasBundleTransport = this.transceivers.length > 0;\n var transceiver = {\n track: null,\n iceGatherer: null,\n iceTransport: null,\n dtlsTransport: null,\n localCapabilities: null,\n remoteCapabilities: null,\n rtpSender: null,\n rtpReceiver: null,\n kind: kind,\n mid: null,\n sendEncodingParameters: null,\n recvEncodingParameters: null,\n stream: null,\n associatedRemoteMediaStreams: [],\n wantReceive: true\n };\n if (this.usingBundle && hasBundleTransport) {\n transceiver.iceTransport = this.transceivers[0].iceTransport;\n transceiver.dtlsTransport = this.transceivers[0].dtlsTransport;\n } else {\n var transports = this._createIceAndDtlsTransports();\n transceiver.iceTransport = transports.iceTransport;\n transceiver.dtlsTransport = transports.dtlsTransport;\n }\n if (!doNotAdd) {\n this.transceivers.push(transceiver);\n }\n return transceiver;\n };\n\n RTCPeerConnection.prototype.addTrack = function(track, stream) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call addTrack on a closed peerconnection.');\n }\n\n var alreadyExists = this.transceivers.find(function(s) {\n return s.track === track;\n });\n\n if (alreadyExists) {\n throw makeError('InvalidAccessError', 'Track already exists.');\n }\n\n var transceiver;\n for (var i = 0; i < this.transceivers.length; i++) {\n if (!this.transceivers[i].track &&\n this.transceivers[i].kind === track.kind) {\n transceiver = this.transceivers[i];\n }\n }\n if (!transceiver) {\n transceiver = this._createTransceiver(track.kind);\n }\n\n this._maybeFireNegotiationNeeded();\n\n if (this.localStreams.indexOf(stream) === -1) {\n this.localStreams.push(stream);\n }\n\n transceiver.track = track;\n transceiver.stream = stream;\n transceiver.rtpSender = new window.RTCRtpSender(track,\n transceiver.dtlsTransport);\n return transceiver.rtpSender;\n };\n\n RTCPeerConnection.prototype.addStream = function(stream) {\n var pc = this;\n if (edgeVersion >= 15025) {\n stream.getTracks().forEach(function(track) {\n pc.addTrack(track, stream);\n });\n } else {\n // Clone is necessary for local demos mostly, attaching directly\n // to two different senders does not work (build 10547).\n // Fixed in 15025 (or earlier)\n var clonedStream = stream.clone();\n stream.getTracks().forEach(function(track, idx) {\n var clonedTrack = clonedStream.getTracks()[idx];\n track.addEventListener('enabled', function(event) {\n clonedTrack.enabled = event.enabled;\n });\n });\n clonedStream.getTracks().forEach(function(track) {\n pc.addTrack(track, clonedStream);\n });\n }\n };\n\n RTCPeerConnection.prototype.removeTrack = function(sender) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call removeTrack on a closed peerconnection.');\n }\n\n if (!(sender instanceof window.RTCRtpSender)) {\n throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.');\n }\n\n var transceiver = this.transceivers.find(function(t) {\n return t.rtpSender === sender;\n });\n\n if (!transceiver) {\n throw makeError('InvalidAccessError',\n 'Sender was not created by this connection.');\n }\n var stream = transceiver.stream;\n\n transceiver.rtpSender.stop();\n transceiver.rtpSender = null;\n transceiver.track = null;\n transceiver.stream = null;\n\n // remove the stream from the set of local streams\n var localStreams = this.transceivers.map(function(t) {\n return t.stream;\n });\n if (localStreams.indexOf(stream) === -1 &&\n this.localStreams.indexOf(stream) > -1) {\n this.localStreams.splice(this.localStreams.indexOf(stream), 1);\n }\n\n this._maybeFireNegotiationNeeded();\n };\n\n RTCPeerConnection.prototype.removeStream = function(stream) {\n var pc = this;\n stream.getTracks().forEach(function(track) {\n var sender = pc.getSenders().find(function(s) {\n return s.track === track;\n });\n if (sender) {\n pc.removeTrack(sender);\n }\n });\n };\n\n RTCPeerConnection.prototype.getSenders = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpSender;\n })\n .map(function(transceiver) {\n return transceiver.rtpSender;\n });\n };\n\n RTCPeerConnection.prototype.getReceivers = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpReceiver;\n })\n .map(function(transceiver) {\n return transceiver.rtpReceiver;\n });\n };\n\n\n RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex,\n usingBundle) {\n var pc = this;\n if (usingBundle && sdpMLineIndex > 0) {\n return this.transceivers[0].iceGatherer;\n } else if (this._iceGatherers.length) {\n return this._iceGatherers.shift();\n }\n var iceGatherer = new window.RTCIceGatherer({\n iceServers: this._config.iceServers,\n gatherPolicy: this._config.iceTransportPolicy\n });\n Object.defineProperty(iceGatherer, 'state',\n {value: 'new', writable: true}\n );\n\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = [];\n this.transceivers[sdpMLineIndex].bufferCandidates = function(event) {\n var end = !event.candidate || Object.keys(event.candidate).length === 0;\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n iceGatherer.state = end ? 'completed' : 'gathering';\n if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) {\n pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event);\n }\n };\n iceGatherer.addEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n return iceGatherer;\n };\n\n // start gathering from an RTCIceGatherer.\n RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) {\n var pc = this;\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer.onlocalcandidate) {\n return;\n }\n var bufferedCandidateEvents =\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents;\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null;\n iceGatherer.removeEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n iceGatherer.onlocalcandidate = function(evt) {\n if (pc.usingBundle && sdpMLineIndex > 0) {\n // if we know that we use bundle we can drop candidates with\n // ѕdpMLineIndex > 0. If we don't do this then our state gets\n // confused since we dispose the extra ice gatherer.\n return;\n }\n var event = new Event('icecandidate');\n event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex};\n\n var cand = evt.candidate;\n // Edge emits an empty object for RTCIceCandidateComplete‥\n var end = !cand || Object.keys(cand).length === 0;\n if (end) {\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') {\n iceGatherer.state = 'completed';\n }\n } else {\n if (iceGatherer.state === 'new') {\n iceGatherer.state = 'gathering';\n }\n // RTCIceCandidate doesn't have a component, needs to be added\n cand.component = 1;\n // also the usernameFragment. TODO: update SDP to take both variants.\n cand.ufrag = iceGatherer.getLocalParameters().usernameFragment;\n\n var serializedCandidate = SDPUtils.writeCandidate(cand);\n event.candidate = Object.assign(event.candidate,\n SDPUtils.parseCandidate(serializedCandidate));\n\n event.candidate.candidate = serializedCandidate;\n event.candidate.toJSON = function() {\n return {\n candidate: event.candidate.candidate,\n sdpMid: event.candidate.sdpMid,\n sdpMLineIndex: event.candidate.sdpMLineIndex,\n usernameFragment: event.candidate.usernameFragment\n };\n };\n }\n\n // update local description.\n var sections = SDPUtils.getMediaSections(pc._localDescription.sdp);\n if (!end) {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=' + event.candidate.candidate + '\\r\\n';\n } else {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=end-of-candidates\\r\\n';\n }\n pc._localDescription.sdp =\n SDPUtils.getDescription(pc._localDescription.sdp) +\n sections.join('');\n var complete = pc.transceivers.every(function(transceiver) {\n return transceiver.iceGatherer &&\n transceiver.iceGatherer.state === 'completed';\n });\n\n if (pc.iceGatheringState !== 'gathering') {\n pc.iceGatheringState = 'gathering';\n pc._emitGatheringStateChange();\n }\n\n // Emit candidate. Also emit null candidate when all gatherers are\n // complete.\n if (!end) {\n pc._dispatchEvent('icecandidate', event);\n }\n if (complete) {\n pc._dispatchEvent('icecandidate', new Event('icecandidate'));\n pc.iceGatheringState = 'complete';\n pc._emitGatheringStateChange();\n }\n };\n\n // emit already gathered candidates.\n window.setTimeout(function() {\n bufferedCandidateEvents.forEach(function(e) {\n iceGatherer.onlocalcandidate(e);\n });\n }, 0);\n };\n\n // Create ICE transport and DTLS transport.\n RTCPeerConnection.prototype._createIceAndDtlsTransports = function() {\n var pc = this;\n var iceTransport = new window.RTCIceTransport(null);\n iceTransport.onicestatechange = function() {\n pc._updateIceConnectionState();\n pc._updateConnectionState();\n };\n\n var dtlsTransport = new window.RTCDtlsTransport(iceTransport);\n dtlsTransport.ondtlsstatechange = function() {\n pc._updateConnectionState();\n };\n dtlsTransport.onerror = function() {\n // onerror does not set state to failed by itself.\n Object.defineProperty(dtlsTransport, 'state',\n {value: 'failed', writable: true});\n pc._updateConnectionState();\n };\n\n return {\n iceTransport: iceTransport,\n dtlsTransport: dtlsTransport\n };\n };\n\n // Destroy ICE gatherer, ICE transport and DTLS transport.\n // Without triggering the callbacks.\n RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function(\n sdpMLineIndex) {\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer) {\n delete iceGatherer.onlocalcandidate;\n delete this.transceivers[sdpMLineIndex].iceGatherer;\n }\n var iceTransport = this.transceivers[sdpMLineIndex].iceTransport;\n if (iceTransport) {\n delete iceTransport.onicestatechange;\n delete this.transceivers[sdpMLineIndex].iceTransport;\n }\n var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport;\n if (dtlsTransport) {\n delete dtlsTransport.ondtlsstatechange;\n delete dtlsTransport.onerror;\n delete this.transceivers[sdpMLineIndex].dtlsTransport;\n }\n };\n\n // Start the RTP Sender and Receiver for a transceiver.\n RTCPeerConnection.prototype._transceive = function(transceiver,\n send, recv) {\n var params = getCommonCapabilities(transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n if (send && transceiver.rtpSender) {\n params.encodings = transceiver.sendEncodingParameters;\n params.rtcp = {\n cname: SDPUtils.localCName,\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.recvEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc;\n }\n transceiver.rtpSender.send(params);\n }\n if (recv && transceiver.rtpReceiver && params.codecs.length > 0) {\n // remove RTX field in Edge 14942\n if (transceiver.kind === 'video'\n && transceiver.recvEncodingParameters\n && edgeVersion < 15019) {\n transceiver.recvEncodingParameters.forEach(function(p) {\n delete p.rtx;\n });\n }\n if (transceiver.recvEncodingParameters.length) {\n params.encodings = transceiver.recvEncodingParameters;\n } else {\n params.encodings = [{}];\n }\n params.rtcp = {\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.rtcpParameters.cname) {\n params.rtcp.cname = transceiver.rtcpParameters.cname;\n }\n if (transceiver.sendEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc;\n }\n transceiver.rtpReceiver.receive(params);\n }\n };\n\n RTCPeerConnection.prototype.setLocalDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setLocalDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set local ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var sections;\n var sessionpart;\n if (description.type === 'offer') {\n // VERY limited support for SDP munging. Limited to:\n // * changing the order of codecs\n sections = SDPUtils.splitSections(description.sdp);\n sessionpart = sections.shift();\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var caps = SDPUtils.parseRtpParameters(mediaSection);\n pc.transceivers[sdpMLineIndex].localCapabilities = caps;\n });\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n });\n } else if (description.type === 'answer') {\n sections = SDPUtils.splitSections(pc._remoteDescription.sdp);\n sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var transceiver = pc.transceivers[sdpMLineIndex];\n var iceGatherer = transceiver.iceGatherer;\n var iceTransport = transceiver.iceTransport;\n var dtlsTransport = transceiver.dtlsTransport;\n var localCapabilities = transceiver.localCapabilities;\n var remoteCapabilities = transceiver.remoteCapabilities;\n\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n\n if (!rejected && !transceiver.rejected) {\n var remoteIceParameters = SDPUtils.getIceParameters(\n mediaSection, sessionpart);\n var remoteDtlsParameters = SDPUtils.getDtlsParameters(\n mediaSection, sessionpart);\n if (isIceLite) {\n remoteDtlsParameters.role = 'server';\n }\n\n if (!pc.usingBundle || sdpMLineIndex === 0) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n isIceLite ? 'controlling' : 'controlled');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // Calculate intersection of capabilities.\n var params = getCommonCapabilities(localCapabilities,\n remoteCapabilities);\n\n // Start the RTCRtpSender. The RTCRtpReceiver for this\n // transceiver has already been started in setRemoteDescription.\n pc._transceive(transceiver,\n params.codecs.length > 0,\n false);\n }\n });\n }\n\n pc._localDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-local-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.setRemoteDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setRemoteDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set remote ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var streams = {};\n pc.remoteStreams.forEach(function(stream) {\n streams[stream.id] = stream;\n });\n var receiverList = [];\n var sections = SDPUtils.splitSections(description.sdp);\n var sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n var usingBundle = SDPUtils.matchPrefix(sessionpart,\n 'a=group:BUNDLE ').length > 0;\n pc.usingBundle = usingBundle;\n var iceOptions = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-options:')[0];\n if (iceOptions) {\n pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ')\n .indexOf('trickle') >= 0;\n } else {\n pc.canTrickleIceCandidates = false;\n }\n\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var lines = SDPUtils.splitLines(mediaSection);\n var kind = SDPUtils.getKind(mediaSection);\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n var protocol = lines[0].substr(2).split(' ')[2];\n\n var direction = SDPUtils.getDirection(mediaSection, sessionpart);\n var remoteMsid = SDPUtils.parseMsid(mediaSection);\n\n var mid = SDPUtils.getMid(mediaSection) || SDPUtils.generateIdentifier();\n\n // Reject datachannels which are not implemented yet.\n if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' ||\n protocol === 'UDP/DTLS/SCTP'))) {\n // TODO: this is dangerous in the case where a non-rejected m-line\n // becomes rejected.\n pc.transceivers[sdpMLineIndex] = {\n mid: mid,\n kind: kind,\n protocol: protocol,\n rejected: true\n };\n return;\n }\n\n if (!rejected && pc.transceivers[sdpMLineIndex] &&\n pc.transceivers[sdpMLineIndex].rejected) {\n // recycle a rejected transceiver.\n pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true);\n }\n\n var transceiver;\n var iceGatherer;\n var iceTransport;\n var dtlsTransport;\n var rtpReceiver;\n var sendEncodingParameters;\n var recvEncodingParameters;\n var localCapabilities;\n\n var track;\n // FIXME: ensure the mediaSection has rtcp-mux set.\n var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection);\n var remoteIceParameters;\n var remoteDtlsParameters;\n if (!rejected) {\n remoteIceParameters = SDPUtils.getIceParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters.role = 'client';\n }\n recvEncodingParameters =\n SDPUtils.parseRtpEncodingParameters(mediaSection);\n\n var rtcpParameters = SDPUtils.parseRtcpParameters(mediaSection);\n\n var isComplete = SDPUtils.matchPrefix(mediaSection,\n 'a=end-of-candidates', sessionpart).length > 0;\n var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:')\n .map(function(cand) {\n return SDPUtils.parseCandidate(cand);\n })\n .filter(function(cand) {\n return cand.component === 1;\n });\n\n // Check if we can use BUNDLE and dispose transports.\n if ((description.type === 'offer' || description.type === 'answer') &&\n !rejected && usingBundle && sdpMLineIndex > 0 &&\n pc.transceivers[sdpMLineIndex]) {\n pc._disposeIceAndDtlsTransports(sdpMLineIndex);\n pc.transceivers[sdpMLineIndex].iceGatherer =\n pc.transceivers[0].iceGatherer;\n pc.transceivers[sdpMLineIndex].iceTransport =\n pc.transceivers[0].iceTransport;\n pc.transceivers[sdpMLineIndex].dtlsTransport =\n pc.transceivers[0].dtlsTransport;\n if (pc.transceivers[sdpMLineIndex].rtpSender) {\n pc.transceivers[sdpMLineIndex].rtpSender.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n if (pc.transceivers[sdpMLineIndex].rtpReceiver) {\n pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n }\n if (description.type === 'offer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex] ||\n pc._createTransceiver(kind);\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n usingBundle);\n }\n\n if (cands.length && transceiver.iceTransport.state === 'new') {\n if (isComplete && (!usingBundle || sdpMLineIndex === 0)) {\n transceiver.iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n localCapabilities = window.RTCRtpReceiver.getCapabilities(kind);\n\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n\n sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 2) * 1001\n }];\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n var isNewTrack = false;\n if (direction === 'sendrecv' || direction === 'sendonly') {\n isNewTrack = !transceiver.rtpReceiver;\n rtpReceiver = transceiver.rtpReceiver ||\n new window.RTCRtpReceiver(transceiver.dtlsTransport, kind);\n\n if (isNewTrack) {\n var stream;\n track = rtpReceiver.track;\n // FIXME: does not work with Plan B.\n if (remoteMsid && remoteMsid.stream === '-') {\n // no-op. a stream id of '-' means: no associated stream.\n } else if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n Object.defineProperty(streams[remoteMsid.stream], 'id', {\n get: function() {\n return remoteMsid.stream;\n }\n });\n }\n Object.defineProperty(track, 'id', {\n get: function() {\n return remoteMsid.track;\n }\n });\n stream = streams[remoteMsid.stream];\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n stream = streams.default;\n }\n if (stream) {\n addTrackToStreamAndFireEvent(track, stream);\n transceiver.associatedRemoteMediaStreams.push(stream);\n }\n receiverList.push([track, rtpReceiver, stream]);\n }\n } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) {\n transceiver.associatedRemoteMediaStreams.forEach(function(s) {\n var nativeTrack = s.getTracks().find(function(t) {\n return t.id === transceiver.rtpReceiver.track.id;\n });\n if (nativeTrack) {\n removeTrackFromStreamAndFireEvent(nativeTrack, s);\n }\n });\n transceiver.associatedRemoteMediaStreams = [];\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.remoteCapabilities = remoteCapabilities;\n transceiver.rtpReceiver = rtpReceiver;\n transceiver.rtcpParameters = rtcpParameters;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n transceiver.recvEncodingParameters = recvEncodingParameters;\n\n // Start the RTCRtpReceiver now. The RTPSender is started in\n // setLocalDescription.\n pc._transceive(pc.transceivers[sdpMLineIndex],\n false,\n isNewTrack);\n } else if (description.type === 'answer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex];\n iceGatherer = transceiver.iceGatherer;\n iceTransport = transceiver.iceTransport;\n dtlsTransport = transceiver.dtlsTransport;\n rtpReceiver = transceiver.rtpReceiver;\n sendEncodingParameters = transceiver.sendEncodingParameters;\n localCapabilities = transceiver.localCapabilities;\n\n pc.transceivers[sdpMLineIndex].recvEncodingParameters =\n recvEncodingParameters;\n pc.transceivers[sdpMLineIndex].remoteCapabilities =\n remoteCapabilities;\n pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters;\n\n if (cands.length && iceTransport.state === 'new') {\n if ((isIceLite || isComplete) &&\n (!usingBundle || sdpMLineIndex === 0)) {\n iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n if (!usingBundle || sdpMLineIndex === 0) {\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n 'controlling');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // If the offer contained RTX but the answer did not,\n // remove RTX from sendEncodingParameters.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n pc._transceive(transceiver,\n direction === 'sendrecv' || direction === 'recvonly',\n direction === 'sendrecv' || direction === 'sendonly');\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n if (rtpReceiver &&\n (direction === 'sendrecv' || direction === 'sendonly')) {\n track = rtpReceiver.track;\n if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]);\n receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]);\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams.default);\n receiverList.push([track, rtpReceiver, streams.default]);\n }\n } else {\n // FIXME: actually the receiver should be created later.\n delete transceiver.rtpReceiver;\n }\n }\n });\n\n if (pc._dtlsRole === undefined) {\n pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive';\n }\n\n pc._remoteDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-remote-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n Object.keys(streams).forEach(function(sid) {\n var stream = streams[sid];\n if (stream.getTracks().length) {\n if (pc.remoteStreams.indexOf(stream) === -1) {\n pc.remoteStreams.push(stream);\n var event = new Event('addstream');\n event.stream = stream;\n window.setTimeout(function() {\n pc._dispatchEvent('addstream', event);\n });\n }\n\n receiverList.forEach(function(item) {\n var track = item[0];\n var receiver = item[1];\n if (stream.id !== item[2].id) {\n return;\n }\n fireAddTrack(pc, track, receiver, [stream]);\n });\n }\n });\n receiverList.forEach(function(item) {\n if (item[2]) {\n return;\n }\n fireAddTrack(pc, item[0], item[1], []);\n });\n\n // check whether addIceCandidate({}) was called within four seconds after\n // setRemoteDescription.\n window.setTimeout(function() {\n if (!(pc && pc.transceivers)) {\n return;\n }\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport &&\n transceiver.iceTransport.state === 'new' &&\n transceiver.iceTransport.getRemoteCandidates().length > 0) {\n console.warn('Timeout for addRemoteCandidate. Consider sending ' +\n 'an end-of-candidates notification');\n transceiver.iceTransport.addRemoteCandidate({});\n }\n });\n }, 4000);\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.close = function() {\n this.transceivers.forEach(function(transceiver) {\n /* not yet\n if (transceiver.iceGatherer) {\n transceiver.iceGatherer.close();\n }\n */\n if (transceiver.iceTransport) {\n transceiver.iceTransport.stop();\n }\n if (transceiver.dtlsTransport) {\n transceiver.dtlsTransport.stop();\n }\n if (transceiver.rtpSender) {\n transceiver.rtpSender.stop();\n }\n if (transceiver.rtpReceiver) {\n transceiver.rtpReceiver.stop();\n }\n });\n // FIXME: clean up tracks, local streams, remote streams, etc\n this._isClosed = true;\n this._updateSignalingState('closed');\n };\n\n // Update the signaling state.\n RTCPeerConnection.prototype._updateSignalingState = function(newState) {\n this.signalingState = newState;\n var event = new Event('signalingstatechange');\n this._dispatchEvent('signalingstatechange', event);\n };\n\n // Determine whether to fire the negotiationneeded event.\n RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() {\n var pc = this;\n if (this.signalingState !== 'stable' || this.needNegotiation === true) {\n return;\n }\n this.needNegotiation = true;\n window.setTimeout(function() {\n if (pc.needNegotiation) {\n pc.needNegotiation = false;\n var event = new Event('negotiationneeded');\n pc._dispatchEvent('negotiationneeded', event);\n }\n }, 0);\n };\n\n // Update the ice connection state.\n RTCPeerConnection.prototype._updateIceConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n checking: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n }\n });\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.checking > 0) {\n newState = 'checking';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n } else if (states.completed > 0) {\n newState = 'completed';\n }\n\n if (newState !== this.iceConnectionState) {\n this.iceConnectionState = newState;\n var event = new Event('iceconnectionstatechange');\n this._dispatchEvent('iceconnectionstatechange', event);\n }\n };\n\n // Update the connection state.\n RTCPeerConnection.prototype._updateConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n connecting: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && transceiver.dtlsTransport &&\n !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n states[transceiver.dtlsTransport.state]++;\n }\n });\n // ICETransport.completed and connected are the same for this purpose.\n states.connected += states.completed;\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.connecting > 0) {\n newState = 'connecting';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n }\n\n if (newState !== this.connectionState) {\n this.connectionState = newState;\n var event = new Event('connectionstatechange');\n this._dispatchEvent('connectionstatechange', event);\n }\n };\n\n RTCPeerConnection.prototype.createOffer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createOffer after close'));\n }\n\n var numAudioTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'audio';\n }).length;\n var numVideoTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'video';\n }).length;\n\n // Determine number of audio and video tracks we need to send/recv.\n var offerOptions = arguments[0];\n if (offerOptions) {\n // Reject Chrome legacy constraints.\n if (offerOptions.mandatory || offerOptions.optional) {\n throw new TypeError(\n 'Legacy mandatory/optional constraints not supported.');\n }\n if (offerOptions.offerToReceiveAudio !== undefined) {\n if (offerOptions.offerToReceiveAudio === true) {\n numAudioTracks = 1;\n } else if (offerOptions.offerToReceiveAudio === false) {\n numAudioTracks = 0;\n } else {\n numAudioTracks = offerOptions.offerToReceiveAudio;\n }\n }\n if (offerOptions.offerToReceiveVideo !== undefined) {\n if (offerOptions.offerToReceiveVideo === true) {\n numVideoTracks = 1;\n } else if (offerOptions.offerToReceiveVideo === false) {\n numVideoTracks = 0;\n } else {\n numVideoTracks = offerOptions.offerToReceiveVideo;\n }\n }\n }\n\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.kind === 'audio') {\n numAudioTracks--;\n if (numAudioTracks < 0) {\n transceiver.wantReceive = false;\n }\n } else if (transceiver.kind === 'video') {\n numVideoTracks--;\n if (numVideoTracks < 0) {\n transceiver.wantReceive = false;\n }\n }\n });\n\n // Create M-lines for recvonly streams.\n while (numAudioTracks > 0 || numVideoTracks > 0) {\n if (numAudioTracks > 0) {\n pc._createTransceiver('audio');\n numAudioTracks--;\n }\n if (numVideoTracks > 0) {\n pc._createTransceiver('video');\n numVideoTracks--;\n }\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n // For each track, create an ice gatherer, ice transport,\n // dtls transport, potentially rtpsender and rtpreceiver.\n var track = transceiver.track;\n var kind = transceiver.kind;\n var mid = transceiver.mid || SDPUtils.generateIdentifier();\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n pc.usingBundle);\n }\n\n var localCapabilities = window.RTCRtpSender.getCapabilities(kind);\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n localCapabilities.codecs.forEach(function(codec) {\n // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552\n // by adding level-asymmetry-allowed=1\n if (codec.name === 'H264' &&\n codec.parameters['level-asymmetry-allowed'] === undefined) {\n codec.parameters['level-asymmetry-allowed'] = '1';\n }\n\n // for subsequent offers, we might have to re-use the payload\n // type of the last offer.\n if (transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.codecs) {\n transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) {\n if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() &&\n codec.clockRate === remoteCodec.clockRate) {\n codec.preferredPayloadType = remoteCodec.payloadType;\n }\n });\n }\n });\n localCapabilities.headerExtensions.forEach(function(hdrExt) {\n var remoteExtensions = transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.headerExtensions || [];\n remoteExtensions.forEach(function(rHdrExt) {\n if (hdrExt.uri === rHdrExt.uri) {\n hdrExt.id = rHdrExt.id;\n }\n });\n });\n\n // generate an ssrc now, to be used later in rtpSender.send\n var sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 1) * 1001\n }];\n if (track) {\n // add RTX\n if (edgeVersion >= 15019 && kind === 'video' &&\n !sendEncodingParameters[0].rtx) {\n sendEncodingParameters[0].rtx = {\n ssrc: sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n\n if (transceiver.wantReceive) {\n transceiver.rtpReceiver = new window.RTCRtpReceiver(\n transceiver.dtlsTransport, kind);\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n });\n\n // always offer BUNDLE and dispose on return if not supported.\n if (pc._config.bundlePolicy !== 'max-compat') {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n sdp += writeMediaSection(transceiver, transceiver.localCapabilities,\n 'offer', transceiver.stream, pc._dtlsRole);\n sdp += 'a=rtcp-rsize\\r\\n';\n\n if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' &&\n (sdpMLineIndex === 0 || !pc.usingBundle)) {\n transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) {\n cand.component = 1;\n sdp += 'a=' + SDPUtils.writeCandidate(cand) + '\\r\\n';\n });\n\n if (transceiver.iceGatherer.state === 'completed') {\n sdp += 'a=end-of-candidates\\r\\n';\n }\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'offer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.createAnswer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer after close'));\n }\n\n if (!(pc.signalingState === 'have-remote-offer' ||\n pc.signalingState === 'have-local-pranswer')) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer in signalingState ' + pc.signalingState));\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n if (pc.usingBundle) {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n var mediaSectionsInOffer = SDPUtils.getMediaSections(\n pc._remoteDescription.sdp).length;\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n if (sdpMLineIndex + 1 > mediaSectionsInOffer) {\n return;\n }\n if (transceiver.rejected) {\n if (transceiver.kind === 'application') {\n if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt\n sdp += 'm=application 0 DTLS/SCTP 5000\\r\\n';\n } else {\n sdp += 'm=application 0 ' + transceiver.protocol +\n ' webrtc-datachannel\\r\\n';\n }\n } else if (transceiver.kind === 'audio') {\n sdp += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\\r\\n' +\n 'a=rtpmap:0 PCMU/8000\\r\\n';\n } else if (transceiver.kind === 'video') {\n sdp += 'm=video 0 UDP/TLS/RTP/SAVPF 120\\r\\n' +\n 'a=rtpmap:120 VP8/90000\\r\\n';\n }\n sdp += 'c=IN IP4 0.0.0.0\\r\\n' +\n 'a=inactive\\r\\n' +\n 'a=mid:' + transceiver.mid + '\\r\\n';\n return;\n }\n\n // FIXME: look at direction.\n if (transceiver.stream) {\n var localTrack;\n if (transceiver.kind === 'audio') {\n localTrack = transceiver.stream.getAudioTracks()[0];\n } else if (transceiver.kind === 'video') {\n localTrack = transceiver.stream.getVideoTracks()[0];\n }\n if (localTrack) {\n // add RTX\n if (edgeVersion >= 15019 && transceiver.kind === 'video' &&\n !transceiver.sendEncodingParameters[0].rtx) {\n transceiver.sendEncodingParameters[0].rtx = {\n ssrc: transceiver.sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n }\n\n // Calculate intersection of capabilities.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n sdp += writeMediaSection(transceiver, commonCapabilities,\n 'answer', transceiver.stream, pc._dtlsRole);\n if (transceiver.rtcpParameters &&\n transceiver.rtcpParameters.reducedSize) {\n sdp += 'a=rtcp-rsize\\r\\n';\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'answer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.addIceCandidate = function(candidate) {\n var pc = this;\n var sections;\n if (candidate && !(candidate.sdpMLineIndex !== undefined ||\n candidate.sdpMid)) {\n return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required'));\n }\n\n // TODO: needs to go into ops queue.\n return new Promise(function(resolve, reject) {\n if (!pc._remoteDescription) {\n return reject(makeError('InvalidStateError',\n 'Can not add ICE candidate without a remote description'));\n } else if (!candidate || candidate.candidate === '') {\n for (var j = 0; j < pc.transceivers.length; j++) {\n if (pc.transceivers[j].rejected) {\n continue;\n }\n pc.transceivers[j].iceTransport.addRemoteCandidate({});\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[j] += 'a=end-of-candidates\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n if (pc.usingBundle) {\n break;\n }\n }\n } else {\n var sdpMLineIndex = candidate.sdpMLineIndex;\n if (candidate.sdpMid) {\n for (var i = 0; i < pc.transceivers.length; i++) {\n if (pc.transceivers[i].mid === candidate.sdpMid) {\n sdpMLineIndex = i;\n break;\n }\n }\n }\n var transceiver = pc.transceivers[sdpMLineIndex];\n if (transceiver) {\n if (transceiver.rejected) {\n return resolve();\n }\n var cand = Object.keys(candidate.candidate).length > 0 ?\n SDPUtils.parseCandidate(candidate.candidate) : {};\n // Ignore Chrome's invalid candidates since Edge does not like them.\n if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) {\n return resolve();\n }\n // Ignore RTCP candidates, we assume RTCP-MUX.\n if (cand.component && cand.component !== 1) {\n return resolve();\n }\n // when using bundle, avoid adding candidates to the wrong\n // ice transport. And avoid adding candidates added in the SDP.\n if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 &&\n transceiver.iceTransport !== pc.transceivers[0].iceTransport)) {\n if (!maybeAddCandidate(transceiver.iceTransport, cand)) {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n\n // update the remoteDescription.\n var candidateString = candidate.candidate.trim();\n if (candidateString.indexOf('a=') === 0) {\n candidateString = candidateString.substr(2);\n }\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[sdpMLineIndex] += 'a=' +\n (cand.type ? candidateString : 'end-of-candidates')\n + '\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n } else {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n resolve();\n });\n };\n\n RTCPeerConnection.prototype.getStats = function(selector) {\n if (selector && selector instanceof window.MediaStreamTrack) {\n var senderOrReceiver = null;\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.rtpSender &&\n transceiver.rtpSender.track === selector) {\n senderOrReceiver = transceiver.rtpSender;\n } else if (transceiver.rtpReceiver &&\n transceiver.rtpReceiver.track === selector) {\n senderOrReceiver = transceiver.rtpReceiver;\n }\n });\n if (!senderOrReceiver) {\n throw makeError('InvalidAccessError', 'Invalid selector.');\n }\n return senderOrReceiver.getStats();\n }\n\n var promises = [];\n this.transceivers.forEach(function(transceiver) {\n ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport',\n 'dtlsTransport'].forEach(function(method) {\n if (transceiver[method]) {\n promises.push(transceiver[method].getStats());\n }\n });\n });\n return Promise.all(promises).then(function(allStats) {\n var results = new Map();\n allStats.forEach(function(stats) {\n stats.forEach(function(stat) {\n results.set(stat.id, stat);\n });\n });\n return results;\n });\n };\n\n // fix low-level stat names and return Map instead of object.\n var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer',\n 'RTCIceTransport', 'RTCDtlsTransport'];\n ortcObjects.forEach(function(ortcObjectName) {\n var obj = window[ortcObjectName];\n if (obj && obj.prototype && obj.prototype.getStats) {\n var nativeGetstats = obj.prototype.getStats;\n obj.prototype.getStats = function() {\n return nativeGetstats.apply(this)\n .then(function(nativeStats) {\n var mapStats = new Map();\n Object.keys(nativeStats).forEach(function(id) {\n nativeStats[id].type = fixStatsType(nativeStats[id]);\n mapStats.set(id, nativeStats[id]);\n });\n return mapStats;\n });\n };\n }\n });\n\n // legacy callback shims. Should be moved to adapter.js some days.\n var methods = ['createOffer', 'createAnswer'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[0] === 'function' ||\n typeof args[1] === 'function') { // legacy\n return nativeMethod.apply(this, [arguments[2]])\n .then(function(description) {\n if (typeof args[0] === 'function') {\n args[0].apply(null, [description]);\n }\n }, function(error) {\n if (typeof args[1] === 'function') {\n args[1].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function' ||\n typeof args[2] === 'function') { // legacy\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n }, function(error) {\n if (typeof args[2] === 'function') {\n args[2].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n // getStats is special. It doesn't have a spec legacy method yet we support\n // getStats(something, cb) without error callbacks.\n ['getStats'].forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function') {\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n return RTCPeerConnection;\n};\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n const shimError_ = function(e) {\n return {\n name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name,\n message: e.message,\n constraint: e.constraint,\n toString() {\n return this.name;\n }\n };\n };\n\n // getUserMedia error shim.\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e)));\n };\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window) {\n if (!('getDisplayMedia' in window.navigator)) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n window.navigator.getDisplayMedia.bind(window.navigator);\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nimport {filterIceServers} from './filtericeservers';\nimport shimRTCPeerConnection from 'rtcpeerconnection-shim';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimPeerConnection(window, browserDetails) {\n if (window.RTCIceGatherer) {\n if (!window.RTCIceCandidate) {\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n return args;\n };\n }\n if (!window.RTCSessionDescription) {\n window.RTCSessionDescription = function RTCSessionDescription(args) {\n return args;\n };\n }\n // this adds an additional event listener to MediaStrackTrack that signals\n // when a tracks enabled property was changed. Workaround for a bug in\n // addStream, see below. No longer required in 15025+\n if (browserDetails.version < 15025) {\n const origMSTEnabled = Object.getOwnPropertyDescriptor(\n window.MediaStreamTrack.prototype, 'enabled');\n Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', {\n set(value) {\n origMSTEnabled.set.call(this, value);\n const ev = new Event('enabled');\n ev.enabled = value;\n this.dispatchEvent(ev);\n }\n });\n }\n }\n\n // ORTC defines the DTMF sender a bit different.\n // https://github.com/w3c/ortc/issues/714\n if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) {\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = new window.RTCDtmfSender(this);\n } else if (this.track.kind === 'video') {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n // Edge currently only implements the RTCDtmfSender, not the\n // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2*\n if (window.RTCDtmfSender && !window.RTCDTMFSender) {\n window.RTCDTMFSender = window.RTCDtmfSender;\n }\n\n const RTCPeerConnectionShim = shimRTCPeerConnection(window,\n browserDetails.version);\n window.RTCPeerConnection = function RTCPeerConnection(config) {\n if (config && config.iceServers) {\n config.iceServers = filterIceServers(config.iceServers,\n browserDetails.version);\n utils.log('ICE servers after filtering:', config.iceServers);\n }\n return new RTCPeerConnectionShim(config);\n };\n window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype;\n}\n\nexport function shimReplaceTrack(window) {\n // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614\n if (window.RTCRtpSender &&\n !('replaceTrack' in window.RTCRtpSender.prototype)) {\n window.RTCRtpSender.prototype.replaceTrack =\n window.RTCRtpSender.prototype.setTrack;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n const MediaStreamTrack = window && window.MediaStreamTrack;\n\n navigator.getUserMedia = function(constraints, onSuccess, onError) {\n // Replace Firefox 44+'s deprecation warning with unprefixed version.\n utils.deprecated('navigator.getUserMedia',\n 'navigator.mediaDevices.getUserMedia');\n navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError);\n };\n\n if (!(browserDetails.version > 55 &&\n 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n\n const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n if (typeof c === 'object' && typeof c.audio === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c.audio, 'autoGainControl', 'mozAutoGainControl');\n remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeGetUserMedia(c);\n };\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) {\n const nativeGetSettings = MediaStreamTrack.prototype.getSettings;\n MediaStreamTrack.prototype.getSettings = function() {\n const obj = nativeGetSettings.apply(this, arguments);\n remap(obj, 'mozAutoGainControl', 'autoGainControl');\n remap(obj, 'mozNoiseSuppression', 'noiseSuppression');\n return obj;\n };\n }\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) {\n const nativeApplyConstraints =\n MediaStreamTrack.prototype.applyConstraints;\n MediaStreamTrack.prototype.applyConstraints = function(c) {\n if (this.kind === 'audio' && typeof c === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c, 'autoGainControl', 'mozAutoGainControl');\n remap(c, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeApplyConstraints.apply(this, [c]);\n };\n }\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window, preferredMediaSource) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n if (!(constraints && constraints.video)) {\n const err = new DOMException('getDisplayMedia without video ' +\n 'constraints is undefined');\n err.name = 'NotFoundError';\n // from https://heycam.github.io/webidl/#idl-DOMException-error-names\n err.code = 8;\n return Promise.reject(err);\n }\n if (constraints.video === true) {\n constraints.video = {mediaSource: preferredMediaSource};\n } else {\n constraints.video.mediaSource = preferredMediaSource;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCTrackEvent &&\n ('receiver' in window.RTCTrackEvent.prototype) &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (typeof window !== 'object' ||\n !(window.RTCPeerConnection || window.mozRTCPeerConnection)) {\n return; // probably media.peerconnection.enabled=false in about:config\n }\n if (!window.RTCPeerConnection && window.mozRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.mozRTCPeerConnection;\n }\n\n if (browserDetails.version < 53) {\n // shim away need for obsolete RTCIceCandidate/RTCSessionDescription.\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n\n const modernStatsTypes = {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n };\n\n const nativeGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n return nativeGetStats.apply(this, [selector || null])\n .then(stats => {\n if (browserDetails.version < 53 && !onSucc) {\n // Shim only promise getStats with spec-hyphens in type names\n // Leave callback version alone; misc old uses of forEach before Map\n try {\n stats.forEach(stat => {\n stat.type = modernStatsTypes[stat.type] || stat.type;\n });\n } catch (e) {\n if (e.name !== 'TypeError') {\n throw e;\n }\n // Avoid TypeError: \"type\" is read-only, in old versions. 34-43ish\n stats.forEach((stat, i) => {\n stats.set(i, Object.assign({}, stat, {\n type: modernStatsTypes[stat.type] || stat.type\n }));\n });\n }\n }\n return stats;\n })\n .then(onSucc, onErr);\n };\n}\n\nexport function shimSenderGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) {\n return;\n }\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n return this.track ? this._pc.getStats(this.track) :\n Promise.resolve(new Map());\n };\n}\n\nexport function shimReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) {\n return;\n }\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers = function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n return this._pc.getStats(this.track);\n };\n}\n\nexport function shimRemoveStream(window) {\n if (!window.RTCPeerConnection ||\n 'removeStream' in window.RTCPeerConnection.prototype) {\n return;\n }\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n utils.deprecated('removeStream', 'removeTrack');\n this.getSenders().forEach(sender => {\n if (sender.track && stream.getTracks().includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n}\n\nexport function shimRTCDataChannel(window) {\n // rename DataChannel to RTCDataChannel (native fix in FF60):\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851\n if (window.DataChannel && !window.RTCDataChannel) {\n window.RTCDataChannel = window.DataChannel;\n }\n}\n\nexport function shimAddTransceiver(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver;\n if (origAddTransceiver) {\n window.RTCPeerConnection.prototype.addTransceiver =\n function addTransceiver() {\n this.setParametersPromises = [];\n const initParameters = arguments[1];\n const shouldPerformCheck = initParameters &&\n 'sendEncodings' in initParameters;\n if (shouldPerformCheck) {\n // If sendEncodings params are provided, validate grammar\n initParameters.sendEncodings.forEach((encodingParam) => {\n if ('rid' in encodingParam) {\n const ridRegex = /^[a-z0-9]{0,16}$/i;\n if (!ridRegex.test(encodingParam.rid)) {\n throw new TypeError('Invalid RID value provided.');\n }\n }\n if ('scaleResolutionDownBy' in encodingParam) {\n if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) {\n throw new RangeError('scale_resolution_down_by must be >= 1.0');\n }\n }\n if ('maxFramerate' in encodingParam) {\n if (!(parseFloat(encodingParam.maxFramerate) >= 0)) {\n throw new RangeError('max_framerate must be >= 0.0');\n }\n }\n });\n }\n const transceiver = origAddTransceiver.apply(this, arguments);\n if (shouldPerformCheck) {\n // Check if the init options were applied. If not we do this in an\n // asynchronous way and save the promise reference in a global object.\n // This is an ugly hack, but at the same time is way more robust than\n // checking the sender parameters before and after the createOffer\n // Also note that after the createoffer we are not 100% sure that\n // the params were asynchronously applied so we might miss the\n // opportunity to recreate offer.\n const {sender} = transceiver;\n const params = sender.getParameters();\n if (!('encodings' in params) ||\n // Avoid being fooled by patched getParameters() below.\n (params.encodings.length === 1 &&\n Object.keys(params.encodings[0]).length === 0)) {\n params.encodings = initParameters.sendEncodings;\n sender.sendEncodings = initParameters.sendEncodings;\n this.setParametersPromises.push(sender.setParameters(params)\n .then(() => {\n delete sender.sendEncodings;\n }).catch(() => {\n delete sender.sendEncodings;\n })\n );\n }\n }\n return transceiver;\n };\n }\n}\n\nexport function shimGetParameters(window) {\n if (!(typeof window === 'object' && window.RTCRtpSender)) {\n return;\n }\n const origGetParameters = window.RTCRtpSender.prototype.getParameters;\n if (origGetParameters) {\n window.RTCRtpSender.prototype.getParameters =\n function getParameters() {\n const params = origGetParameters.apply(this, arguments);\n if (!('encodings' in params)) {\n params.encodings = [].concat(this.sendEncodings || [{}]);\n }\n return params;\n };\n }\n}\n\nexport function shimCreateOffer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer = function createOffer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateOffer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimCreateAnswer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer;\n window.RTCPeerConnection.prototype.createAnswer = function createAnswer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateAnswer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateAnswer.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n'use strict';\nimport * as utils from '../utils';\n\nexport function shimLocalStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n return this._localStreams;\n };\n }\n if (!('addStream' in window.RTCPeerConnection.prototype)) {\n const _addTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n // Try to emulate Chrome's behaviour of adding in audio-video order.\n // Safari orders by track id.\n stream.getAudioTracks().forEach(track => _addTrack.call(this, track,\n stream));\n stream.getVideoTracks().forEach(track => _addTrack.call(this, track,\n stream));\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, ...streams) {\n if (streams) {\n streams.forEach((stream) => {\n if (!this._localStreams) {\n this._localStreams = [stream];\n } else if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n });\n }\n return _addTrack.apply(this, arguments);\n };\n }\n if (!('removeStream' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n const index = this._localStreams.indexOf(stream);\n if (index === -1) {\n return;\n }\n this._localStreams.splice(index, 1);\n const tracks = stream.getTracks();\n this.getSenders().forEach(sender => {\n if (tracks.includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n }\n}\n\nexport function shimRemoteStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getRemoteStreams =\n function getRemoteStreams() {\n return this._remoteStreams ? this._remoteStreams : [];\n };\n }\n if (!('onaddstream' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', {\n get() {\n return this._onaddstream;\n },\n set(f) {\n if (this._onaddstream) {\n this.removeEventListener('addstream', this._onaddstream);\n this.removeEventListener('track', this._onaddstreampoly);\n }\n this.addEventListener('addstream', this._onaddstream = f);\n this.addEventListener('track', this._onaddstreampoly = (e) => {\n e.streams.forEach(stream => {\n if (!this._remoteStreams) {\n this._remoteStreams = [];\n }\n if (this._remoteStreams.includes(stream)) {\n return;\n }\n this._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n this.dispatchEvent(event);\n });\n });\n }\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n const pc = this;\n if (!this._onaddstreampoly) {\n this.addEventListener('track', this._onaddstreampoly = function(e) {\n e.streams.forEach(stream => {\n if (!pc._remoteStreams) {\n pc._remoteStreams = [];\n }\n if (pc._remoteStreams.indexOf(stream) >= 0) {\n return;\n }\n pc._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n pc.dispatchEvent(event);\n });\n });\n }\n return origSetRemoteDescription.apply(pc, arguments);\n };\n }\n}\n\nexport function shimCallbacksAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n const prototype = window.RTCPeerConnection.prototype;\n const origCreateOffer = prototype.createOffer;\n const origCreateAnswer = prototype.createAnswer;\n const setLocalDescription = prototype.setLocalDescription;\n const setRemoteDescription = prototype.setRemoteDescription;\n const addIceCandidate = prototype.addIceCandidate;\n\n prototype.createOffer =\n function createOffer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateOffer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n prototype.createAnswer =\n function createAnswer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateAnswer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n let withCallback = function(description, successCallback, failureCallback) {\n const promise = setLocalDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setLocalDescription = withCallback;\n\n withCallback = function(description, successCallback, failureCallback) {\n const promise = setRemoteDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setRemoteDescription = withCallback;\n\n withCallback = function(candidate, successCallback, failureCallback) {\n const promise = addIceCandidate.apply(this, [candidate]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.addIceCandidate = withCallback;\n}\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {\n // shim not needed in Safari 12.1\n const mediaDevices = navigator.mediaDevices;\n const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices);\n navigator.mediaDevices.getUserMedia = (constraints) => {\n return _getUserMedia(shimConstraints(constraints));\n };\n }\n\n if (!navigator.getUserMedia && navigator.mediaDevices &&\n navigator.mediaDevices.getUserMedia) {\n navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) {\n navigator.mediaDevices.getUserMedia(constraints)\n .then(cb, errcb);\n }.bind(navigator);\n }\n}\n\nexport function shimConstraints(constraints) {\n if (constraints && constraints.video !== undefined) {\n return Object.assign({},\n constraints,\n {video: utils.compactObject(constraints.video)}\n );\n }\n\n return constraints;\n}\n\nexport function shimRTCIceServerUrls(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // migrate from non-spec RTCIceServer.url to RTCIceServer.urls\n const OrigPeerConnection = window.RTCPeerConnection;\n window.RTCPeerConnection =\n function RTCPeerConnection(pcConfig, pcConstraints) {\n if (pcConfig && pcConfig.iceServers) {\n const newIceServers = [];\n for (let i = 0; i < pcConfig.iceServers.length; i++) {\n let server = pcConfig.iceServers[i];\n if (!server.hasOwnProperty('urls') &&\n server.hasOwnProperty('url')) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n server = JSON.parse(JSON.stringify(server));\n server.urls = server.url;\n delete server.url;\n newIceServers.push(server);\n } else {\n newIceServers.push(pcConfig.iceServers[i]);\n }\n }\n pcConfig.iceServers = newIceServers;\n }\n return new OrigPeerConnection(pcConfig, pcConstraints);\n };\n window.RTCPeerConnection.prototype = OrigPeerConnection.prototype;\n // wrap static methods. Currently just generateCertificate.\n if ('generateCertificate' in OrigPeerConnection) {\n Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', {\n get() {\n return OrigPeerConnection.generateCertificate;\n }\n });\n }\n}\n\nexport function shimTrackEventTransceiver(window) {\n // Add event.transceiver member over deprecated event.receiver\n if (typeof window === 'object' && window.RTCTrackEvent &&\n 'receiver' in window.RTCTrackEvent.prototype &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimCreateOfferLegacy(window) {\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer =\n function createOffer(offerOptions) {\n if (offerOptions) {\n if (typeof offerOptions.offerToReceiveAudio !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveAudio =\n !!offerOptions.offerToReceiveAudio;\n }\n const audioTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'audio');\n if (offerOptions.offerToReceiveAudio === false && audioTransceiver) {\n if (audioTransceiver.direction === 'sendrecv') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('sendonly');\n } else {\n audioTransceiver.direction = 'sendonly';\n }\n } else if (audioTransceiver.direction === 'recvonly') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('inactive');\n } else {\n audioTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveAudio === true &&\n !audioTransceiver) {\n this.addTransceiver('audio');\n }\n\n if (typeof offerOptions.offerToReceiveVideo !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveVideo =\n !!offerOptions.offerToReceiveVideo;\n }\n const videoTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'video');\n if (offerOptions.offerToReceiveVideo === false && videoTransceiver) {\n if (videoTransceiver.direction === 'sendrecv') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('sendonly');\n } else {\n videoTransceiver.direction = 'sendonly';\n }\n } else if (videoTransceiver.direction === 'recvonly') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('inactive');\n } else {\n videoTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveVideo === true &&\n !videoTransceiver) {\n this.addTransceiver('video');\n }\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimAudioContext(window) {\n if (typeof window !== 'object' || window.AudioContext) {\n return;\n }\n window.AudioContext = window.webkitAudioContext;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport SDPUtils from 'sdp';\nimport * as utils from './utils';\n\nexport function shimRTCIceCandidate(window) {\n // foundation is arbitrarily chosen as an indicator for full support for\n // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface\n if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in\n window.RTCIceCandidate.prototype)) {\n return;\n }\n\n const NativeRTCIceCandidate = window.RTCIceCandidate;\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n // Remove the a= which shouldn't be part of the candidate string.\n if (typeof args === 'object' && args.candidate &&\n args.candidate.indexOf('a=') === 0) {\n args = JSON.parse(JSON.stringify(args));\n args.candidate = args.candidate.substr(2);\n }\n\n if (args.candidate && args.candidate.length) {\n // Augment the native candidate with the parsed fields.\n const nativeCandidate = new NativeRTCIceCandidate(args);\n const parsedCandidate = SDPUtils.parseCandidate(args.candidate);\n const augmentedCandidate = Object.assign(nativeCandidate,\n parsedCandidate);\n\n // Add a serializer that does not serialize the extra attributes.\n augmentedCandidate.toJSON = function toJSON() {\n return {\n candidate: augmentedCandidate.candidate,\n sdpMid: augmentedCandidate.sdpMid,\n sdpMLineIndex: augmentedCandidate.sdpMLineIndex,\n usernameFragment: augmentedCandidate.usernameFragment,\n };\n };\n return augmentedCandidate;\n }\n return new NativeRTCIceCandidate(args);\n };\n window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype;\n\n // Hook up the augmented candidate in onicecandidate and\n // addEventListener('icecandidate', ...)\n utils.wrapPeerConnectionEvent(window, 'icecandidate', e => {\n if (e.candidate) {\n Object.defineProperty(e, 'candidate', {\n value: new window.RTCIceCandidate(e.candidate),\n writable: 'false'\n });\n }\n return e;\n });\n}\n\nexport function shimMaxMessageSize(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n if (!('sctp' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n }\n });\n }\n\n const sctpInDescription = function(description) {\n if (!description || !description.sdp) {\n return false;\n }\n const sections = SDPUtils.splitSections(description.sdp);\n sections.shift();\n return sections.some(mediaSection => {\n const mLine = SDPUtils.parseMLine(mediaSection);\n return mLine && mLine.kind === 'application'\n && mLine.protocol.indexOf('SCTP') !== -1;\n });\n };\n\n const getRemoteFirefoxVersion = function(description) {\n // TODO: Is there a better solution for detecting Firefox?\n const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\\d+)/);\n if (match === null || match.length < 2) {\n return -1;\n }\n const version = parseInt(match[1], 10);\n // Test for NaN (yes, this is ugly)\n return version !== version ? -1 : version;\n };\n\n const getCanSendMaxMessageSize = function(remoteIsFirefox) {\n // Every implementation we know can send at least 64 KiB.\n // Note: Although Chrome is technically able to send up to 256 KiB, the\n // data does not reach the other peer reliably.\n // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419\n let canSendMaxMessageSize = 65536;\n if (browserDetails.browser === 'firefox') {\n if (browserDetails.version < 57) {\n if (remoteIsFirefox === -1) {\n // FF < 57 will send in 16 KiB chunks using the deprecated PPID\n // fragmentation.\n canSendMaxMessageSize = 16384;\n } else {\n // However, other FF (and RAWRTC) can reassemble PPID-fragmented\n // messages. Thus, supporting ~2 GiB when sending.\n canSendMaxMessageSize = 2147483637;\n }\n } else if (browserDetails.version < 60) {\n // Currently, all FF >= 57 will reset the remote maximum message size\n // to the default value when a data channel is created at a later\n // stage. :(\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n canSendMaxMessageSize =\n browserDetails.version === 57 ? 65535 : 65536;\n } else {\n // FF >= 60 supports sending ~2 GiB\n canSendMaxMessageSize = 2147483637;\n }\n }\n return canSendMaxMessageSize;\n };\n\n const getMaxMessageSize = function(description, remoteIsFirefox) {\n // Note: 65536 bytes is the default value from the SDP spec. Also,\n // every implementation we know supports receiving 65536 bytes.\n let maxMessageSize = 65536;\n\n // FF 57 has a slightly incorrect default remote max message size, so\n // we need to adjust it here to avoid a failure when sending.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697\n if (browserDetails.browser === 'firefox'\n && browserDetails.version === 57) {\n maxMessageSize = 65535;\n }\n\n const match = SDPUtils.matchPrefix(description.sdp,\n 'a=max-message-size:');\n if (match.length > 0) {\n maxMessageSize = parseInt(match[0].substr(19), 10);\n } else if (browserDetails.browser === 'firefox' &&\n remoteIsFirefox !== -1) {\n // If the maximum message size is not present in the remote SDP and\n // both local and remote are Firefox, the remote peer can receive\n // ~2 GiB.\n maxMessageSize = 2147483637;\n }\n return maxMessageSize;\n };\n\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n this._sctp = null;\n // Chrome decided to not expose .sctp in plan-b mode.\n // As usual, adapter.js has to do an 'ugly worakaround'\n // to cover up the mess.\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) {\n const {sdpSemantics} = this.getConfiguration();\n if (sdpSemantics === 'plan-b') {\n Object.defineProperty(this, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n },\n enumerable: true,\n configurable: true,\n });\n }\n }\n\n if (sctpInDescription(arguments[0])) {\n // Check if the remote is FF.\n const isFirefox = getRemoteFirefoxVersion(arguments[0]);\n\n // Get the maximum message size the local peer is capable of sending\n const canSendMMS = getCanSendMaxMessageSize(isFirefox);\n\n // Get the maximum message size of the remote peer.\n const remoteMMS = getMaxMessageSize(arguments[0], isFirefox);\n\n // Determine final maximum message size\n let maxMessageSize;\n if (canSendMMS === 0 && remoteMMS === 0) {\n maxMessageSize = Number.POSITIVE_INFINITY;\n } else if (canSendMMS === 0 || remoteMMS === 0) {\n maxMessageSize = Math.max(canSendMMS, remoteMMS);\n } else {\n maxMessageSize = Math.min(canSendMMS, remoteMMS);\n }\n\n // Create a dummy RTCSctpTransport object and the 'maxMessageSize'\n // attribute.\n const sctp = {};\n Object.defineProperty(sctp, 'maxMessageSize', {\n get() {\n return maxMessageSize;\n }\n });\n this._sctp = sctp;\n }\n\n return origSetRemoteDescription.apply(this, arguments);\n };\n}\n\nexport function shimSendThrowTypeError(window) {\n if (!(window.RTCPeerConnection &&\n 'createDataChannel' in window.RTCPeerConnection.prototype)) {\n return;\n }\n\n // Note: Although Firefox >= 57 has a native implementation, the maximum\n // message size can be reset for all data channels at a later stage.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n\n function wrapDcSend(dc, pc) {\n const origDataChannelSend = dc.send;\n dc.send = function send() {\n const data = arguments[0];\n const length = data.length || data.size || data.byteLength;\n if (dc.readyState === 'open' &&\n pc.sctp && length > pc.sctp.maxMessageSize) {\n throw new TypeError('Message too large (can send a maximum of ' +\n pc.sctp.maxMessageSize + ' bytes)');\n }\n return origDataChannelSend.apply(dc, arguments);\n };\n }\n const origCreateDataChannel =\n window.RTCPeerConnection.prototype.createDataChannel;\n window.RTCPeerConnection.prototype.createDataChannel =\n function createDataChannel() {\n const dataChannel = origCreateDataChannel.apply(this, arguments);\n wrapDcSend(dataChannel, this);\n return dataChannel;\n };\n utils.wrapPeerConnectionEvent(window, 'datachannel', e => {\n wrapDcSend(e.channel, e.target);\n return e;\n });\n}\n\n\n/* shims RTCConnectionState by pretending it is the same as iceConnectionState.\n * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12\n * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect\n * since DTLS failures would be hidden. See\n * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827\n * for the Firefox tracking bug.\n */\nexport function shimConnectionState(window) {\n if (!window.RTCPeerConnection ||\n 'connectionState' in window.RTCPeerConnection.prototype) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n Object.defineProperty(proto, 'connectionState', {\n get() {\n return {\n completed: 'connected',\n checking: 'connecting'\n }[this.iceConnectionState] || this.iceConnectionState;\n },\n enumerable: true,\n configurable: true\n });\n Object.defineProperty(proto, 'onconnectionstatechange', {\n get() {\n return this._onconnectionstatechange || null;\n },\n set(cb) {\n if (this._onconnectionstatechange) {\n this.removeEventListener('connectionstatechange',\n this._onconnectionstatechange);\n delete this._onconnectionstatechange;\n }\n if (cb) {\n this.addEventListener('connectionstatechange',\n this._onconnectionstatechange = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n\n ['setLocalDescription', 'setRemoteDescription'].forEach((method) => {\n const origMethod = proto[method];\n proto[method] = function() {\n if (!this._connectionstatechangepoly) {\n this._connectionstatechangepoly = e => {\n const pc = e.target;\n if (pc._lastConnectionState !== pc.connectionState) {\n pc._lastConnectionState = pc.connectionState;\n const newEvent = new Event('connectionstatechange', e);\n pc.dispatchEvent(newEvent);\n }\n return e;\n };\n this.addEventListener('iceconnectionstatechange',\n this._connectionstatechangepoly);\n }\n return origMethod.apply(this, arguments);\n };\n });\n}\n\nexport function removeExtmapAllowMixed(window, browserDetails) {\n /* remove a=extmap-allow-mixed for webrtc.org < M71 */\n if (!window.RTCPeerConnection) {\n return;\n }\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) {\n return;\n }\n if (browserDetails.browser === 'safari' && browserDetails.version >= 605) {\n return;\n }\n const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription(desc) {\n if (desc && desc.sdp && desc.sdp.indexOf('\\na=extmap-allow-mixed') !== -1) {\n const sdp = desc.sdp.split('\\n').filter((line) => {\n return line.trim() !== 'a=extmap-allow-mixed';\n }).join('\\n');\n // Safari enforces read-only-ness of RTCSessionDescription fields.\n if (window.RTCSessionDescription &&\n desc instanceof window.RTCSessionDescription) {\n arguments[0] = new window.RTCSessionDescription({\n type: desc.type,\n sdp,\n });\n } else {\n desc.sdp = sdp;\n }\n }\n return nativeSRD.apply(this, arguments);\n };\n}\n\nexport function shimAddIceCandidateNullOrEmpty(window, browserDetails) {\n // Support for addIceCandidate(null or undefined)\n // as well as addIceCandidate({candidate: \"\", ...})\n // https://bugs.chromium.org/p/chromium/issues/detail?id=978582\n // Note: must be called before other polyfills which change the signature.\n if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) {\n return;\n }\n const nativeAddIceCandidate =\n window.RTCPeerConnection.prototype.addIceCandidate;\n if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) {\n return;\n }\n window.RTCPeerConnection.prototype.addIceCandidate =\n function addIceCandidate() {\n if (!arguments[0]) {\n if (arguments[1]) {\n arguments[1].apply(null);\n }\n return Promise.resolve();\n }\n // Firefox 68+ emits and processes {candidate: \"\", ...}, ignore\n // in older versions.\n // Native support for ignoring exists for Chrome M77+.\n // Safari ignores as well, exact version unknown but works in the same\n // version that also ignores addIceCandidate(null).\n if (((browserDetails.browser === 'chrome' && browserDetails.version < 78)\n || (browserDetails.browser === 'firefox'\n && browserDetails.version < 68)\n || (browserDetails.browser === 'safari'))\n && arguments[0] && arguments[0].candidate === '') {\n return Promise.resolve();\n }\n return nativeAddIceCandidate.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\nimport * as utils from './utils';\n\n // Browser shims.\nimport * as chromeShim from './chrome/chrome_shim';\nimport * as edgeShim from './edge/edge_shim';\nimport * as firefoxShim from './firefox/firefox_shim';\nimport * as safariShim from './safari/safari_shim';\nimport * as commonShim from './common_shim';\n\n// Shimming starts here.\nexport function adapterFactory({window} = {}, options = {\n shimChrome: true,\n shimFirefox: true,\n shimEdge: true,\n shimSafari: true,\n}) {\n // Utils.\n const logging = utils.log;\n const browserDetails = utils.detectBrowser(window);\n\n const adapter = {\n browserDetails,\n commonShim,\n extractVersion: utils.extractVersion,\n disableLog: utils.disableLog,\n disableWarnings: utils.disableWarnings\n };\n\n // Shim browser if found.\n switch (browserDetails.browser) {\n case 'chrome':\n if (!chromeShim || !chromeShim.shimPeerConnection ||\n !options.shimChrome) {\n logging('Chrome shim is not included in this adapter release.');\n return adapter;\n }\n if (browserDetails.version === null) {\n logging('Chrome shim can not determine version, not shimming.');\n return adapter;\n }\n logging('adapter.js shimming chrome.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = chromeShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n chromeShim.shimGetUserMedia(window, browserDetails);\n chromeShim.shimMediaStream(window, browserDetails);\n chromeShim.shimPeerConnection(window, browserDetails);\n chromeShim.shimOnTrack(window, browserDetails);\n chromeShim.shimAddTrackRemoveTrack(window, browserDetails);\n chromeShim.shimGetSendersWithDtmf(window, browserDetails);\n chromeShim.shimGetStats(window, browserDetails);\n chromeShim.shimSenderReceiverGetStats(window, browserDetails);\n chromeShim.fixNegotiationNeeded(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n case 'firefox':\n if (!firefoxShim || !firefoxShim.shimPeerConnection ||\n !options.shimFirefox) {\n logging('Firefox shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming firefox.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = firefoxShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n firefoxShim.shimGetUserMedia(window, browserDetails);\n firefoxShim.shimPeerConnection(window, browserDetails);\n firefoxShim.shimOnTrack(window, browserDetails);\n firefoxShim.shimRemoveStream(window, browserDetails);\n firefoxShim.shimSenderGetStats(window, browserDetails);\n firefoxShim.shimReceiverGetStats(window, browserDetails);\n firefoxShim.shimRTCDataChannel(window, browserDetails);\n firefoxShim.shimAddTransceiver(window, browserDetails);\n firefoxShim.shimGetParameters(window, browserDetails);\n firefoxShim.shimCreateOffer(window, browserDetails);\n firefoxShim.shimCreateAnswer(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'edge':\n if (!edgeShim || !edgeShim.shimPeerConnection || !options.shimEdge) {\n logging('MS edge shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming edge.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = edgeShim;\n\n edgeShim.shimGetUserMedia(window, browserDetails);\n edgeShim.shimGetDisplayMedia(window, browserDetails);\n edgeShim.shimPeerConnection(window, browserDetails);\n edgeShim.shimReplaceTrack(window, browserDetails);\n\n // the edge shim implements the full RTCIceCandidate object.\n\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'safari':\n if (!safariShim || !options.shimSafari) {\n logging('Safari shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming safari.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = safariShim;\n\n // Must be called before shimCallbackAPI.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n safariShim.shimRTCIceServerUrls(window, browserDetails);\n safariShim.shimCreateOfferLegacy(window, browserDetails);\n safariShim.shimCallbacksAPI(window, browserDetails);\n safariShim.shimLocalStreamsAPI(window, browserDetails);\n safariShim.shimRemoteStreamsAPI(window, browserDetails);\n safariShim.shimTrackEventTransceiver(window, browserDetails);\n safariShim.shimGetUserMedia(window, browserDetails);\n safariShim.shimAudioContext(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n default:\n logging('Unsupported browser!');\n break;\n }\n\n return adapter;\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n\n'use strict';\n\nimport {adapterFactory} from './adapter_factory.js';\n\nconst adapter =\n adapterFactory({window: typeof window === 'undefined' ? undefined : window});\nexport default adapter;\n","\n'use strict';\nimport * as utils from './utils.js';\nimport * as MediaFormatModule from './mediaformat.js';\nimport adapter from 'webrtc-adapter';\n\n/**\n * @class AudioTrackConstraints\n * @classDesc Constraints for creating an audio MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track.\n */\nexport class AudioTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.AudioSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Values could be \"mic\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Do not provide deviceId if source is not \"mic\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n this.deviceId = undefined;\n }\n}\n\n/**\n * @class VideoTrackConstraints\n * @classDesc Constraints for creating a video MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.VideoSourceInfo} source Source info of this video track.\n */\nexport class VideoTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.VideoSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Values could be \"camera\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Do not provide deviceId if source is not \"camera\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n\n this.deviceId = undefined;\n\n /**\n * @member {Owt.Base.Resolution} resolution\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.resolution = undefined;\n\n /**\n * @member {number} frameRate\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.frameRate = undefined;\n }\n}\n/**\n * @class StreamConstraints\n * @classDesc Constraints for creating a MediaStream from screen mic and camera.\n * @memberof Owt.Base\n * @constructor\n * @param {?Owt.Base.AudioTrackConstraints} audioConstraints\n * @param {?Owt.Base.VideoTrackConstraints} videoConstraints\n */\nexport class StreamConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(audioConstraints = false, videoConstraints = false) {\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.audio = audioConstraints;\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.video = videoConstraints;\n }\n}\n\n// eslint-disable-next-line require-jsdoc\nfunction isVideoConstrainsForScreenCast(constraints) {\n return (typeof constraints.video === 'object' && constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST);\n}\n\n/**\n * @class MediaStreamFactory\n * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself.\n * @memberof Owt.Base\n */\nexport class MediaStreamFactory {\n /**\n * @function createMediaStream\n * @static\n * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are \"screen-cast\".\n * @memberof Owt.Base.MediaStreamFactory\n * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened:\n * - One or more parameters cannot be satisfied.\n * - Specified device is busy.\n * - Cannot obtain necessary permission or operation is canceled by user.\n * - Video source is screen cast, while audio source is not.\n * - Audio source is screen cast, while video source is disabled.\n * @param {Owt.Base.StreamConstraints} constraints\n */\n static createMediaStream(constraints) {\n if (typeof constraints !== 'object' ||\n (!constraints.audio && !constraints.video)) {\n return Promise.reject(new TypeError('Invalid constrains'));\n }\n if (!isVideoConstrainsForScreenCast(constraints) &&\n (typeof constraints.audio === 'object') &&\n constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(\n new TypeError('Cannot share screen without video.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) && !utils.isChrome() &&\n !utils.isFirefox()) {\n return Promise.reject(\n new TypeError('Screen sharing only supports Chrome and Firefox.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) &&\n typeof constraints.audio === 'object' &&\n constraints.audio.source !==\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(new TypeError(\n 'Cannot capture video from screen cast while capture audio from'\n + ' other source.'));\n }\n\n // Check and convert constraints.\n if (!constraints.audio && !constraints.video) {\n return Promise.reject(new TypeError(\n 'At least one of audio and video must be requested.'));\n }\n const mediaConstraints = Object.create({});\n if (typeof constraints.audio === 'object' &&\n constraints.audio.source === MediaFormatModule.AudioSourceInfo.MIC) {\n mediaConstraints.audio = Object.create({});\n if (utils.isEdge()) {\n mediaConstraints.audio.deviceId = constraints.audio.deviceId;\n } else {\n mediaConstraints.audio.deviceId = {\n exact: constraints.audio.deviceId,\n };\n }\n } else {\n if (constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n mediaConstraints.audio = true;\n } else {\n mediaConstraints.audio = constraints.audio;\n }\n }\n if (typeof constraints.video === 'object') {\n mediaConstraints.video = Object.create({});\n if (typeof constraints.video.frameRate === 'number') {\n mediaConstraints.video.frameRate = constraints.video.frameRate;\n }\n if (constraints.video.resolution &&\n constraints.video.resolution.width &&\n constraints.video.resolution.height) {\n if (constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.width = constraints.video.resolution.width;\n mediaConstraints.video.height = constraints.video.resolution.height;\n } else {\n mediaConstraints.video.width = Object.create({});\n mediaConstraints.video.width.exact =\n constraints.video.resolution.width;\n mediaConstraints.video.height = Object.create({});\n mediaConstraints.video.height.exact =\n constraints.video.resolution.height;\n }\n }\n if (typeof constraints.video.deviceId === 'string') {\n mediaConstraints.video.deviceId = {exact: constraints.video.deviceId};\n }\n if (utils.isFirefox() &&\n constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.mediaSource = 'screen';\n }\n } else {\n mediaConstraints.video = constraints.video;\n }\n\n if (isVideoConstrainsForScreenCast(constraints)) {\n return navigator.mediaDevices.getDisplayMedia(mediaConstraints);\n } else {\n return navigator.mediaDevices.getUserMedia(mediaConstraints);\n }\n }\n}\n","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n\nexport * from './mediastream-factory.js';\nexport * from './mediaformat.js';","let logger;\nlet errorLogger;\n\nexport function setLogger() {\n /*eslint-disable */\n logger = console.log;\n errorLogger = console.error;\n /*eslint-enable */\n}\n\nexport function isEnable() {\n return logger != null;\n}\n\nexport function log(message, ...optionalParams) {\n if (logger) {\n logger(message, ...optionalParams);\n }\n}\nexport function error(message, ...optionalParams) {\n if (errorLogger) {\n errorLogger(message, ...optionalParams);\n }\n}\n","export default class Event {\n constructor(type) {\n this.listener = {};\n this.type = type | '';\n }\n\n on(event, fn) {\n if (!this.listener[event]) {\n this.listener[event] = [];\n }\n this.listener[event].push(fn);\n return true;\n }\n\n off(event, fn) {\n if (this.listener[event]) {\n var index = this.listener[event].indexOf(fn);\n if (index > -1) {\n this.listener[event].splice(index, 1);\n }\n return true;\n }\n return false;\n }\n\n offAll() {\n this.listener = {};\n }\n\n dispatch(event, data) {\n if (this.listener[event]) {\n this.listener[event].map((each) => {\n each.apply(null, [data]);\n });\n return true;\n }\n return false;\n }\n}\n","'use strict';\n\nmodule.exports = function bind(fn, thisArg) {\n return function wrap() {\n var args = new Array(arguments.length);\n for (var i = 0; i < args.length; i++) {\n args[i] = arguments[i];\n }\n return fn.apply(thisArg, args);\n };\n};\n","'use strict';\n\nvar bind = require('./helpers/bind');\n\n/*global toString:true*/\n\n// utils is a library of generic helper functions non-specific to axios\n\nvar toString = Object.prototype.toString;\n\n/**\n * Determine if a value is an Array\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Array, otherwise false\n */\nfunction isArray(val) {\n return toString.call(val) === '[object Array]';\n}\n\n/**\n * Determine if a value is undefined\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if the value is undefined, otherwise false\n */\nfunction isUndefined(val) {\n return typeof val === 'undefined';\n}\n\n/**\n * Determine if a value is a Buffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Buffer, otherwise false\n */\nfunction isBuffer(val) {\n return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor)\n && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val);\n}\n\n/**\n * Determine if a value is an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an ArrayBuffer, otherwise false\n */\nfunction isArrayBuffer(val) {\n return toString.call(val) === '[object ArrayBuffer]';\n}\n\n/**\n * Determine if a value is a FormData\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an FormData, otherwise false\n */\nfunction isFormData(val) {\n return (typeof FormData !== 'undefined') && (val instanceof FormData);\n}\n\n/**\n * Determine if a value is a view on an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false\n */\nfunction isArrayBufferView(val) {\n var result;\n if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) {\n result = ArrayBuffer.isView(val);\n } else {\n result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer);\n }\n return result;\n}\n\n/**\n * Determine if a value is a String\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a String, otherwise false\n */\nfunction isString(val) {\n return typeof val === 'string';\n}\n\n/**\n * Determine if a value is a Number\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Number, otherwise false\n */\nfunction isNumber(val) {\n return typeof val === 'number';\n}\n\n/**\n * Determine if a value is an Object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Object, otherwise false\n */\nfunction isObject(val) {\n return val !== null && typeof val === 'object';\n}\n\n/**\n * Determine if a value is a plain Object\n *\n * @param {Object} val The value to test\n * @return {boolean} True if value is a plain Object, otherwise false\n */\nfunction isPlainObject(val) {\n if (toString.call(val) !== '[object Object]') {\n return false;\n }\n\n var prototype = Object.getPrototypeOf(val);\n return prototype === null || prototype === Object.prototype;\n}\n\n/**\n * Determine if a value is a Date\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Date, otherwise false\n */\nfunction isDate(val) {\n return toString.call(val) === '[object Date]';\n}\n\n/**\n * Determine if a value is a File\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a File, otherwise false\n */\nfunction isFile(val) {\n return toString.call(val) === '[object File]';\n}\n\n/**\n * Determine if a value is a Blob\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Blob, otherwise false\n */\nfunction isBlob(val) {\n return toString.call(val) === '[object Blob]';\n}\n\n/**\n * Determine if a value is a Function\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Function, otherwise false\n */\nfunction isFunction(val) {\n return toString.call(val) === '[object Function]';\n}\n\n/**\n * Determine if a value is a Stream\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Stream, otherwise false\n */\nfunction isStream(val) {\n return isObject(val) && isFunction(val.pipe);\n}\n\n/**\n * Determine if a value is a URLSearchParams object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a URLSearchParams object, otherwise false\n */\nfunction isURLSearchParams(val) {\n return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams;\n}\n\n/**\n * Trim excess whitespace off the beginning and end of a string\n *\n * @param {String} str The String to trim\n * @returns {String} The String freed of excess whitespace\n */\nfunction trim(str) {\n return str.replace(/^\\s*/, '').replace(/\\s*$/, '');\n}\n\n/**\n * Determine if we're running in a standard browser environment\n *\n * This allows axios to run in a web worker, and react-native.\n * Both environments support XMLHttpRequest, but not fully standard globals.\n *\n * web workers:\n * typeof window -> undefined\n * typeof document -> undefined\n *\n * react-native:\n * navigator.product -> 'ReactNative'\n * nativescript\n * navigator.product -> 'NativeScript' or 'NS'\n */\nfunction isStandardBrowserEnv() {\n if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' ||\n navigator.product === 'NativeScript' ||\n navigator.product === 'NS')) {\n return false;\n }\n return (\n typeof window !== 'undefined' &&\n typeof document !== 'undefined'\n );\n}\n\n/**\n * Iterate over an Array or an Object invoking a function for each item.\n *\n * If `obj` is an Array callback will be called passing\n * the value, index, and complete array for each item.\n *\n * If 'obj' is an Object callback will be called passing\n * the value, key, and complete object for each property.\n *\n * @param {Object|Array} obj The object to iterate\n * @param {Function} fn The callback to invoke for each item\n */\nfunction forEach(obj, fn) {\n // Don't bother if no value provided\n if (obj === null || typeof obj === 'undefined') {\n return;\n }\n\n // Force an array if not already something iterable\n if (typeof obj !== 'object') {\n /*eslint no-param-reassign:0*/\n obj = [obj];\n }\n\n if (isArray(obj)) {\n // Iterate over array values\n for (var i = 0, l = obj.length; i < l; i++) {\n fn.call(null, obj[i], i, obj);\n }\n } else {\n // Iterate over object keys\n for (var key in obj) {\n if (Object.prototype.hasOwnProperty.call(obj, key)) {\n fn.call(null, obj[key], key, obj);\n }\n }\n }\n}\n\n/**\n * Accepts varargs expecting each argument to be an object, then\n * immutably merges the properties of each object and returns result.\n *\n * When multiple objects contain the same key the later object in\n * the arguments list will take precedence.\n *\n * Example:\n *\n * ```js\n * var result = merge({foo: 123}, {foo: 456});\n * console.log(result.foo); // outputs 456\n * ```\n *\n * @param {Object} obj1 Object to merge\n * @returns {Object} Result of all merge properties\n */\nfunction merge(/* obj1, obj2, obj3, ... */) {\n var result = {};\n function assignValue(val, key) {\n if (isPlainObject(result[key]) && isPlainObject(val)) {\n result[key] = merge(result[key], val);\n } else if (isPlainObject(val)) {\n result[key] = merge({}, val);\n } else if (isArray(val)) {\n result[key] = val.slice();\n } else {\n result[key] = val;\n }\n }\n\n for (var i = 0, l = arguments.length; i < l; i++) {\n forEach(arguments[i], assignValue);\n }\n return result;\n}\n\n/**\n * Extends object a by mutably adding to it the properties of object b.\n *\n * @param {Object} a The object to be extended\n * @param {Object} b The object to copy properties from\n * @param {Object} thisArg The object to bind function to\n * @return {Object} The resulting value of object a\n */\nfunction extend(a, b, thisArg) {\n forEach(b, function assignValue(val, key) {\n if (thisArg && typeof val === 'function') {\n a[key] = bind(val, thisArg);\n } else {\n a[key] = val;\n }\n });\n return a;\n}\n\n/**\n * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM)\n *\n * @param {string} content with BOM\n * @return {string} content value without BOM\n */\nfunction stripBOM(content) {\n if (content.charCodeAt(0) === 0xFEFF) {\n content = content.slice(1);\n }\n return content;\n}\n\nmodule.exports = {\n isArray: isArray,\n isArrayBuffer: isArrayBuffer,\n isBuffer: isBuffer,\n isFormData: isFormData,\n isArrayBufferView: isArrayBufferView,\n isString: isString,\n isNumber: isNumber,\n isObject: isObject,\n isPlainObject: isPlainObject,\n isUndefined: isUndefined,\n isDate: isDate,\n isFile: isFile,\n isBlob: isBlob,\n isFunction: isFunction,\n isStream: isStream,\n isURLSearchParams: isURLSearchParams,\n isStandardBrowserEnv: isStandardBrowserEnv,\n forEach: forEach,\n merge: merge,\n extend: extend,\n trim: trim,\n stripBOM: stripBOM\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction encode(val) {\n return encodeURIComponent(val).\n replace(/%3A/gi, ':').\n replace(/%24/g, '$').\n replace(/%2C/gi, ',').\n replace(/%20/g, '+').\n replace(/%5B/gi, '[').\n replace(/%5D/gi, ']');\n}\n\n/**\n * Build a URL by appending params to the end\n *\n * @param {string} url The base of the url (e.g., http://www.google.com)\n * @param {object} [params] The params to be appended\n * @returns {string} The formatted url\n */\nmodule.exports = function buildURL(url, params, paramsSerializer) {\n /*eslint no-param-reassign:0*/\n if (!params) {\n return url;\n }\n\n var serializedParams;\n if (paramsSerializer) {\n serializedParams = paramsSerializer(params);\n } else if (utils.isURLSearchParams(params)) {\n serializedParams = params.toString();\n } else {\n var parts = [];\n\n utils.forEach(params, function serialize(val, key) {\n if (val === null || typeof val === 'undefined') {\n return;\n }\n\n if (utils.isArray(val)) {\n key = key + '[]';\n } else {\n val = [val];\n }\n\n utils.forEach(val, function parseValue(v) {\n if (utils.isDate(v)) {\n v = v.toISOString();\n } else if (utils.isObject(v)) {\n v = JSON.stringify(v);\n }\n parts.push(encode(key) + '=' + encode(v));\n });\n });\n\n serializedParams = parts.join('&');\n }\n\n if (serializedParams) {\n var hashmarkIndex = url.indexOf('#');\n if (hashmarkIndex !== -1) {\n url = url.slice(0, hashmarkIndex);\n }\n\n url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams;\n }\n\n return url;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction InterceptorManager() {\n this.handlers = [];\n}\n\n/**\n * Add a new interceptor to the stack\n *\n * @param {Function} fulfilled The function to handle `then` for a `Promise`\n * @param {Function} rejected The function to handle `reject` for a `Promise`\n *\n * @return {Number} An ID used to remove interceptor later\n */\nInterceptorManager.prototype.use = function use(fulfilled, rejected) {\n this.handlers.push({\n fulfilled: fulfilled,\n rejected: rejected\n });\n return this.handlers.length - 1;\n};\n\n/**\n * Remove an interceptor from the stack\n *\n * @param {Number} id The ID that was returned by `use`\n */\nInterceptorManager.prototype.eject = function eject(id) {\n if (this.handlers[id]) {\n this.handlers[id] = null;\n }\n};\n\n/**\n * Iterate over all the registered interceptors\n *\n * This method is particularly useful for skipping over any\n * interceptors that may have become `null` calling `eject`.\n *\n * @param {Function} fn The function to call for each interceptor\n */\nInterceptorManager.prototype.forEach = function forEach(fn) {\n utils.forEach(this.handlers, function forEachHandler(h) {\n if (h !== null) {\n fn(h);\n }\n });\n};\n\nmodule.exports = InterceptorManager;\n","'use strict';\n\nvar utils = require('./../utils');\n\n/**\n * Transform the data for a request or a response\n *\n * @param {Object|String} data The data to be transformed\n * @param {Array} headers The headers for the request or response\n * @param {Array|Function} fns A single function or Array of functions\n * @returns {*} The resulting transformed data\n */\nmodule.exports = function transformData(data, headers, fns) {\n /*eslint no-param-reassign:0*/\n utils.forEach(fns, function transform(fn) {\n data = fn(data, headers);\n });\n\n return data;\n};\n","'use strict';\n\nmodule.exports = function isCancel(value) {\n return !!(value && value.__CANCEL__);\n};\n","'use strict';\n\nvar utils = require('../utils');\n\nmodule.exports = function normalizeHeaderName(headers, normalizedName) {\n utils.forEach(headers, function processHeader(value, name) {\n if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) {\n headers[normalizedName] = value;\n delete headers[name];\n }\n });\n};\n","'use strict';\n\n/**\n * Update an Error with the specified config, error code, and response.\n *\n * @param {Error} error The error to update.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The error.\n */\nmodule.exports = function enhanceError(error, config, code, request, response) {\n error.config = config;\n if (code) {\n error.code = code;\n }\n\n error.request = request;\n error.response = response;\n error.isAxiosError = true;\n\n error.toJSON = function toJSON() {\n return {\n // Standard\n message: this.message,\n name: this.name,\n // Microsoft\n description: this.description,\n number: this.number,\n // Mozilla\n fileName: this.fileName,\n lineNumber: this.lineNumber,\n columnNumber: this.columnNumber,\n stack: this.stack,\n // Axios\n config: this.config,\n code: this.code\n };\n };\n return error;\n};\n","'use strict';\n\nvar enhanceError = require('./enhanceError');\n\n/**\n * Create an Error with the specified message, config, error code, request and response.\n *\n * @param {string} message The error message.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The created error.\n */\nmodule.exports = function createError(message, config, code, request, response) {\n var error = new Error(message);\n return enhanceError(error, config, code, request, response);\n};\n","'use strict';\n\nvar createError = require('./createError');\n\n/**\n * Resolve or reject a Promise based on response status.\n *\n * @param {Function} resolve A function that resolves the promise.\n * @param {Function} reject A function that rejects the promise.\n * @param {object} response The response.\n */\nmodule.exports = function settle(resolve, reject, response) {\n var validateStatus = response.config.validateStatus;\n if (!response.status || !validateStatus || validateStatus(response.status)) {\n resolve(response);\n } else {\n reject(createError(\n 'Request failed with status code ' + response.status,\n response.config,\n null,\n response.request,\n response\n ));\n }\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs support document.cookie\n (function standardBrowserEnv() {\n return {\n write: function write(name, value, expires, path, domain, secure) {\n var cookie = [];\n cookie.push(name + '=' + encodeURIComponent(value));\n\n if (utils.isNumber(expires)) {\n cookie.push('expires=' + new Date(expires).toGMTString());\n }\n\n if (utils.isString(path)) {\n cookie.push('path=' + path);\n }\n\n if (utils.isString(domain)) {\n cookie.push('domain=' + domain);\n }\n\n if (secure === true) {\n cookie.push('secure');\n }\n\n document.cookie = cookie.join('; ');\n },\n\n read: function read(name) {\n var match = document.cookie.match(new RegExp('(^|;\\\\s*)(' + name + ')=([^;]*)'));\n return (match ? decodeURIComponent(match[3]) : null);\n },\n\n remove: function remove(name) {\n this.write(name, '', Date.now() - 86400000);\n }\n };\n })() :\n\n // Non standard browser env (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return {\n write: function write() {},\n read: function read() { return null; },\n remove: function remove() {}\n };\n })()\n);\n","'use strict';\n\n/**\n * Determines whether the specified URL is absolute\n *\n * @param {string} url The URL to test\n * @returns {boolean} True if the specified URL is absolute, otherwise false\n */\nmodule.exports = function isAbsoluteURL(url) {\n // A URL is considered absolute if it begins with \"://\" or \"//\" (protocol-relative URL).\n // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed\n // by any combination of letters, digits, plus, period, or hyphen.\n return /^([a-z][a-z\\d\\+\\-\\.]*:)?\\/\\//i.test(url);\n};\n","'use strict';\n\n/**\n * Creates a new URL by combining the specified URLs\n *\n * @param {string} baseURL The base URL\n * @param {string} relativeURL The relative URL\n * @returns {string} The combined URL\n */\nmodule.exports = function combineURLs(baseURL, relativeURL) {\n return relativeURL\n ? baseURL.replace(/\\/+$/, '') + '/' + relativeURL.replace(/^\\/+/, '')\n : baseURL;\n};\n","'use strict';\n\nvar isAbsoluteURL = require('../helpers/isAbsoluteURL');\nvar combineURLs = require('../helpers/combineURLs');\n\n/**\n * Creates a new URL by combining the baseURL with the requestedURL,\n * only when the requestedURL is not already an absolute URL.\n * If the requestURL is absolute, this function returns the requestedURL untouched.\n *\n * @param {string} baseURL The base URL\n * @param {string} requestedURL Absolute or relative URL to combine\n * @returns {string} The combined full path\n */\nmodule.exports = function buildFullPath(baseURL, requestedURL) {\n if (baseURL && !isAbsoluteURL(requestedURL)) {\n return combineURLs(baseURL, requestedURL);\n }\n return requestedURL;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\n// Headers whose duplicates are ignored by node\n// c.f. https://nodejs.org/api/http.html#http_message_headers\nvar ignoreDuplicateOf = [\n 'age', 'authorization', 'content-length', 'content-type', 'etag',\n 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since',\n 'last-modified', 'location', 'max-forwards', 'proxy-authorization',\n 'referer', 'retry-after', 'user-agent'\n];\n\n/**\n * Parse headers into an object\n *\n * ```\n * Date: Wed, 27 Aug 2014 08:58:49 GMT\n * Content-Type: application/json\n * Connection: keep-alive\n * Transfer-Encoding: chunked\n * ```\n *\n * @param {String} headers Headers needing to be parsed\n * @returns {Object} Headers parsed into an object\n */\nmodule.exports = function parseHeaders(headers) {\n var parsed = {};\n var key;\n var val;\n var i;\n\n if (!headers) { return parsed; }\n\n utils.forEach(headers.split('\\n'), function parser(line) {\n i = line.indexOf(':');\n key = utils.trim(line.substr(0, i)).toLowerCase();\n val = utils.trim(line.substr(i + 1));\n\n if (key) {\n if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) {\n return;\n }\n if (key === 'set-cookie') {\n parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]);\n } else {\n parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val;\n }\n }\n });\n\n return parsed;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs have full support of the APIs needed to test\n // whether the request URL is of the same origin as current location.\n (function standardBrowserEnv() {\n var msie = /(msie|trident)/i.test(navigator.userAgent);\n var urlParsingNode = document.createElement('a');\n var originURL;\n\n /**\n * Parse a URL to discover it's components\n *\n * @param {String} url The URL to be parsed\n * @returns {Object}\n */\n function resolveURL(url) {\n var href = url;\n\n if (msie) {\n // IE needs attribute set twice to normalize properties\n urlParsingNode.setAttribute('href', href);\n href = urlParsingNode.href;\n }\n\n urlParsingNode.setAttribute('href', href);\n\n // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils\n return {\n href: urlParsingNode.href,\n protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '',\n host: urlParsingNode.host,\n search: urlParsingNode.search ? urlParsingNode.search.replace(/^\\?/, '') : '',\n hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '',\n hostname: urlParsingNode.hostname,\n port: urlParsingNode.port,\n pathname: (urlParsingNode.pathname.charAt(0) === '/') ?\n urlParsingNode.pathname :\n '/' + urlParsingNode.pathname\n };\n }\n\n originURL = resolveURL(window.location.href);\n\n /**\n * Determine if a URL shares the same origin as the current location\n *\n * @param {String} requestURL The URL to test\n * @returns {boolean} True if URL shares the same origin, otherwise false\n */\n return function isURLSameOrigin(requestURL) {\n var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL;\n return (parsed.protocol === originURL.protocol &&\n parsed.host === originURL.host);\n };\n })() :\n\n // Non standard browser envs (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return function isURLSameOrigin() {\n return true;\n };\n })()\n);\n","'use strict';\n\nvar utils = require('./../utils');\nvar settle = require('./../core/settle');\nvar cookies = require('./../helpers/cookies');\nvar buildURL = require('./../helpers/buildURL');\nvar buildFullPath = require('../core/buildFullPath');\nvar parseHeaders = require('./../helpers/parseHeaders');\nvar isURLSameOrigin = require('./../helpers/isURLSameOrigin');\nvar createError = require('../core/createError');\n\nmodule.exports = function xhrAdapter(config) {\n return new Promise(function dispatchXhrRequest(resolve, reject) {\n var requestData = config.data;\n var requestHeaders = config.headers;\n\n if (utils.isFormData(requestData)) {\n delete requestHeaders['Content-Type']; // Let the browser set it\n }\n\n var request = new XMLHttpRequest();\n\n // HTTP basic authentication\n if (config.auth) {\n var username = config.auth.username || '';\n var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : '';\n requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password);\n }\n\n var fullPath = buildFullPath(config.baseURL, config.url);\n request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true);\n\n // Set the request timeout in MS\n request.timeout = config.timeout;\n\n // Listen for ready state\n request.onreadystatechange = function handleLoad() {\n if (!request || request.readyState !== 4) {\n return;\n }\n\n // The request errored out and we didn't get a response, this will be\n // handled by onerror instead\n // With one exception: request that using file: protocol, most browsers\n // will return status as 0 even though it's a successful request\n if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) {\n return;\n }\n\n // Prepare the response\n var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null;\n var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response;\n var response = {\n data: responseData,\n status: request.status,\n statusText: request.statusText,\n headers: responseHeaders,\n config: config,\n request: request\n };\n\n settle(resolve, reject, response);\n\n // Clean up request\n request = null;\n };\n\n // Handle browser request cancellation (as opposed to a manual cancellation)\n request.onabort = function handleAbort() {\n if (!request) {\n return;\n }\n\n reject(createError('Request aborted', config, 'ECONNABORTED', request));\n\n // Clean up request\n request = null;\n };\n\n // Handle low level network errors\n request.onerror = function handleError() {\n // Real errors are hidden from us by the browser\n // onerror should only fire if it's a network error\n reject(createError('Network Error', config, null, request));\n\n // Clean up request\n request = null;\n };\n\n // Handle timeout\n request.ontimeout = function handleTimeout() {\n var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded';\n if (config.timeoutErrorMessage) {\n timeoutErrorMessage = config.timeoutErrorMessage;\n }\n reject(createError(timeoutErrorMessage, config, 'ECONNABORTED',\n request));\n\n // Clean up request\n request = null;\n };\n\n // Add xsrf header\n // This is only done if running in a standard browser environment.\n // Specifically not if we're in a web worker, or react-native.\n if (utils.isStandardBrowserEnv()) {\n // Add xsrf header\n var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ?\n cookies.read(config.xsrfCookieName) :\n undefined;\n\n if (xsrfValue) {\n requestHeaders[config.xsrfHeaderName] = xsrfValue;\n }\n }\n\n // Add headers to the request\n if ('setRequestHeader' in request) {\n utils.forEach(requestHeaders, function setRequestHeader(val, key) {\n if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') {\n // Remove Content-Type if data is undefined\n delete requestHeaders[key];\n } else {\n // Otherwise add header to the request\n request.setRequestHeader(key, val);\n }\n });\n }\n\n // Add withCredentials to request if needed\n if (!utils.isUndefined(config.withCredentials)) {\n request.withCredentials = !!config.withCredentials;\n }\n\n // Add responseType to request if needed\n if (config.responseType) {\n try {\n request.responseType = config.responseType;\n } catch (e) {\n // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2.\n // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function.\n if (config.responseType !== 'json') {\n throw e;\n }\n }\n }\n\n // Handle progress if needed\n if (typeof config.onDownloadProgress === 'function') {\n request.addEventListener('progress', config.onDownloadProgress);\n }\n\n // Not all browsers support upload events\n if (typeof config.onUploadProgress === 'function' && request.upload) {\n request.upload.addEventListener('progress', config.onUploadProgress);\n }\n\n if (config.cancelToken) {\n // Handle cancellation\n config.cancelToken.promise.then(function onCanceled(cancel) {\n if (!request) {\n return;\n }\n\n request.abort();\n reject(cancel);\n // Clean up request\n request = null;\n });\n }\n\n if (!requestData) {\n requestData = null;\n }\n\n // Send the request\n request.send(requestData);\n });\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar normalizeHeaderName = require('./helpers/normalizeHeaderName');\n\nvar DEFAULT_CONTENT_TYPE = {\n 'Content-Type': 'application/x-www-form-urlencoded'\n};\n\nfunction setContentTypeIfUnset(headers, value) {\n if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) {\n headers['Content-Type'] = value;\n }\n}\n\nfunction getDefaultAdapter() {\n var adapter;\n if (typeof XMLHttpRequest !== 'undefined') {\n // For browsers use XHR adapter\n adapter = require('./adapters/xhr');\n } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') {\n // For node use HTTP adapter\n adapter = require('./adapters/http');\n }\n return adapter;\n}\n\nvar defaults = {\n adapter: getDefaultAdapter(),\n\n transformRequest: [function transformRequest(data, headers) {\n normalizeHeaderName(headers, 'Accept');\n normalizeHeaderName(headers, 'Content-Type');\n if (utils.isFormData(data) ||\n utils.isArrayBuffer(data) ||\n utils.isBuffer(data) ||\n utils.isStream(data) ||\n utils.isFile(data) ||\n utils.isBlob(data)\n ) {\n return data;\n }\n if (utils.isArrayBufferView(data)) {\n return data.buffer;\n }\n if (utils.isURLSearchParams(data)) {\n setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8');\n return data.toString();\n }\n if (utils.isObject(data)) {\n setContentTypeIfUnset(headers, 'application/json;charset=utf-8');\n return JSON.stringify(data);\n }\n return data;\n }],\n\n transformResponse: [function transformResponse(data) {\n /*eslint no-param-reassign:0*/\n if (typeof data === 'string') {\n try {\n data = JSON.parse(data);\n } catch (e) { /* Ignore */ }\n }\n return data;\n }],\n\n /**\n * A timeout in milliseconds to abort a request. If set to 0 (default) a\n * timeout is not created.\n */\n timeout: 0,\n\n xsrfCookieName: 'XSRF-TOKEN',\n xsrfHeaderName: 'X-XSRF-TOKEN',\n\n maxContentLength: -1,\n maxBodyLength: -1,\n\n validateStatus: function validateStatus(status) {\n return status >= 200 && status < 300;\n }\n};\n\ndefaults.headers = {\n common: {\n 'Accept': 'application/json, text/plain, */*'\n }\n};\n\nutils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) {\n defaults.headers[method] = {};\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE);\n});\n\nmodule.exports = defaults;\n","'use strict';\n\nvar utils = require('./../utils');\nvar transformData = require('./transformData');\nvar isCancel = require('../cancel/isCancel');\nvar defaults = require('../defaults');\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nfunction throwIfCancellationRequested(config) {\n if (config.cancelToken) {\n config.cancelToken.throwIfRequested();\n }\n}\n\n/**\n * Dispatch a request to the server using the configured adapter.\n *\n * @param {object} config The config that is to be used for the request\n * @returns {Promise} The Promise to be fulfilled\n */\nmodule.exports = function dispatchRequest(config) {\n throwIfCancellationRequested(config);\n\n // Ensure headers exist\n config.headers = config.headers || {};\n\n // Transform request data\n config.data = transformData(\n config.data,\n config.headers,\n config.transformRequest\n );\n\n // Flatten headers\n config.headers = utils.merge(\n config.headers.common || {},\n config.headers[config.method] || {},\n config.headers\n );\n\n utils.forEach(\n ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'],\n function cleanHeaderConfig(method) {\n delete config.headers[method];\n }\n );\n\n var adapter = config.adapter || defaults.adapter;\n\n return adapter(config).then(function onAdapterResolution(response) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n response.data = transformData(\n response.data,\n response.headers,\n config.transformResponse\n );\n\n return response;\n }, function onAdapterRejection(reason) {\n if (!isCancel(reason)) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n if (reason && reason.response) {\n reason.response.data = transformData(\n reason.response.data,\n reason.response.headers,\n config.transformResponse\n );\n }\n }\n\n return Promise.reject(reason);\n });\n};\n","'use strict';\n\nvar utils = require('../utils');\n\n/**\n * Config-specific merge-function which creates a new config-object\n * by merging two configuration objects together.\n *\n * @param {Object} config1\n * @param {Object} config2\n * @returns {Object} New object resulting from merging config2 to config1\n */\nmodule.exports = function mergeConfig(config1, config2) {\n // eslint-disable-next-line no-param-reassign\n config2 = config2 || {};\n var config = {};\n\n var valueFromConfig2Keys = ['url', 'method', 'data'];\n var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params'];\n var defaultToConfig2Keys = [\n 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer',\n 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName',\n 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress',\n 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent',\n 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding'\n ];\n var directMergeKeys = ['validateStatus'];\n\n function getMergedValue(target, source) {\n if (utils.isPlainObject(target) && utils.isPlainObject(source)) {\n return utils.merge(target, source);\n } else if (utils.isPlainObject(source)) {\n return utils.merge({}, source);\n } else if (utils.isArray(source)) {\n return source.slice();\n }\n return source;\n }\n\n function mergeDeepProperties(prop) {\n if 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Events = {\n\tWEBRTC_NOT_SUPPORT : 'WEBRTC_NOT_SUPPORT',\n\tWEBRTC_ICE_CANDIDATE_ERROR : 'WEBRTC_ICE_CANDIDATE_ERROR',\n\tWEBRTC_OFFER_ANWSER_EXCHANGE_FAILED:'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED',\n\tWEBRTC_ON_REMOTE_STREAMS:'WEBRTC_ON_REMOTE_STREAMS',\n\tWEBRTC_ON_LOCAL_STREAM:'WEBRTC_ON_LOCAL_STREAM',\n\tCAPTURE_STREAM_FAILED:'CAPTURE_STREAM_FAILED'\n};\n\nexport default Events;","export const VERSION = '__VERSION__';\nexport const BUILD_DATE = '__BUILD_DATE__';","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n\n// eslint-disable-next-line require-jsdoc\nexport function isFirefox() {\n return window.navigator.userAgent.match('Firefox') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isChrome() {\n return window.navigator.userAgent.match('Chrome') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isSafari() {\n return /^((?!chrome|android).)*safari/i.test(window.navigator.userAgent);\n}\n// eslint-disable-next-line require-jsdoc\nexport function isEdge() {\n return window.navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/) !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function createUuid() {\n return 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx'.replace(/[xy]/g, function(c) {\n const r = Math.random() * 16 | 0;\n const v = c === 'x' ? r : (r & 0x3 | 0x8);\n return v.toString(16);\n });\n}","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n/**\n * @class AudioSourceInfo\n * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const AudioSourceInfo = {\n MIC: 'mic',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class VideoSourceInfo\n * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const VideoSourceInfo = {\n CAMERA: 'camera',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class TrackKind\n * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const TrackKind = {\n /**\n * Audio tracks.\n * @type string\n */\n AUDIO: 'audio',\n /**\n * Video tracks.\n * @type string\n */\n VIDEO: 'video',\n /**\n * Both audio and video tracks.\n * @type string\n */\n AUDIO_AND_VIDEO: 'av',\n};\n/**\n * @class Resolution\n * @memberOf Owt.Base\n * @classDesc The Resolution defines the size of a rectangle.\n * @constructor\n * @param {number} width\n * @param {number} height\n */\nexport class Resolution {\n // eslint-disable-next-line require-jsdoc\n constructor(width, height) {\n /**\n * @member {number} width\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.width = width;\n /**\n * @member {number} height\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.height = height;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nlet logDisabled_ = true;\nlet deprecationWarnings_ = true;\n\n/**\n * Extract browser version out of the provided user agent string.\n *\n * @param {!string} uastring userAgent string.\n * @param {!string} expr Regular expression used as match criteria.\n * @param {!number} pos position in the version string to be returned.\n * @return {!number} browser version.\n */\nexport function extractVersion(uastring, expr, pos) {\n const match = uastring.match(expr);\n return match && match.length >= pos && parseInt(match[pos], 10);\n}\n\n// Wraps the peerconnection event eventNameToWrap in a function\n// which returns the modified event object (or false to prevent\n// the event).\nexport function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) {\n if (!window.RTCPeerConnection) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n const nativeAddEventListener = proto.addEventListener;\n proto.addEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap) {\n return nativeAddEventListener.apply(this, arguments);\n }\n const wrappedCallback = (e) => {\n const modifiedEvent = wrapper(e);\n if (modifiedEvent) {\n if (cb.handleEvent) {\n cb.handleEvent(modifiedEvent);\n } else {\n cb(modifiedEvent);\n }\n }\n };\n this._eventMap = this._eventMap || {};\n if (!this._eventMap[eventNameToWrap]) {\n this._eventMap[eventNameToWrap] = new Map();\n }\n this._eventMap[eventNameToWrap].set(cb, wrappedCallback);\n return nativeAddEventListener.apply(this, [nativeEventName,\n wrappedCallback]);\n };\n\n const nativeRemoveEventListener = proto.removeEventListener;\n proto.removeEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap || !this._eventMap\n || !this._eventMap[eventNameToWrap]) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n if (!this._eventMap[eventNameToWrap].has(cb)) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n const unwrappedCb = this._eventMap[eventNameToWrap].get(cb);\n this._eventMap[eventNameToWrap].delete(cb);\n if (this._eventMap[eventNameToWrap].size === 0) {\n delete this._eventMap[eventNameToWrap];\n }\n if (Object.keys(this._eventMap).length === 0) {\n delete this._eventMap;\n }\n return nativeRemoveEventListener.apply(this, [nativeEventName,\n unwrappedCb]);\n };\n\n Object.defineProperty(proto, 'on' + eventNameToWrap, {\n get() {\n return this['_on' + eventNameToWrap];\n },\n set(cb) {\n if (this['_on' + eventNameToWrap]) {\n this.removeEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap]);\n delete this['_on' + eventNameToWrap];\n }\n if (cb) {\n this.addEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap] = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n}\n\nexport function disableLog(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n logDisabled_ = bool;\n return (bool) ? 'adapter.js logging disabled' :\n 'adapter.js logging enabled';\n}\n\n/**\n * Disable or enable deprecation warnings\n * @param {!boolean} bool set to true to disable warnings.\n */\nexport function disableWarnings(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n deprecationWarnings_ = !bool;\n return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled');\n}\n\nexport function log() {\n if (typeof window === 'object') {\n if (logDisabled_) {\n return;\n }\n if (typeof console !== 'undefined' && typeof console.log === 'function') {\n console.log.apply(console, arguments);\n }\n }\n}\n\n/**\n * Shows a deprecation warning suggesting the modern and spec-compatible API.\n */\nexport function deprecated(oldMethod, newMethod) {\n if (!deprecationWarnings_) {\n return;\n }\n console.warn(oldMethod + ' is deprecated, please use ' + newMethod +\n ' instead.');\n}\n\n/**\n * Browser detector.\n *\n * @return {object} result containing browser and version\n * properties.\n */\nexport function detectBrowser(window) {\n // Returned result object.\n const result = {browser: null, version: null};\n\n // Fail early if it's not a browser\n if (typeof window === 'undefined' || !window.navigator) {\n result.browser = 'Not a browser.';\n return result;\n }\n\n const {navigator} = window;\n\n if (navigator.mozGetUserMedia) { // Firefox.\n result.browser = 'firefox';\n result.version = extractVersion(navigator.userAgent,\n /Firefox\\/(\\d+)\\./, 1);\n } else if (navigator.webkitGetUserMedia ||\n (window.isSecureContext === false && window.webkitRTCPeerConnection &&\n !window.RTCIceGatherer)) {\n // Chrome, Chromium, Webview, Opera.\n // Version matches Chrome/WebRTC version.\n // Chrome 74 removed webkitGetUserMedia on http as well so we need the\n // more complicated fallback to webkitRTCPeerConnection.\n result.browser = 'chrome';\n result.version = extractVersion(navigator.userAgent,\n /Chrom(e|ium)\\/(\\d+)\\./, 2);\n } else if (navigator.mediaDevices &&\n navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/)) { // Edge.\n result.browser = 'edge';\n result.version = extractVersion(navigator.userAgent,\n /Edge\\/(\\d+).(\\d+)$/, 2);\n } else if (window.RTCPeerConnection &&\n navigator.userAgent.match(/AppleWebKit\\/(\\d+)\\./)) { // Safari.\n result.browser = 'safari';\n result.version = extractVersion(navigator.userAgent,\n /AppleWebKit\\/(\\d+)\\./, 1);\n result.supportsUnifiedPlan = window.RTCRtpTransceiver &&\n 'currentDirection' in window.RTCRtpTransceiver.prototype;\n } else { // Default fallthrough: not supported.\n result.browser = 'Not a supported browser.';\n return result;\n }\n\n return result;\n}\n\n/**\n * Checks if something is an object.\n *\n * @param {*} val The something you want to check.\n * @return true if val is an object, false otherwise.\n */\nfunction isObject(val) {\n return Object.prototype.toString.call(val) === '[object Object]';\n}\n\n/**\n * Remove all empty objects and undefined values\n * from a nested object -- an enhanced and vanilla version\n * of Lodash's `compact`.\n */\nexport function compactObject(data) {\n if (!isObject(data)) {\n return data;\n }\n\n return Object.keys(data).reduce(function(accumulator, key) {\n const isObj = isObject(data[key]);\n const value = isObj ? compactObject(data[key]) : data[key];\n const isEmptyObject = isObj && !Object.keys(value).length;\n if (value === undefined || isEmptyObject) {\n return accumulator;\n }\n return Object.assign(accumulator, {[key]: value});\n }, {});\n}\n\n/* iterates the stats graph recursively. */\nexport function walkStats(stats, base, resultSet) {\n if (!base || resultSet.has(base.id)) {\n return;\n }\n resultSet.set(base.id, base);\n Object.keys(base).forEach(name => {\n if (name.endsWith('Id')) {\n walkStats(stats, stats.get(base[name]), resultSet);\n } else if (name.endsWith('Ids')) {\n base[name].forEach(id => {\n walkStats(stats, stats.get(id), resultSet);\n });\n }\n });\n}\n\n/* filter getStats for a sender/receiver track. */\nexport function filterStats(result, track, outbound) {\n const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp';\n const filteredResult = new Map();\n if (track === null) {\n return filteredResult;\n }\n const trackStats = [];\n result.forEach(value => {\n if (value.type === 'track' &&\n value.trackIdentifier === track.id) {\n trackStats.push(value);\n }\n });\n trackStats.forEach(trackStat => {\n result.forEach(stats => {\n if (stats.type === streamStatsType && stats.trackId === trackStat.id) {\n walkStats(result, stats, filteredResult);\n }\n });\n });\n return filteredResult;\n}\n\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\nconst logging = utils.log;\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n\n if (!navigator.mediaDevices) {\n return;\n }\n\n const constraintsToChrome_ = function(c) {\n if (typeof c !== 'object' || c.mandatory || c.optional) {\n return c;\n }\n const cc = {};\n Object.keys(c).forEach(key => {\n if (key === 'require' || key === 'advanced' || key === 'mediaSource') {\n return;\n }\n const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]};\n if (r.exact !== undefined && typeof r.exact === 'number') {\n r.min = r.max = r.exact;\n }\n const oldname_ = function(prefix, name) {\n if (prefix) {\n return prefix + name.charAt(0).toUpperCase() + name.slice(1);\n }\n return (name === 'deviceId') ? 'sourceId' : name;\n };\n if (r.ideal !== undefined) {\n cc.optional = cc.optional || [];\n let oc = {};\n if (typeof r.ideal === 'number') {\n oc[oldname_('min', key)] = r.ideal;\n cc.optional.push(oc);\n oc = {};\n oc[oldname_('max', key)] = r.ideal;\n cc.optional.push(oc);\n } else {\n oc[oldname_('', key)] = r.ideal;\n cc.optional.push(oc);\n }\n }\n if (r.exact !== undefined && typeof r.exact !== 'number') {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_('', key)] = r.exact;\n } else {\n ['min', 'max'].forEach(mix => {\n if (r[mix] !== undefined) {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_(mix, key)] = r[mix];\n }\n });\n }\n });\n if (c.advanced) {\n cc.optional = (cc.optional || []).concat(c.advanced);\n }\n return cc;\n };\n\n const shimConstraints_ = function(constraints, func) {\n if (browserDetails.version >= 61) {\n return func(constraints);\n }\n constraints = JSON.parse(JSON.stringify(constraints));\n if (constraints && typeof constraints.audio === 'object') {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n constraints = JSON.parse(JSON.stringify(constraints));\n remap(constraints.audio, 'autoGainControl', 'googAutoGainControl');\n remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression');\n constraints.audio = constraintsToChrome_(constraints.audio);\n }\n if (constraints && typeof constraints.video === 'object') {\n // Shim facingMode for mobile & surface pro.\n let face = constraints.video.facingMode;\n face = face && ((typeof face === 'object') ? face : {ideal: face});\n const getSupportedFacingModeLies = browserDetails.version < 66;\n\n if ((face && (face.exact === 'user' || face.exact === 'environment' ||\n face.ideal === 'user' || face.ideal === 'environment')) &&\n !(navigator.mediaDevices.getSupportedConstraints &&\n navigator.mediaDevices.getSupportedConstraints().facingMode &&\n !getSupportedFacingModeLies)) {\n delete constraints.video.facingMode;\n let matches;\n if (face.exact === 'environment' || face.ideal === 'environment') {\n matches = ['back', 'rear'];\n } else if (face.exact === 'user' || face.ideal === 'user') {\n matches = ['front'];\n }\n if (matches) {\n // Look for matches in label, or use last cam for back (typical).\n return navigator.mediaDevices.enumerateDevices()\n .then(devices => {\n devices = devices.filter(d => d.kind === 'videoinput');\n let dev = devices.find(d => matches.some(match =>\n d.label.toLowerCase().includes(match)));\n if (!dev && devices.length && matches.includes('back')) {\n dev = devices[devices.length - 1]; // more likely the back cam\n }\n if (dev) {\n constraints.video.deviceId = face.exact ? {exact: dev.deviceId} :\n {ideal: dev.deviceId};\n }\n constraints.video = constraintsToChrome_(constraints.video);\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n });\n }\n }\n constraints.video = constraintsToChrome_(constraints.video);\n }\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n };\n\n const shimError_ = function(e) {\n if (browserDetails.version >= 64) {\n return e;\n }\n return {\n name: {\n PermissionDeniedError: 'NotAllowedError',\n PermissionDismissedError: 'NotAllowedError',\n InvalidStateError: 'NotAllowedError',\n DevicesNotFoundError: 'NotFoundError',\n ConstraintNotSatisfiedError: 'OverconstrainedError',\n TrackStartError: 'NotReadableError',\n MediaDeviceFailedDueToShutdown: 'NotAllowedError',\n MediaDeviceKillSwitchOn: 'NotAllowedError',\n TabCaptureError: 'AbortError',\n ScreenCaptureError: 'AbortError',\n DeviceCaptureError: 'AbortError'\n }[e.name] || e.name,\n message: e.message,\n constraint: e.constraint || e.constraintName,\n toString() {\n return this.name + (this.message && ': ') + this.message;\n }\n };\n };\n\n const getUserMedia_ = function(constraints, onSuccess, onError) {\n shimConstraints_(constraints, c => {\n navigator.webkitGetUserMedia(c, onSuccess, e => {\n if (onError) {\n onError(shimError_(e));\n }\n });\n });\n };\n navigator.getUserMedia = getUserMedia_.bind(navigator);\n\n // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia\n // function which returns a Promise, it does not accept spec-style\n // constraints.\n if (navigator.mediaDevices.getUserMedia) {\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(cs) {\n return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => {\n if (c.audio && !stream.getAudioTracks().length ||\n c.video && !stream.getVideoTracks().length) {\n stream.getTracks().forEach(track => {\n track.stop();\n });\n throw new DOMException('', 'NotFoundError');\n }\n return stream;\n }, e => Promise.reject(shimError_(e))));\n };\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nexport function shimGetDisplayMedia(window, getSourceId) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n // getSourceId is a function that returns a promise resolving with\n // the sourceId of the screen/window/tab to be shared.\n if (typeof getSourceId !== 'function') {\n console.error('shimGetDisplayMedia: getSourceId argument is not ' +\n 'a function');\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n return getSourceId(constraints)\n .then(sourceId => {\n const widthSpecified = constraints.video && constraints.video.width;\n const heightSpecified = constraints.video &&\n constraints.video.height;\n const frameRateSpecified = constraints.video &&\n constraints.video.frameRate;\n constraints.video = {\n mandatory: {\n chromeMediaSource: 'desktop',\n chromeMediaSourceId: sourceId,\n maxFrameRate: frameRateSpecified || 3\n }\n };\n if (widthSpecified) {\n constraints.video.mandatory.maxWidth = widthSpecified;\n }\n if (heightSpecified) {\n constraints.video.mandatory.maxHeight = heightSpecified;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n });\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimMediaStream(window) {\n window.MediaStream = window.MediaStream || window.webkitMediaStream;\n}\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in\n window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', {\n get() {\n return this._ontrack;\n },\n set(f) {\n if (this._ontrack) {\n this.removeEventListener('track', this._ontrack);\n }\n this.addEventListener('track', this._ontrack = f);\n },\n enumerable: true,\n configurable: true\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n if (!this._ontrackpoly) {\n this._ontrackpoly = (e) => {\n // onaddstream does not fire when a track is added to an existing\n // stream. But stream.onaddtrack is implemented so we use that.\n e.stream.addEventListener('addtrack', te => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === te.track.id);\n } else {\n receiver = {track: te.track};\n }\n\n const event = new Event('track');\n event.track = te.track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n e.stream.getTracks().forEach(track => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === track.id);\n } else {\n receiver = {track};\n }\n const event = new Event('track');\n event.track = track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n };\n this.addEventListener('addstream', this._ontrackpoly);\n }\n return origSetRemoteDescription.apply(this, arguments);\n };\n } else {\n // even if RTCRtpTransceiver is in window, it is only used and\n // emitted in unified-plan. Unfortunately this means we need\n // to unconditionally wrap the event.\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n if (!e.transceiver) {\n Object.defineProperty(e, 'transceiver',\n {value: {receiver: e.receiver}});\n }\n return e;\n });\n }\n}\n\nexport function shimGetSendersWithDtmf(window) {\n // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack.\n if (typeof window === 'object' && window.RTCPeerConnection &&\n !('getSenders' in window.RTCPeerConnection.prototype) &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype) {\n const shimSenderWithDtmf = function(pc, track) {\n return {\n track,\n get dtmf() {\n if (this._dtmf === undefined) {\n if (track.kind === 'audio') {\n this._dtmf = pc.createDTMFSender(track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n },\n _pc: pc\n };\n };\n\n // augment addTrack when getSenders is not available.\n if (!window.RTCPeerConnection.prototype.getSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n this._senders = this._senders || [];\n return this._senders.slice(); // return a copy of the internal state.\n };\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n let sender = origAddTrack.apply(this, arguments);\n if (!sender) {\n sender = shimSenderWithDtmf(this, track);\n this._senders.push(sender);\n }\n return sender;\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n origRemoveTrack.apply(this, arguments);\n const idx = this._senders.indexOf(sender);\n if (idx !== -1) {\n this._senders.splice(idx, 1);\n }\n };\n }\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._senders = this._senders || [];\n origAddStream.apply(this, [stream]);\n stream.getTracks().forEach(track => {\n this._senders.push(shimSenderWithDtmf(this, track));\n });\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._senders = this._senders || [];\n origRemoveStream.apply(this, [stream]);\n\n stream.getTracks().forEach(track => {\n const sender = this._senders.find(s => s.track === track);\n if (sender) { // remove sender\n this._senders.splice(this._senders.indexOf(sender), 1);\n }\n });\n };\n } else if (typeof window === 'object' && window.RTCPeerConnection &&\n 'getSenders' in window.RTCPeerConnection.prototype &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype &&\n window.RTCRtpSender &&\n !('dtmf' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = this._pc.createDTMFSender(this.track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n}\n\nexport function shimGetStats(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n\n // If selector is a function then we are in the old style stats so just\n // pass back the original getStats format to avoid breaking old users.\n if (arguments.length > 0 && typeof selector === 'function') {\n return origGetStats.apply(this, arguments);\n }\n\n // When spec-style getStats is supported, return those when called with\n // either no arguments or the selector argument is null.\n if (origGetStats.length === 0 && (arguments.length === 0 ||\n typeof selector !== 'function')) {\n return origGetStats.apply(this, []);\n }\n\n const fixChromeStats_ = function(response) {\n const standardReport = {};\n const reports = response.result();\n reports.forEach(report => {\n const standardStats = {\n id: report.id,\n timestamp: report.timestamp,\n type: {\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[report.type] || report.type\n };\n report.names().forEach(name => {\n standardStats[name] = report.stat(name);\n });\n standardReport[standardStats.id] = standardStats;\n });\n\n return standardReport;\n };\n\n // shim getStats with maplike support\n const makeMapStats = function(stats) {\n return new Map(Object.keys(stats).map(key => [key, stats[key]]));\n };\n\n if (arguments.length >= 2) {\n const successCallbackWrapper_ = function(response) {\n onSucc(makeMapStats(fixChromeStats_(response)));\n };\n\n return origGetStats.apply(this, [successCallbackWrapper_,\n selector]);\n }\n\n // promise-support\n return new Promise((resolve, reject) => {\n origGetStats.apply(this, [\n function(response) {\n resolve(makeMapStats(fixChromeStats_(response)));\n }, reject]);\n }).then(onSucc, onErr);\n };\n}\n\nexport function shimSenderReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender && window.RTCRtpReceiver)) {\n return;\n }\n\n // shim sender stats.\n if (!('getStats' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n const sender = this;\n return this._pc.getStats().then(result =>\n /* Note: this will include stats of all senders that\n * send a track with the same id as sender.track as\n * it is not possible to identify the RTCRtpSender.\n */\n utils.filterStats(result, sender.track, true));\n };\n }\n\n // shim receiver stats.\n if (!('getStats' in window.RTCRtpReceiver.prototype)) {\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers =\n function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n const receiver = this;\n return this._pc.getStats().then(result =>\n utils.filterStats(result, receiver.track, false));\n };\n }\n\n if (!('getStats' in window.RTCRtpSender.prototype &&\n 'getStats' in window.RTCRtpReceiver.prototype)) {\n return;\n }\n\n // shim RTCPeerConnection.getStats(track).\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n if (arguments.length > 0 &&\n arguments[0] instanceof window.MediaStreamTrack) {\n const track = arguments[0];\n let sender;\n let receiver;\n let err;\n this.getSenders().forEach(s => {\n if (s.track === track) {\n if (sender) {\n err = true;\n } else {\n sender = s;\n }\n }\n });\n this.getReceivers().forEach(r => {\n if (r.track === track) {\n if (receiver) {\n err = true;\n } else {\n receiver = r;\n }\n }\n return r.track === track;\n });\n if (err || (sender && receiver)) {\n return Promise.reject(new DOMException(\n 'There are more than one sender or receiver for the track.',\n 'InvalidAccessError'));\n } else if (sender) {\n return sender.getStats();\n } else if (receiver) {\n return receiver.getStats();\n }\n return Promise.reject(new DOMException(\n 'There is no sender or receiver for the track.',\n 'InvalidAccessError'));\n }\n return origGetStats.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrackWithNative(window) {\n // shim addTrack/removeTrack with native variants in order to make\n // the interactions with legacy getLocalStreams behave as in other browsers.\n // Keeps a mapping stream.id => [stream, rtpsenders...]\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n return Object.keys(this._shimmedLocalStreams)\n .map(streamId => this._shimmedLocalStreams[streamId][0]);\n };\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (!stream) {\n return origAddTrack.apply(this, arguments);\n }\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n const sender = origAddTrack.apply(this, arguments);\n if (!this._shimmedLocalStreams[stream.id]) {\n this._shimmedLocalStreams[stream.id] = [stream, sender];\n } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) {\n this._shimmedLocalStreams[stream.id].push(sender);\n }\n return sender;\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n const existingSenders = this.getSenders();\n origAddStream.apply(this, arguments);\n const newSenders = this.getSenders()\n .filter(newSender => existingSenders.indexOf(newSender) === -1);\n this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n delete this._shimmedLocalStreams[stream.id];\n return origRemoveStream.apply(this, arguments);\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n if (sender) {\n Object.keys(this._shimmedLocalStreams).forEach(streamId => {\n const idx = this._shimmedLocalStreams[streamId].indexOf(sender);\n if (idx !== -1) {\n this._shimmedLocalStreams[streamId].splice(idx, 1);\n }\n if (this._shimmedLocalStreams[streamId].length === 1) {\n delete this._shimmedLocalStreams[streamId];\n }\n });\n }\n return origRemoveTrack.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrack(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // shim addTrack and removeTrack.\n if (window.RTCPeerConnection.prototype.addTrack &&\n browserDetails.version >= 65) {\n return shimAddTrackRemoveTrackWithNative(window);\n }\n\n // also shim pc.getLocalStreams when addTrack is shimmed\n // to return the original streams.\n const origGetLocalStreams = window.RTCPeerConnection.prototype\n .getLocalStreams;\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n const nativeStreams = origGetLocalStreams.apply(this);\n this._reverseStreams = this._reverseStreams || {};\n return nativeStreams.map(stream => this._reverseStreams[stream.id]);\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n // Add identity mapping for consistency with addTrack.\n // Unless this is being used with a stream from addTrack.\n if (!this._reverseStreams[stream.id]) {\n const newStream = new window.MediaStream(stream.getTracks());\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n stream = newStream;\n }\n origAddStream.apply(this, [stream]);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]);\n delete this._reverseStreams[(this._streams[stream.id] ?\n this._streams[stream.id].id : stream.id)];\n delete this._streams[stream.id];\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n const streams = [].slice.call(arguments, 1);\n if (streams.length !== 1 ||\n !streams[0].getTracks().find(t => t === track)) {\n // this is not fully correct but all we can manage without\n // [[associated MediaStreams]] internal slot.\n throw new DOMException(\n 'The adapter.js addTrack polyfill only supports a single ' +\n ' stream which is associated with the specified track.',\n 'NotSupportedError');\n }\n\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n const oldStream = this._streams[stream.id];\n if (oldStream) {\n // this is using odd Chrome behaviour, use with caution:\n // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815\n // Note: we rely on the high-level addTrack/dtmf shim to\n // create the sender with a dtmf sender.\n oldStream.addTrack(track);\n\n // Trigger ONN async.\n Promise.resolve().then(() => {\n this.dispatchEvent(new Event('negotiationneeded'));\n });\n } else {\n const newStream = new window.MediaStream([track]);\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n this.addStream(newStream);\n }\n return this.getSenders().find(s => s.track === track);\n };\n\n // replace the internal stream id with the external one and\n // vice versa.\n function replaceInternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(internalStream.id, 'g'),\n externalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n function replaceExternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(externalStream.id, 'g'),\n internalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n ['createOffer', 'createAnswer'].forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n const args = arguments;\n const isLegacyCall = arguments.length &&\n typeof arguments[0] === 'function';\n if (isLegacyCall) {\n return nativeMethod.apply(this, [\n (description) => {\n const desc = replaceInternalStreamId(this, description);\n args[0].apply(null, [desc]);\n },\n (err) => {\n if (args[1]) {\n args[1].apply(null, err);\n }\n }, arguments[2]\n ]);\n }\n return nativeMethod.apply(this, arguments)\n .then(description => replaceInternalStreamId(this, description));\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n\n const origSetLocalDescription =\n window.RTCPeerConnection.prototype.setLocalDescription;\n window.RTCPeerConnection.prototype.setLocalDescription =\n function setLocalDescription() {\n if (!arguments.length || !arguments[0].type) {\n return origSetLocalDescription.apply(this, arguments);\n }\n arguments[0] = replaceExternalStreamId(this, arguments[0]);\n return origSetLocalDescription.apply(this, arguments);\n };\n\n // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier\n\n const origLocalDescription = Object.getOwnPropertyDescriptor(\n window.RTCPeerConnection.prototype, 'localDescription');\n Object.defineProperty(window.RTCPeerConnection.prototype,\n 'localDescription', {\n get() {\n const description = origLocalDescription.get.apply(this);\n if (description.type === '') {\n return description;\n }\n return replaceInternalStreamId(this, description);\n }\n });\n\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n // We can not yet check for sender instanceof RTCRtpSender\n // since we shim RTPSender. So we check if sender._pc is set.\n if (!sender._pc) {\n throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.', 'TypeError');\n }\n const isLocal = sender._pc === this;\n if (!isLocal) {\n throw new DOMException('Sender was not created by this connection.',\n 'InvalidAccessError');\n }\n\n // Search for the native stream the senders track belongs to.\n this._streams = this._streams || {};\n let stream;\n Object.keys(this._streams).forEach(streamid => {\n const hasTrack = this._streams[streamid].getTracks()\n .find(track => sender.track === track);\n if (hasTrack) {\n stream = this._streams[streamid];\n }\n });\n\n if (stream) {\n if (stream.getTracks().length === 1) {\n // if this is the last track of the stream, remove the stream. This\n // takes care of any shimmed _senders.\n this.removeStream(this._reverseStreams[stream.id]);\n } else {\n // relying on the same odd chrome behaviour as above.\n stream.removeTrack(sender.track);\n }\n this.dispatchEvent(new Event('negotiationneeded'));\n }\n };\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.webkitRTCPeerConnection;\n }\n if (!window.RTCPeerConnection) {\n return;\n }\n\n // shim implicit creation of RTCSessionDescription/RTCIceCandidate\n if (browserDetails.version < 53) {\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n}\n\n// Attempt to fix ONN in plan-b mode.\nexport function fixNegotiationNeeded(window, browserDetails) {\n utils.wrapPeerConnectionEvent(window, 'negotiationneeded', e => {\n const pc = e.target;\n if (browserDetails.version < 72 || (pc.getConfiguration &&\n pc.getConfiguration().sdpSemantics === 'plan-b')) {\n if (pc.signalingState !== 'stable') {\n return;\n }\n }\n return e;\n });\n}\n","/*\n * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nexport function filterIceServers(iceServers, edgeVersion) {\n let hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(server => {\n if (server && (server.urls || server.url)) {\n let urls = server.urls || server.url;\n if (server.url && !server.urls) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n }\n const isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(url => {\n // filter STUN unconditionally.\n if (url.indexOf('stun:') === 0) {\n return false;\n }\n\n const validTurn = url.startsWith('turn') &&\n !url.startsWith('turn:[') &&\n url.includes('transport=udp');\n if (validTurn && !hasTurn) {\n hasTurn = true;\n return true;\n }\n return validTurn && !hasTurn;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n","/* eslint-env node */\n'use strict';\n\n// SDP helpers.\nvar SDPUtils = {};\n\n// Generate an alphanumeric identifier for cname or mids.\n// TODO: use UUIDs instead? https://gist.github.com/jed/982883\nSDPUtils.generateIdentifier = function() {\n return Math.random().toString(36).substr(2, 10);\n};\n\n// The RTCP CNAME used by all peerconnections from the same JS.\nSDPUtils.localCName = SDPUtils.generateIdentifier();\n\n// Splits SDP into lines, dealing with both CRLF and LF.\nSDPUtils.splitLines = function(blob) {\n return blob.trim().split('\\n').map(function(line) {\n return line.trim();\n });\n};\n// Splits SDP into sessionpart and mediasections. Ensures CRLF.\nSDPUtils.splitSections = function(blob) {\n var parts = blob.split('\\nm=');\n return parts.map(function(part, index) {\n return (index > 0 ? 'm=' + part : part).trim() + '\\r\\n';\n });\n};\n\n// returns the session description.\nSDPUtils.getDescription = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n return sections && sections[0];\n};\n\n// returns the individual media sections.\nSDPUtils.getMediaSections = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n sections.shift();\n return sections;\n};\n\n// Returns lines that start with a certain prefix.\nSDPUtils.matchPrefix = function(blob, prefix) {\n return SDPUtils.splitLines(blob).filter(function(line) {\n return line.indexOf(prefix) === 0;\n });\n};\n\n// Parses an ICE candidate line. Sample input:\n// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8\n// rport 55996\"\nSDPUtils.parseCandidate = function(line) {\n var parts;\n // Parse both variants.\n if (line.indexOf('a=candidate:') === 0) {\n parts = line.substring(12).split(' ');\n } else {\n parts = line.substring(10).split(' ');\n }\n\n var candidate = {\n foundation: parts[0],\n component: parseInt(parts[1], 10),\n protocol: parts[2].toLowerCase(),\n priority: parseInt(parts[3], 10),\n ip: parts[4],\n address: parts[4], // address is an alias for ip.\n port: parseInt(parts[5], 10),\n // skip parts[6] == 'typ'\n type: parts[7]\n };\n\n for (var i = 8; i < parts.length; i += 2) {\n switch (parts[i]) {\n case 'raddr':\n candidate.relatedAddress = parts[i + 1];\n break;\n case 'rport':\n candidate.relatedPort = parseInt(parts[i + 1], 10);\n break;\n case 'tcptype':\n candidate.tcpType = parts[i + 1];\n break;\n case 'ufrag':\n candidate.ufrag = parts[i + 1]; // for backward compability.\n candidate.usernameFragment = parts[i + 1];\n break;\n default: // extension handling, in particular ufrag\n candidate[parts[i]] = parts[i + 1];\n break;\n }\n }\n return candidate;\n};\n\n// Translates a candidate object into SDP candidate attribute.\nSDPUtils.writeCandidate = function(candidate) {\n var sdp = [];\n sdp.push(candidate.foundation);\n sdp.push(candidate.component);\n sdp.push(candidate.protocol.toUpperCase());\n sdp.push(candidate.priority);\n sdp.push(candidate.address || candidate.ip);\n sdp.push(candidate.port);\n\n var type = candidate.type;\n sdp.push('typ');\n sdp.push(type);\n if (type !== 'host' && candidate.relatedAddress &&\n candidate.relatedPort) {\n sdp.push('raddr');\n sdp.push(candidate.relatedAddress);\n sdp.push('rport');\n sdp.push(candidate.relatedPort);\n }\n if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') {\n sdp.push('tcptype');\n sdp.push(candidate.tcpType);\n }\n if (candidate.usernameFragment || candidate.ufrag) {\n sdp.push('ufrag');\n sdp.push(candidate.usernameFragment || candidate.ufrag);\n }\n return 'candidate:' + sdp.join(' ');\n};\n\n// Parses an ice-options line, returns an array of option tags.\n// a=ice-options:foo bar\nSDPUtils.parseIceOptions = function(line) {\n return line.substr(14).split(' ');\n};\n\n// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:\n// a=rtpmap:111 opus/48000/2\nSDPUtils.parseRtpMap = function(line) {\n var parts = line.substr(9).split(' ');\n var parsed = {\n payloadType: parseInt(parts.shift(), 10) // was: id\n };\n\n parts = parts[0].split('/');\n\n parsed.name = parts[0];\n parsed.clockRate = parseInt(parts[1], 10); // was: clockrate\n parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;\n // legacy alias, got renamed back to channels in ORTC.\n parsed.numChannels = parsed.channels;\n return parsed;\n};\n\n// Generate an a=rtpmap line from RTCRtpCodecCapability or\n// RTCRtpCodecParameters.\nSDPUtils.writeRtpMap = function(codec) {\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n var channels = codec.channels || codec.numChannels || 1;\n return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate +\n (channels !== 1 ? '/' + channels : '') + '\\r\\n';\n};\n\n// Parses an a=extmap line (headerextension from RFC 5285). Sample input:\n// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset\nSDPUtils.parseExtmap = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n id: parseInt(parts[0], 10),\n direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv',\n uri: parts[1]\n };\n};\n\n// Generates a=extmap line from RTCRtpHeaderExtensionParameters or\n// RTCRtpHeaderExtension.\nSDPUtils.writeExtmap = function(headerExtension) {\n return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) +\n (headerExtension.direction && headerExtension.direction !== 'sendrecv'\n ? '/' + headerExtension.direction\n : '') +\n ' ' + headerExtension.uri + '\\r\\n';\n};\n\n// Parses an ftmp line, returns dictionary. Sample input:\n// a=fmtp:96 vbr=on;cng=on\n// Also deals with vbr=on; cng=on\nSDPUtils.parseFmtp = function(line) {\n var parsed = {};\n var kv;\n var parts = line.substr(line.indexOf(' ') + 1).split(';');\n for (var j = 0; j < parts.length; j++) {\n kv = parts[j].trim().split('=');\n parsed[kv[0].trim()] = kv[1];\n }\n return parsed;\n};\n\n// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeFmtp = function(codec) {\n var line = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.parameters && Object.keys(codec.parameters).length) {\n var params = [];\n Object.keys(codec.parameters).forEach(function(param) {\n if (codec.parameters[param]) {\n params.push(param + '=' + codec.parameters[param]);\n } else {\n params.push(param);\n }\n });\n line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\\r\\n';\n }\n return line;\n};\n\n// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:\n// a=rtcp-fb:98 nack rpsi\nSDPUtils.parseRtcpFb = function(line) {\n var parts = line.substr(line.indexOf(' ') + 1).split(' ');\n return {\n type: parts.shift(),\n parameter: parts.join(' ')\n };\n};\n// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeRtcpFb = function(codec) {\n var lines = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.rtcpFeedback && codec.rtcpFeedback.length) {\n // FIXME: special handling for trr-int?\n codec.rtcpFeedback.forEach(function(fb) {\n lines += 'a=rtcp-fb:' + pt + ' ' + fb.type +\n (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') +\n '\\r\\n';\n });\n }\n return lines;\n};\n\n// Parses an RFC 5576 ssrc media attribute. Sample input:\n// a=ssrc:3735928559 cname:something\nSDPUtils.parseSsrcMedia = function(line) {\n var sp = line.indexOf(' ');\n var parts = {\n ssrc: parseInt(line.substr(7, sp - 7), 10)\n };\n var colon = line.indexOf(':', sp);\n if (colon > -1) {\n parts.attribute = line.substr(sp + 1, colon - sp - 1);\n parts.value = line.substr(colon + 1);\n } else {\n parts.attribute = line.substr(sp + 1);\n }\n return parts;\n};\n\nSDPUtils.parseSsrcGroup = function(line) {\n var parts = line.substr(13).split(' ');\n return {\n semantics: parts.shift(),\n ssrcs: parts.map(function(ssrc) {\n return parseInt(ssrc, 10);\n })\n };\n};\n\n// Extracts the MID (RFC 5888) from a media section.\n// returns the MID or undefined if no mid line was found.\nSDPUtils.getMid = function(mediaSection) {\n var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0];\n if (mid) {\n return mid.substr(6);\n }\n};\n\nSDPUtils.parseFingerprint = function(line) {\n var parts = line.substr(14).split(' ');\n return {\n algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.\n value: parts[1]\n };\n};\n\n// Extracts DTLS parameters from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the fingerprint line as input. See also getIceParameters.\nSDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=fingerprint:');\n // Note: a=setup line is ignored since we use the 'auto' role.\n // Note2: 'algorithm' is not case sensitive except in Edge.\n return {\n role: 'auto',\n fingerprints: lines.map(SDPUtils.parseFingerprint)\n };\n};\n\n// Serializes DTLS parameters to SDP.\nSDPUtils.writeDtlsParameters = function(params, setupType) {\n var sdp = 'a=setup:' + setupType + '\\r\\n';\n params.fingerprints.forEach(function(fp) {\n sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\\r\\n';\n });\n return sdp;\n};\n\n// Parses a=crypto lines into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members\nSDPUtils.parseCryptoLine = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n tag: parseInt(parts[0], 10),\n cryptoSuite: parts[1],\n keyParams: parts[2],\n sessionParams: parts.slice(3),\n };\n};\n\nSDPUtils.writeCryptoLine = function(parameters) {\n return 'a=crypto:' + parameters.tag + ' ' +\n parameters.cryptoSuite + ' ' +\n (typeof parameters.keyParams === 'object'\n ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)\n : parameters.keyParams) +\n (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') +\n '\\r\\n';\n};\n\n// Parses the crypto key parameters into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*\nSDPUtils.parseCryptoKeyParams = function(keyParams) {\n if (keyParams.indexOf('inline:') !== 0) {\n return null;\n }\n var parts = keyParams.substr(7).split('|');\n return {\n keyMethod: 'inline',\n keySalt: parts[0],\n lifeTime: parts[1],\n mkiValue: parts[2] ? parts[2].split(':')[0] : undefined,\n mkiLength: parts[2] ? parts[2].split(':')[1] : undefined,\n };\n};\n\nSDPUtils.writeCryptoKeyParams = function(keyParams) {\n return keyParams.keyMethod + ':'\n + keyParams.keySalt +\n (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') +\n (keyParams.mkiValue && keyParams.mkiLength\n ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength\n : '');\n};\n\n// Extracts all SDES paramters.\nSDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=crypto:');\n return lines.map(SDPUtils.parseCryptoLine);\n};\n\n// Parses ICE information from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the ice-ufrag and ice-pwd lines as input.\nSDPUtils.getIceParameters = function(mediaSection, sessionpart) {\n var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-ufrag:')[0];\n var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-pwd:')[0];\n if (!(ufrag && pwd)) {\n return null;\n }\n return {\n usernameFragment: ufrag.substr(12),\n password: pwd.substr(10),\n };\n};\n\n// Serializes ICE parameters to SDP.\nSDPUtils.writeIceParameters = function(params) {\n return 'a=ice-ufrag:' + params.usernameFragment + '\\r\\n' +\n 'a=ice-pwd:' + params.password + '\\r\\n';\n};\n\n// Parses the SDP media section and returns RTCRtpParameters.\nSDPUtils.parseRtpParameters = function(mediaSection) {\n var description = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: [],\n rtcp: []\n };\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..]\n var pt = mline[i];\n var rtpmapline = SDPUtils.matchPrefix(\n mediaSection, 'a=rtpmap:' + pt + ' ')[0];\n if (rtpmapline) {\n var codec = SDPUtils.parseRtpMap(rtpmapline);\n var fmtps = SDPUtils.matchPrefix(\n mediaSection, 'a=fmtp:' + pt + ' ');\n // Only the first a=fmtp: is considered.\n codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};\n codec.rtcpFeedback = SDPUtils.matchPrefix(\n mediaSection, 'a=rtcp-fb:' + pt + ' ')\n .map(SDPUtils.parseRtcpFb);\n description.codecs.push(codec);\n // parse FEC mechanisms from rtpmap lines.\n switch (codec.name.toUpperCase()) {\n case 'RED':\n case 'ULPFEC':\n description.fecMechanisms.push(codec.name.toUpperCase());\n break;\n default: // only RED and ULPFEC are recognized as FEC mechanisms.\n break;\n }\n }\n }\n SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) {\n description.headerExtensions.push(SDPUtils.parseExtmap(line));\n });\n // FIXME: parse rtcp.\n return description;\n};\n\n// Generates parts of the SDP media section describing the capabilities /\n// parameters.\nSDPUtils.writeRtpDescription = function(kind, caps) {\n var sdp = '';\n\n // Build the mline.\n sdp += 'm=' + kind + ' ';\n sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs.\n sdp += ' UDP/TLS/RTP/SAVPF ';\n sdp += caps.codecs.map(function(codec) {\n if (codec.preferredPayloadType !== undefined) {\n return codec.preferredPayloadType;\n }\n return codec.payloadType;\n }).join(' ') + '\\r\\n';\n\n sdp += 'c=IN IP4 0.0.0.0\\r\\n';\n sdp += 'a=rtcp:9 IN IP4 0.0.0.0\\r\\n';\n\n // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.\n caps.codecs.forEach(function(codec) {\n sdp += SDPUtils.writeRtpMap(codec);\n sdp += SDPUtils.writeFmtp(codec);\n sdp += SDPUtils.writeRtcpFb(codec);\n });\n var maxptime = 0;\n caps.codecs.forEach(function(codec) {\n if (codec.maxptime > maxptime) {\n maxptime = codec.maxptime;\n }\n });\n if (maxptime > 0) {\n sdp += 'a=maxptime:' + maxptime + '\\r\\n';\n }\n sdp += 'a=rtcp-mux\\r\\n';\n\n if (caps.headerExtensions) {\n caps.headerExtensions.forEach(function(extension) {\n sdp += SDPUtils.writeExtmap(extension);\n });\n }\n // FIXME: write fecMechanisms.\n return sdp;\n};\n\n// Parses the SDP media section and returns an array of\n// RTCRtpEncodingParameters.\nSDPUtils.parseRtpEncodingParameters = function(mediaSection) {\n var encodingParameters = [];\n var description = SDPUtils.parseRtpParameters(mediaSection);\n var hasRed = description.fecMechanisms.indexOf('RED') !== -1;\n var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1;\n\n // filter a=ssrc:... cname:, ignore PlanB-msid\n var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(parts) {\n return parts.attribute === 'cname';\n });\n var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;\n var secondarySsrc;\n\n var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID')\n .map(function(line) {\n var parts = line.substr(17).split(' ');\n return parts.map(function(part) {\n return parseInt(part, 10);\n });\n });\n if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {\n secondarySsrc = flows[0][1];\n }\n\n description.codecs.forEach(function(codec) {\n if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) {\n var encParam = {\n ssrc: primarySsrc,\n codecPayloadType: parseInt(codec.parameters.apt, 10)\n };\n if (primarySsrc && secondarySsrc) {\n encParam.rtx = {ssrc: secondarySsrc};\n }\n encodingParameters.push(encParam);\n if (hasRed) {\n encParam = JSON.parse(JSON.stringify(encParam));\n encParam.fec = {\n ssrc: primarySsrc,\n mechanism: hasUlpfec ? 'red+ulpfec' : 'red'\n };\n encodingParameters.push(encParam);\n }\n }\n });\n if (encodingParameters.length === 0 && primarySsrc) {\n encodingParameters.push({\n ssrc: primarySsrc\n });\n }\n\n // we support both b=AS and b=TIAS but interpret AS as TIAS.\n var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b=');\n if (bandwidth.length) {\n if (bandwidth[0].indexOf('b=TIAS:') === 0) {\n bandwidth = parseInt(bandwidth[0].substr(7), 10);\n } else if (bandwidth[0].indexOf('b=AS:') === 0) {\n // use formula from JSEP to convert b=AS to TIAS value.\n bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95\n - (50 * 40 * 8);\n } else {\n bandwidth = undefined;\n }\n encodingParameters.forEach(function(params) {\n params.maxBitrate = bandwidth;\n });\n }\n return encodingParameters;\n};\n\n// parses http://draft.ortc.org/#rtcrtcpparameters*\nSDPUtils.parseRtcpParameters = function(mediaSection) {\n var rtcpParameters = {};\n\n // Gets the first SSRC. Note tha with RTX there might be multiple\n // SSRCs.\n var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(obj) {\n return obj.attribute === 'cname';\n })[0];\n if (remoteSsrc) {\n rtcpParameters.cname = remoteSsrc.value;\n rtcpParameters.ssrc = remoteSsrc.ssrc;\n }\n\n // Edge uses the compound attribute instead of reducedSize\n // compound is !reducedSize\n var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize');\n rtcpParameters.reducedSize = rsize.length > 0;\n rtcpParameters.compound = rsize.length === 0;\n\n // parses the rtcp-mux attrіbute.\n // Note that Edge does not support unmuxed RTCP.\n var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux');\n rtcpParameters.mux = mux.length > 0;\n\n return rtcpParameters;\n};\n\n// parses either a=msid: or a=ssrc:... msid lines and returns\n// the id of the MediaStream and MediaStreamTrack.\nSDPUtils.parseMsid = function(mediaSection) {\n var parts;\n var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:');\n if (spec.length === 1) {\n parts = spec[0].substr(7).split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(msidParts) {\n return msidParts.attribute === 'msid';\n });\n if (planB.length > 0) {\n parts = planB[0].value.split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n};\n\n// SCTP\n// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back\n// to draft-ietf-mmusic-sctp-sdp-05\nSDPUtils.parseSctpDescription = function(mediaSection) {\n var mline = SDPUtils.parseMLine(mediaSection);\n var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:');\n var maxMessageSize;\n if (maxSizeLine.length > 0) {\n maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);\n }\n if (isNaN(maxMessageSize)) {\n maxMessageSize = 65536;\n }\n var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:');\n if (sctpPort.length > 0) {\n return {\n port: parseInt(sctpPort[0].substr(12), 10),\n protocol: mline.fmt,\n maxMessageSize: maxMessageSize\n };\n }\n var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:');\n if (sctpMapLines.length > 0) {\n var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0]\n .substr(10)\n .split(' ');\n return {\n port: parseInt(parts[0], 10),\n protocol: parts[1],\n maxMessageSize: maxMessageSize\n };\n }\n};\n\n// SCTP\n// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers\n// support by now receiving in this format, unless we originally parsed\n// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line\n// protocol of DTLS/SCTP -- without UDP/ or TCP/)\nSDPUtils.writeSctpDescription = function(media, sctp) {\n var output = [];\n if (media.protocol !== 'DTLS/SCTP') {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctp-port:' + sctp.port + '\\r\\n'\n ];\n } else {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\\r\\n'\n ];\n }\n if (sctp.maxMessageSize !== undefined) {\n output.push('a=max-message-size:' + sctp.maxMessageSize + '\\r\\n');\n }\n return output.join('');\n};\n\n// Generate a session ID for SDP.\n// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1\n// recommends using a cryptographically random +ve 64-bit value\n// but right now this should be acceptable and within the right range\nSDPUtils.generateSessionId = function() {\n return Math.random().toString().substr(2, 21);\n};\n\n// Write boilder plate for start of SDP\n// sessId argument is optional - if not supplied it will\n// be generated randomly\n// sessVersion is optional and defaults to 2\n// sessUser is optional and defaults to 'thisisadapterortc'\nSDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {\n var sessionId;\n var version = sessVer !== undefined ? sessVer : 2;\n if (sessId) {\n sessionId = sessId;\n } else {\n sessionId = SDPUtils.generateSessionId();\n }\n var user = sessUser || 'thisisadapterortc';\n // FIXME: sess-id should be an NTP timestamp.\n return 'v=0\\r\\n' +\n 'o=' + user + ' ' + sessionId + ' ' + version +\n ' IN IP4 127.0.0.1\\r\\n' +\n 's=-\\r\\n' +\n 't=0 0\\r\\n';\n};\n\nSDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.direction) {\n sdp += 'a=' + transceiver.direction + '\\r\\n';\n } else if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n // spec.\n var msid = 'msid:' + stream.id + ' ' +\n transceiver.rtpSender.track.id + '\\r\\n';\n sdp += 'a=' + msid;\n\n // for Chrome.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n};\n\n// Gets the direction from the mediaSection or the sessionpart.\nSDPUtils.getDirection = function(mediaSection, sessionpart) {\n // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.\n var lines = SDPUtils.splitLines(mediaSection);\n for (var i = 0; i < lines.length; i++) {\n switch (lines[i]) {\n case 'a=sendrecv':\n case 'a=sendonly':\n case 'a=recvonly':\n case 'a=inactive':\n return lines[i].substr(2);\n default:\n // FIXME: What should happen here?\n }\n }\n if (sessionpart) {\n return SDPUtils.getDirection(sessionpart);\n }\n return 'sendrecv';\n};\n\nSDPUtils.getKind = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n return mline[0].substr(2);\n};\n\nSDPUtils.isRejected = function(mediaSection) {\n return mediaSection.split(' ', 2)[1] === '0';\n};\n\nSDPUtils.parseMLine = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var parts = lines[0].substr(2).split(' ');\n return {\n kind: parts[0],\n port: parseInt(parts[1], 10),\n protocol: parts[2],\n fmt: parts.slice(3).join(' ')\n };\n};\n\nSDPUtils.parseOLine = function(mediaSection) {\n var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0];\n var parts = line.substr(2).split(' ');\n return {\n username: parts[0],\n sessionId: parts[1],\n sessionVersion: parseInt(parts[2], 10),\n netType: parts[3],\n addressType: parts[4],\n address: parts[5]\n };\n};\n\n// a very naive interpretation of a valid SDP.\nSDPUtils.isValidSDP = function(blob) {\n if (typeof blob !== 'string' || blob.length === 0) {\n return false;\n }\n var lines = SDPUtils.splitLines(blob);\n for (var i = 0; i < lines.length; i++) {\n if (lines[i].length < 2 || lines[i].charAt(1) !== '=') {\n return false;\n }\n // TODO: check the modifier a bit more.\n }\n return true;\n};\n\n// Expose public methods.\nif (typeof module === 'object') {\n module.exports = SDPUtils;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nvar SDPUtils = require('sdp');\n\nfunction fixStatsType(stat) {\n return {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[stat.type] || stat.type;\n}\n\nfunction writeMediaSection(transceiver, caps, type, stream, dtlsRole) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : dtlsRole || 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n var trackId = transceiver.rtpSender._initialTrackId ||\n transceiver.rtpSender.track.id;\n transceiver.rtpSender._initialTrackId = trackId;\n // spec.\n var msid = 'msid:' + (stream ? stream.id : '-') + ' ' +\n trackId + '\\r\\n';\n sdp += 'a=' + msid;\n // for Chrome. Legacy should no longer be required.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n\n // RTX\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n}\n\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nfunction filterIceServers(iceServers, edgeVersion) {\n var hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(function(server) {\n if (server && (server.urls || server.url)) {\n var urls = server.urls || server.url;\n if (server.url && !server.urls) {\n console.warn('RTCIceServer.url is deprecated! Use urls instead.');\n }\n var isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(function(url) {\n var validTurn = url.indexOf('turn:') === 0 &&\n url.indexOf('transport=udp') !== -1 &&\n url.indexOf('turn:[') === -1 &&\n !hasTurn;\n\n if (validTurn) {\n hasTurn = true;\n return true;\n }\n return url.indexOf('stun:') === 0 && edgeVersion >= 14393 &&\n url.indexOf('?transport=udp') === -1;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n\n// Determines the intersection of local and remote capabilities.\nfunction getCommonCapabilities(localCapabilities, remoteCapabilities) {\n var commonCapabilities = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: []\n };\n\n var findCodecByPayloadType = function(pt, codecs) {\n pt = parseInt(pt, 10);\n for (var i = 0; i < codecs.length; i++) {\n if (codecs[i].payloadType === pt ||\n codecs[i].preferredPayloadType === pt) {\n return codecs[i];\n }\n }\n };\n\n var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) {\n var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs);\n var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs);\n return lCodec && rCodec &&\n lCodec.name.toLowerCase() === rCodec.name.toLowerCase();\n };\n\n localCapabilities.codecs.forEach(function(lCodec) {\n for (var i = 0; i < remoteCapabilities.codecs.length; i++) {\n var rCodec = remoteCapabilities.codecs[i];\n if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() &&\n lCodec.clockRate === rCodec.clockRate) {\n if (lCodec.name.toLowerCase() === 'rtx' &&\n lCodec.parameters && rCodec.parameters.apt) {\n // for RTX we need to find the local rtx that has a apt\n // which points to the same local codec as the remote one.\n if (!rtxCapabilityMatches(lCodec, rCodec,\n localCapabilities.codecs, remoteCapabilities.codecs)) {\n continue;\n }\n }\n rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy\n // number of channels is the highest common number of channels\n rCodec.numChannels = Math.min(lCodec.numChannels,\n rCodec.numChannels);\n // push rCodec so we reply with offerer payload type\n commonCapabilities.codecs.push(rCodec);\n\n // determine common feedback mechanisms\n rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) {\n for (var j = 0; j < lCodec.rtcpFeedback.length; j++) {\n if (lCodec.rtcpFeedback[j].type === fb.type &&\n lCodec.rtcpFeedback[j].parameter === fb.parameter) {\n return true;\n }\n }\n return false;\n });\n // FIXME: also need to determine .parameters\n // see https://github.com/openpeer/ortc/issues/569\n break;\n }\n }\n });\n\n localCapabilities.headerExtensions.forEach(function(lHeaderExtension) {\n for (var i = 0; i < remoteCapabilities.headerExtensions.length;\n i++) {\n var rHeaderExtension = remoteCapabilities.headerExtensions[i];\n if (lHeaderExtension.uri === rHeaderExtension.uri) {\n commonCapabilities.headerExtensions.push(rHeaderExtension);\n break;\n }\n }\n });\n\n // FIXME: fecMechanisms\n return commonCapabilities;\n}\n\n// is action=setLocalDescription with type allowed in signalingState\nfunction isActionAllowedInSignalingState(action, type, signalingState) {\n return {\n offer: {\n setLocalDescription: ['stable', 'have-local-offer'],\n setRemoteDescription: ['stable', 'have-remote-offer']\n },\n answer: {\n setLocalDescription: ['have-remote-offer', 'have-local-pranswer'],\n setRemoteDescription: ['have-local-offer', 'have-remote-pranswer']\n }\n }[type][action].indexOf(signalingState) !== -1;\n}\n\nfunction maybeAddCandidate(iceTransport, candidate) {\n // Edge's internal representation adds some fields therefore\n // not all fieldѕ are taken into account.\n var alreadyAdded = iceTransport.getRemoteCandidates()\n .find(function(remoteCandidate) {\n return candidate.foundation === remoteCandidate.foundation &&\n candidate.ip === remoteCandidate.ip &&\n candidate.port === remoteCandidate.port &&\n candidate.priority === remoteCandidate.priority &&\n candidate.protocol === remoteCandidate.protocol &&\n candidate.type === remoteCandidate.type;\n });\n if (!alreadyAdded) {\n iceTransport.addRemoteCandidate(candidate);\n }\n return !alreadyAdded;\n}\n\n\nfunction makeError(name, description) {\n var e = new Error(description);\n e.name = name;\n // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names\n e.code = {\n NotSupportedError: 9,\n InvalidStateError: 11,\n InvalidAccessError: 15,\n TypeError: undefined,\n OperationError: undefined\n }[name];\n return e;\n}\n\nmodule.exports = function(window, edgeVersion) {\n // https://w3c.github.io/mediacapture-main/#mediastream\n // Helper function to add the track to the stream and\n // dispatch the event ourselves.\n function addTrackToStreamAndFireEvent(track, stream) {\n stream.addTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack',\n {track: track}));\n }\n\n function removeTrackFromStreamAndFireEvent(track, stream) {\n stream.removeTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack',\n {track: track}));\n }\n\n function fireAddTrack(pc, track, receiver, streams) {\n var trackEvent = new Event('track');\n trackEvent.track = track;\n trackEvent.receiver = receiver;\n trackEvent.transceiver = {receiver: receiver};\n trackEvent.streams = streams;\n window.setTimeout(function() {\n pc._dispatchEvent('track', trackEvent);\n });\n }\n\n var RTCPeerConnection = function(config) {\n var pc = this;\n\n var _eventTarget = document.createDocumentFragment();\n ['addEventListener', 'removeEventListener', 'dispatchEvent']\n .forEach(function(method) {\n pc[method] = _eventTarget[method].bind(_eventTarget);\n });\n\n this.canTrickleIceCandidates = null;\n\n this.needNegotiation = false;\n\n this.localStreams = [];\n this.remoteStreams = [];\n\n this._localDescription = null;\n this._remoteDescription = null;\n\n this.signalingState = 'stable';\n this.iceConnectionState = 'new';\n this.connectionState = 'new';\n this.iceGatheringState = 'new';\n\n config = JSON.parse(JSON.stringify(config || {}));\n\n this.usingBundle = config.bundlePolicy === 'max-bundle';\n if (config.rtcpMuxPolicy === 'negotiate') {\n throw(makeError('NotSupportedError',\n 'rtcpMuxPolicy \\'negotiate\\' is not supported'));\n } else if (!config.rtcpMuxPolicy) {\n config.rtcpMuxPolicy = 'require';\n }\n\n switch (config.iceTransportPolicy) {\n case 'all':\n case 'relay':\n break;\n default:\n config.iceTransportPolicy = 'all';\n break;\n }\n\n switch (config.bundlePolicy) {\n case 'balanced':\n case 'max-compat':\n case 'max-bundle':\n break;\n default:\n config.bundlePolicy = 'balanced';\n break;\n }\n\n config.iceServers = filterIceServers(config.iceServers || [], edgeVersion);\n\n this._iceGatherers = [];\n if (config.iceCandidatePoolSize) {\n for (var i = config.iceCandidatePoolSize; i > 0; i--) {\n this._iceGatherers.push(new window.RTCIceGatherer({\n iceServers: config.iceServers,\n gatherPolicy: config.iceTransportPolicy\n }));\n }\n } else {\n config.iceCandidatePoolSize = 0;\n }\n\n this._config = config;\n\n // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ...\n // everything that is needed to describe a SDP m-line.\n this.transceivers = [];\n\n this._sdpSessionId = SDPUtils.generateSessionId();\n this._sdpSessionVersion = 0;\n\n this._dtlsRole = undefined; // role for a=setup to use in answers.\n\n this._isClosed = false;\n };\n\n Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', {\n configurable: true,\n get: function() {\n return this._localDescription;\n }\n });\n Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', {\n configurable: true,\n get: function() {\n return this._remoteDescription;\n }\n });\n\n // set up event handlers on prototype\n RTCPeerConnection.prototype.onicecandidate = null;\n RTCPeerConnection.prototype.onaddstream = null;\n RTCPeerConnection.prototype.ontrack = null;\n RTCPeerConnection.prototype.onremovestream = null;\n RTCPeerConnection.prototype.onsignalingstatechange = null;\n RTCPeerConnection.prototype.oniceconnectionstatechange = null;\n RTCPeerConnection.prototype.onconnectionstatechange = null;\n RTCPeerConnection.prototype.onicegatheringstatechange = null;\n RTCPeerConnection.prototype.onnegotiationneeded = null;\n RTCPeerConnection.prototype.ondatachannel = null;\n\n RTCPeerConnection.prototype._dispatchEvent = function(name, event) {\n if (this._isClosed) {\n return;\n }\n this.dispatchEvent(event);\n if (typeof this['on' + name] === 'function') {\n this['on' + name](event);\n }\n };\n\n RTCPeerConnection.prototype._emitGatheringStateChange = function() {\n var event = new Event('icegatheringstatechange');\n this._dispatchEvent('icegatheringstatechange', event);\n };\n\n RTCPeerConnection.prototype.getConfiguration = function() {\n return this._config;\n };\n\n RTCPeerConnection.prototype.getLocalStreams = function() {\n return this.localStreams;\n };\n\n RTCPeerConnection.prototype.getRemoteStreams = function() {\n return this.remoteStreams;\n };\n\n // internal helper to create a transceiver object.\n // (which is not yet the same as the WebRTC 1.0 transceiver)\n RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) {\n var hasBundleTransport = this.transceivers.length > 0;\n var transceiver = {\n track: null,\n iceGatherer: null,\n iceTransport: null,\n dtlsTransport: null,\n localCapabilities: null,\n remoteCapabilities: null,\n rtpSender: null,\n rtpReceiver: null,\n kind: kind,\n mid: null,\n sendEncodingParameters: null,\n recvEncodingParameters: null,\n stream: null,\n associatedRemoteMediaStreams: [],\n wantReceive: true\n };\n if (this.usingBundle && hasBundleTransport) {\n transceiver.iceTransport = this.transceivers[0].iceTransport;\n transceiver.dtlsTransport = this.transceivers[0].dtlsTransport;\n } else {\n var transports = this._createIceAndDtlsTransports();\n transceiver.iceTransport = transports.iceTransport;\n transceiver.dtlsTransport = transports.dtlsTransport;\n }\n if (!doNotAdd) {\n this.transceivers.push(transceiver);\n }\n return transceiver;\n };\n\n RTCPeerConnection.prototype.addTrack = function(track, stream) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call addTrack on a closed peerconnection.');\n }\n\n var alreadyExists = this.transceivers.find(function(s) {\n return s.track === track;\n });\n\n if (alreadyExists) {\n throw makeError('InvalidAccessError', 'Track already exists.');\n }\n\n var transceiver;\n for (var i = 0; i < this.transceivers.length; i++) {\n if (!this.transceivers[i].track &&\n this.transceivers[i].kind === track.kind) {\n transceiver = this.transceivers[i];\n }\n }\n if (!transceiver) {\n transceiver = this._createTransceiver(track.kind);\n }\n\n this._maybeFireNegotiationNeeded();\n\n if (this.localStreams.indexOf(stream) === -1) {\n this.localStreams.push(stream);\n }\n\n transceiver.track = track;\n transceiver.stream = stream;\n transceiver.rtpSender = new window.RTCRtpSender(track,\n transceiver.dtlsTransport);\n return transceiver.rtpSender;\n };\n\n RTCPeerConnection.prototype.addStream = function(stream) {\n var pc = this;\n if (edgeVersion >= 15025) {\n stream.getTracks().forEach(function(track) {\n pc.addTrack(track, stream);\n });\n } else {\n // Clone is necessary for local demos mostly, attaching directly\n // to two different senders does not work (build 10547).\n // Fixed in 15025 (or earlier)\n var clonedStream = stream.clone();\n stream.getTracks().forEach(function(track, idx) {\n var clonedTrack = clonedStream.getTracks()[idx];\n track.addEventListener('enabled', function(event) {\n clonedTrack.enabled = event.enabled;\n });\n });\n clonedStream.getTracks().forEach(function(track) {\n pc.addTrack(track, clonedStream);\n });\n }\n };\n\n RTCPeerConnection.prototype.removeTrack = function(sender) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call removeTrack on a closed peerconnection.');\n }\n\n if (!(sender instanceof window.RTCRtpSender)) {\n throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.');\n }\n\n var transceiver = this.transceivers.find(function(t) {\n return t.rtpSender === sender;\n });\n\n if (!transceiver) {\n throw makeError('InvalidAccessError',\n 'Sender was not created by this connection.');\n }\n var stream = transceiver.stream;\n\n transceiver.rtpSender.stop();\n transceiver.rtpSender = null;\n transceiver.track = null;\n transceiver.stream = null;\n\n // remove the stream from the set of local streams\n var localStreams = this.transceivers.map(function(t) {\n return t.stream;\n });\n if (localStreams.indexOf(stream) === -1 &&\n this.localStreams.indexOf(stream) > -1) {\n this.localStreams.splice(this.localStreams.indexOf(stream), 1);\n }\n\n this._maybeFireNegotiationNeeded();\n };\n\n RTCPeerConnection.prototype.removeStream = function(stream) {\n var pc = this;\n stream.getTracks().forEach(function(track) {\n var sender = pc.getSenders().find(function(s) {\n return s.track === track;\n });\n if (sender) {\n pc.removeTrack(sender);\n }\n });\n };\n\n RTCPeerConnection.prototype.getSenders = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpSender;\n })\n .map(function(transceiver) {\n return transceiver.rtpSender;\n });\n };\n\n RTCPeerConnection.prototype.getReceivers = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpReceiver;\n })\n .map(function(transceiver) {\n return transceiver.rtpReceiver;\n });\n };\n\n\n RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex,\n usingBundle) {\n var pc = this;\n if (usingBundle && sdpMLineIndex > 0) {\n return this.transceivers[0].iceGatherer;\n } else if (this._iceGatherers.length) {\n return this._iceGatherers.shift();\n }\n var iceGatherer = new window.RTCIceGatherer({\n iceServers: this._config.iceServers,\n gatherPolicy: this._config.iceTransportPolicy\n });\n Object.defineProperty(iceGatherer, 'state',\n {value: 'new', writable: true}\n );\n\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = [];\n this.transceivers[sdpMLineIndex].bufferCandidates = function(event) {\n var end = !event.candidate || Object.keys(event.candidate).length === 0;\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n iceGatherer.state = end ? 'completed' : 'gathering';\n if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) {\n pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event);\n }\n };\n iceGatherer.addEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n return iceGatherer;\n };\n\n // start gathering from an RTCIceGatherer.\n RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) {\n var pc = this;\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer.onlocalcandidate) {\n return;\n }\n var bufferedCandidateEvents =\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents;\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null;\n iceGatherer.removeEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n iceGatherer.onlocalcandidate = function(evt) {\n if (pc.usingBundle && sdpMLineIndex > 0) {\n // if we know that we use bundle we can drop candidates with\n // ѕdpMLineIndex > 0. If we don't do this then our state gets\n // confused since we dispose the extra ice gatherer.\n return;\n }\n var event = new Event('icecandidate');\n event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex};\n\n var cand = evt.candidate;\n // Edge emits an empty object for RTCIceCandidateComplete‥\n var end = !cand || Object.keys(cand).length === 0;\n if (end) {\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') {\n iceGatherer.state = 'completed';\n }\n } else {\n if (iceGatherer.state === 'new') {\n iceGatherer.state = 'gathering';\n }\n // RTCIceCandidate doesn't have a component, needs to be added\n cand.component = 1;\n // also the usernameFragment. TODO: update SDP to take both variants.\n cand.ufrag = iceGatherer.getLocalParameters().usernameFragment;\n\n var serializedCandidate = SDPUtils.writeCandidate(cand);\n event.candidate = Object.assign(event.candidate,\n SDPUtils.parseCandidate(serializedCandidate));\n\n event.candidate.candidate = serializedCandidate;\n event.candidate.toJSON = function() {\n return {\n candidate: event.candidate.candidate,\n sdpMid: event.candidate.sdpMid,\n sdpMLineIndex: event.candidate.sdpMLineIndex,\n usernameFragment: event.candidate.usernameFragment\n };\n };\n }\n\n // update local description.\n var sections = SDPUtils.getMediaSections(pc._localDescription.sdp);\n if (!end) {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=' + event.candidate.candidate + '\\r\\n';\n } else {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=end-of-candidates\\r\\n';\n }\n pc._localDescription.sdp =\n SDPUtils.getDescription(pc._localDescription.sdp) +\n sections.join('');\n var complete = pc.transceivers.every(function(transceiver) {\n return transceiver.iceGatherer &&\n transceiver.iceGatherer.state === 'completed';\n });\n\n if (pc.iceGatheringState !== 'gathering') {\n pc.iceGatheringState = 'gathering';\n pc._emitGatheringStateChange();\n }\n\n // Emit candidate. Also emit null candidate when all gatherers are\n // complete.\n if (!end) {\n pc._dispatchEvent('icecandidate', event);\n }\n if (complete) {\n pc._dispatchEvent('icecandidate', new Event('icecandidate'));\n pc.iceGatheringState = 'complete';\n pc._emitGatheringStateChange();\n }\n };\n\n // emit already gathered candidates.\n window.setTimeout(function() {\n bufferedCandidateEvents.forEach(function(e) {\n iceGatherer.onlocalcandidate(e);\n });\n }, 0);\n };\n\n // Create ICE transport and DTLS transport.\n RTCPeerConnection.prototype._createIceAndDtlsTransports = function() {\n var pc = this;\n var iceTransport = new window.RTCIceTransport(null);\n iceTransport.onicestatechange = function() {\n pc._updateIceConnectionState();\n pc._updateConnectionState();\n };\n\n var dtlsTransport = new window.RTCDtlsTransport(iceTransport);\n dtlsTransport.ondtlsstatechange = function() {\n pc._updateConnectionState();\n };\n dtlsTransport.onerror = function() {\n // onerror does not set state to failed by itself.\n Object.defineProperty(dtlsTransport, 'state',\n {value: 'failed', writable: true});\n pc._updateConnectionState();\n };\n\n return {\n iceTransport: iceTransport,\n dtlsTransport: dtlsTransport\n };\n };\n\n // Destroy ICE gatherer, ICE transport and DTLS transport.\n // Without triggering the callbacks.\n RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function(\n sdpMLineIndex) {\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer) {\n delete iceGatherer.onlocalcandidate;\n delete this.transceivers[sdpMLineIndex].iceGatherer;\n }\n var iceTransport = this.transceivers[sdpMLineIndex].iceTransport;\n if (iceTransport) {\n delete iceTransport.onicestatechange;\n delete this.transceivers[sdpMLineIndex].iceTransport;\n }\n var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport;\n if (dtlsTransport) {\n delete dtlsTransport.ondtlsstatechange;\n delete dtlsTransport.onerror;\n delete this.transceivers[sdpMLineIndex].dtlsTransport;\n }\n };\n\n // Start the RTP Sender and Receiver for a transceiver.\n RTCPeerConnection.prototype._transceive = function(transceiver,\n send, recv) {\n var params = getCommonCapabilities(transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n if (send && transceiver.rtpSender) {\n params.encodings = transceiver.sendEncodingParameters;\n params.rtcp = {\n cname: SDPUtils.localCName,\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.recvEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc;\n }\n transceiver.rtpSender.send(params);\n }\n if (recv && transceiver.rtpReceiver && params.codecs.length > 0) {\n // remove RTX field in Edge 14942\n if (transceiver.kind === 'video'\n && transceiver.recvEncodingParameters\n && edgeVersion < 15019) {\n transceiver.recvEncodingParameters.forEach(function(p) {\n delete p.rtx;\n });\n }\n if (transceiver.recvEncodingParameters.length) {\n params.encodings = transceiver.recvEncodingParameters;\n } else {\n params.encodings = [{}];\n }\n params.rtcp = {\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.rtcpParameters.cname) {\n params.rtcp.cname = transceiver.rtcpParameters.cname;\n }\n if (transceiver.sendEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc;\n }\n transceiver.rtpReceiver.receive(params);\n }\n };\n\n RTCPeerConnection.prototype.setLocalDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setLocalDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set local ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var sections;\n var sessionpart;\n if (description.type === 'offer') {\n // VERY limited support for SDP munging. Limited to:\n // * changing the order of codecs\n sections = SDPUtils.splitSections(description.sdp);\n sessionpart = sections.shift();\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var caps = SDPUtils.parseRtpParameters(mediaSection);\n pc.transceivers[sdpMLineIndex].localCapabilities = caps;\n });\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n });\n } else if (description.type === 'answer') {\n sections = SDPUtils.splitSections(pc._remoteDescription.sdp);\n sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var transceiver = pc.transceivers[sdpMLineIndex];\n var iceGatherer = transceiver.iceGatherer;\n var iceTransport = transceiver.iceTransport;\n var dtlsTransport = transceiver.dtlsTransport;\n var localCapabilities = transceiver.localCapabilities;\n var remoteCapabilities = transceiver.remoteCapabilities;\n\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n\n if (!rejected && !transceiver.rejected) {\n var remoteIceParameters = SDPUtils.getIceParameters(\n mediaSection, sessionpart);\n var remoteDtlsParameters = SDPUtils.getDtlsParameters(\n mediaSection, sessionpart);\n if (isIceLite) {\n remoteDtlsParameters.role = 'server';\n }\n\n if (!pc.usingBundle || sdpMLineIndex === 0) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n isIceLite ? 'controlling' : 'controlled');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // Calculate intersection of capabilities.\n var params = getCommonCapabilities(localCapabilities,\n remoteCapabilities);\n\n // Start the RTCRtpSender. The RTCRtpReceiver for this\n // transceiver has already been started in setRemoteDescription.\n pc._transceive(transceiver,\n params.codecs.length > 0,\n false);\n }\n });\n }\n\n pc._localDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-local-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.setRemoteDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setRemoteDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set remote ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var streams = {};\n pc.remoteStreams.forEach(function(stream) {\n streams[stream.id] = stream;\n });\n var receiverList = [];\n var sections = SDPUtils.splitSections(description.sdp);\n var sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n var usingBundle = SDPUtils.matchPrefix(sessionpart,\n 'a=group:BUNDLE ').length > 0;\n pc.usingBundle = usingBundle;\n var iceOptions = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-options:')[0];\n if (iceOptions) {\n pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ')\n .indexOf('trickle') >= 0;\n } else {\n pc.canTrickleIceCandidates = false;\n }\n\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var lines = SDPUtils.splitLines(mediaSection);\n var kind = SDPUtils.getKind(mediaSection);\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n var protocol = lines[0].substr(2).split(' ')[2];\n\n var direction = SDPUtils.getDirection(mediaSection, sessionpart);\n var remoteMsid = SDPUtils.parseMsid(mediaSection);\n\n var mid = SDPUtils.getMid(mediaSection) || SDPUtils.generateIdentifier();\n\n // Reject datachannels which are not implemented yet.\n if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' ||\n protocol === 'UDP/DTLS/SCTP'))) {\n // TODO: this is dangerous in the case where a non-rejected m-line\n // becomes rejected.\n pc.transceivers[sdpMLineIndex] = {\n mid: mid,\n kind: kind,\n protocol: protocol,\n rejected: true\n };\n return;\n }\n\n if (!rejected && pc.transceivers[sdpMLineIndex] &&\n pc.transceivers[sdpMLineIndex].rejected) {\n // recycle a rejected transceiver.\n pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true);\n }\n\n var transceiver;\n var iceGatherer;\n var iceTransport;\n var dtlsTransport;\n var rtpReceiver;\n var sendEncodingParameters;\n var recvEncodingParameters;\n var localCapabilities;\n\n var track;\n // FIXME: ensure the mediaSection has rtcp-mux set.\n var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection);\n var remoteIceParameters;\n var remoteDtlsParameters;\n if (!rejected) {\n remoteIceParameters = SDPUtils.getIceParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters.role = 'client';\n }\n recvEncodingParameters =\n SDPUtils.parseRtpEncodingParameters(mediaSection);\n\n var rtcpParameters = SDPUtils.parseRtcpParameters(mediaSection);\n\n var isComplete = SDPUtils.matchPrefix(mediaSection,\n 'a=end-of-candidates', sessionpart).length > 0;\n var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:')\n .map(function(cand) {\n return SDPUtils.parseCandidate(cand);\n })\n .filter(function(cand) {\n return cand.component === 1;\n });\n\n // Check if we can use BUNDLE and dispose transports.\n if ((description.type === 'offer' || description.type === 'answer') &&\n !rejected && usingBundle && sdpMLineIndex > 0 &&\n pc.transceivers[sdpMLineIndex]) {\n pc._disposeIceAndDtlsTransports(sdpMLineIndex);\n pc.transceivers[sdpMLineIndex].iceGatherer =\n pc.transceivers[0].iceGatherer;\n pc.transceivers[sdpMLineIndex].iceTransport =\n pc.transceivers[0].iceTransport;\n pc.transceivers[sdpMLineIndex].dtlsTransport =\n pc.transceivers[0].dtlsTransport;\n if (pc.transceivers[sdpMLineIndex].rtpSender) {\n pc.transceivers[sdpMLineIndex].rtpSender.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n if (pc.transceivers[sdpMLineIndex].rtpReceiver) {\n pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n }\n if (description.type === 'offer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex] ||\n pc._createTransceiver(kind);\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n usingBundle);\n }\n\n if (cands.length && transceiver.iceTransport.state === 'new') {\n if (isComplete && (!usingBundle || sdpMLineIndex === 0)) {\n transceiver.iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n localCapabilities = window.RTCRtpReceiver.getCapabilities(kind);\n\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n\n sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 2) * 1001\n }];\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n var isNewTrack = false;\n if (direction === 'sendrecv' || direction === 'sendonly') {\n isNewTrack = !transceiver.rtpReceiver;\n rtpReceiver = transceiver.rtpReceiver ||\n new window.RTCRtpReceiver(transceiver.dtlsTransport, kind);\n\n if (isNewTrack) {\n var stream;\n track = rtpReceiver.track;\n // FIXME: does not work with Plan B.\n if (remoteMsid && remoteMsid.stream === '-') {\n // no-op. a stream id of '-' means: no associated stream.\n } else if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n Object.defineProperty(streams[remoteMsid.stream], 'id', {\n get: function() {\n return remoteMsid.stream;\n }\n });\n }\n Object.defineProperty(track, 'id', {\n get: function() {\n return remoteMsid.track;\n }\n });\n stream = streams[remoteMsid.stream];\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n stream = streams.default;\n }\n if (stream) {\n addTrackToStreamAndFireEvent(track, stream);\n transceiver.associatedRemoteMediaStreams.push(stream);\n }\n receiverList.push([track, rtpReceiver, stream]);\n }\n } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) {\n transceiver.associatedRemoteMediaStreams.forEach(function(s) {\n var nativeTrack = s.getTracks().find(function(t) {\n return t.id === transceiver.rtpReceiver.track.id;\n });\n if (nativeTrack) {\n removeTrackFromStreamAndFireEvent(nativeTrack, s);\n }\n });\n transceiver.associatedRemoteMediaStreams = [];\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.remoteCapabilities = remoteCapabilities;\n transceiver.rtpReceiver = rtpReceiver;\n transceiver.rtcpParameters = rtcpParameters;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n transceiver.recvEncodingParameters = recvEncodingParameters;\n\n // Start the RTCRtpReceiver now. The RTPSender is started in\n // setLocalDescription.\n pc._transceive(pc.transceivers[sdpMLineIndex],\n false,\n isNewTrack);\n } else if (description.type === 'answer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex];\n iceGatherer = transceiver.iceGatherer;\n iceTransport = transceiver.iceTransport;\n dtlsTransport = transceiver.dtlsTransport;\n rtpReceiver = transceiver.rtpReceiver;\n sendEncodingParameters = transceiver.sendEncodingParameters;\n localCapabilities = transceiver.localCapabilities;\n\n pc.transceivers[sdpMLineIndex].recvEncodingParameters =\n recvEncodingParameters;\n pc.transceivers[sdpMLineIndex].remoteCapabilities =\n remoteCapabilities;\n pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters;\n\n if (cands.length && iceTransport.state === 'new') {\n if ((isIceLite || isComplete) &&\n (!usingBundle || sdpMLineIndex === 0)) {\n iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n if (!usingBundle || sdpMLineIndex === 0) {\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n 'controlling');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // If the offer contained RTX but the answer did not,\n // remove RTX from sendEncodingParameters.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n pc._transceive(transceiver,\n direction === 'sendrecv' || direction === 'recvonly',\n direction === 'sendrecv' || direction === 'sendonly');\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n if (rtpReceiver &&\n (direction === 'sendrecv' || direction === 'sendonly')) {\n track = rtpReceiver.track;\n if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]);\n receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]);\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams.default);\n receiverList.push([track, rtpReceiver, streams.default]);\n }\n } else {\n // FIXME: actually the receiver should be created later.\n delete transceiver.rtpReceiver;\n }\n }\n });\n\n if (pc._dtlsRole === undefined) {\n pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive';\n }\n\n pc._remoteDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-remote-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n Object.keys(streams).forEach(function(sid) {\n var stream = streams[sid];\n if (stream.getTracks().length) {\n if (pc.remoteStreams.indexOf(stream) === -1) {\n pc.remoteStreams.push(stream);\n var event = new Event('addstream');\n event.stream = stream;\n window.setTimeout(function() {\n pc._dispatchEvent('addstream', event);\n });\n }\n\n receiverList.forEach(function(item) {\n var track = item[0];\n var receiver = item[1];\n if (stream.id !== item[2].id) {\n return;\n }\n fireAddTrack(pc, track, receiver, [stream]);\n });\n }\n });\n receiverList.forEach(function(item) {\n if (item[2]) {\n return;\n }\n fireAddTrack(pc, item[0], item[1], []);\n });\n\n // check whether addIceCandidate({}) was called within four seconds after\n // setRemoteDescription.\n window.setTimeout(function() {\n if (!(pc && pc.transceivers)) {\n return;\n }\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport &&\n transceiver.iceTransport.state === 'new' &&\n transceiver.iceTransport.getRemoteCandidates().length > 0) {\n console.warn('Timeout for addRemoteCandidate. Consider sending ' +\n 'an end-of-candidates notification');\n transceiver.iceTransport.addRemoteCandidate({});\n }\n });\n }, 4000);\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.close = function() {\n this.transceivers.forEach(function(transceiver) {\n /* not yet\n if (transceiver.iceGatherer) {\n transceiver.iceGatherer.close();\n }\n */\n if (transceiver.iceTransport) {\n transceiver.iceTransport.stop();\n }\n if (transceiver.dtlsTransport) {\n transceiver.dtlsTransport.stop();\n }\n if (transceiver.rtpSender) {\n transceiver.rtpSender.stop();\n }\n if (transceiver.rtpReceiver) {\n transceiver.rtpReceiver.stop();\n }\n });\n // FIXME: clean up tracks, local streams, remote streams, etc\n this._isClosed = true;\n this._updateSignalingState('closed');\n };\n\n // Update the signaling state.\n RTCPeerConnection.prototype._updateSignalingState = function(newState) {\n this.signalingState = newState;\n var event = new Event('signalingstatechange');\n this._dispatchEvent('signalingstatechange', event);\n };\n\n // Determine whether to fire the negotiationneeded event.\n RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() {\n var pc = this;\n if (this.signalingState !== 'stable' || this.needNegotiation === true) {\n return;\n }\n this.needNegotiation = true;\n window.setTimeout(function() {\n if (pc.needNegotiation) {\n pc.needNegotiation = false;\n var event = new Event('negotiationneeded');\n pc._dispatchEvent('negotiationneeded', event);\n }\n }, 0);\n };\n\n // Update the ice connection state.\n RTCPeerConnection.prototype._updateIceConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n checking: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n }\n });\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.checking > 0) {\n newState = 'checking';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n } else if (states.completed > 0) {\n newState = 'completed';\n }\n\n if (newState !== this.iceConnectionState) {\n this.iceConnectionState = newState;\n var event = new Event('iceconnectionstatechange');\n this._dispatchEvent('iceconnectionstatechange', event);\n }\n };\n\n // Update the connection state.\n RTCPeerConnection.prototype._updateConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n connecting: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && transceiver.dtlsTransport &&\n !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n states[transceiver.dtlsTransport.state]++;\n }\n });\n // ICETransport.completed and connected are the same for this purpose.\n states.connected += states.completed;\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.connecting > 0) {\n newState = 'connecting';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n }\n\n if (newState !== this.connectionState) {\n this.connectionState = newState;\n var event = new Event('connectionstatechange');\n this._dispatchEvent('connectionstatechange', event);\n }\n };\n\n RTCPeerConnection.prototype.createOffer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createOffer after close'));\n }\n\n var numAudioTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'audio';\n }).length;\n var numVideoTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'video';\n }).length;\n\n // Determine number of audio and video tracks we need to send/recv.\n var offerOptions = arguments[0];\n if (offerOptions) {\n // Reject Chrome legacy constraints.\n if (offerOptions.mandatory || offerOptions.optional) {\n throw new TypeError(\n 'Legacy mandatory/optional constraints not supported.');\n }\n if (offerOptions.offerToReceiveAudio !== undefined) {\n if (offerOptions.offerToReceiveAudio === true) {\n numAudioTracks = 1;\n } else if (offerOptions.offerToReceiveAudio === false) {\n numAudioTracks = 0;\n } else {\n numAudioTracks = offerOptions.offerToReceiveAudio;\n }\n }\n if (offerOptions.offerToReceiveVideo !== undefined) {\n if (offerOptions.offerToReceiveVideo === true) {\n numVideoTracks = 1;\n } else if (offerOptions.offerToReceiveVideo === false) {\n numVideoTracks = 0;\n } else {\n numVideoTracks = offerOptions.offerToReceiveVideo;\n }\n }\n }\n\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.kind === 'audio') {\n numAudioTracks--;\n if (numAudioTracks < 0) {\n transceiver.wantReceive = false;\n }\n } else if (transceiver.kind === 'video') {\n numVideoTracks--;\n if (numVideoTracks < 0) {\n transceiver.wantReceive = false;\n }\n }\n });\n\n // Create M-lines for recvonly streams.\n while (numAudioTracks > 0 || numVideoTracks > 0) {\n if (numAudioTracks > 0) {\n pc._createTransceiver('audio');\n numAudioTracks--;\n }\n if (numVideoTracks > 0) {\n pc._createTransceiver('video');\n numVideoTracks--;\n }\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n // For each track, create an ice gatherer, ice transport,\n // dtls transport, potentially rtpsender and rtpreceiver.\n var track = transceiver.track;\n var kind = transceiver.kind;\n var mid = transceiver.mid || SDPUtils.generateIdentifier();\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n pc.usingBundle);\n }\n\n var localCapabilities = window.RTCRtpSender.getCapabilities(kind);\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n localCapabilities.codecs.forEach(function(codec) {\n // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552\n // by adding level-asymmetry-allowed=1\n if (codec.name === 'H264' &&\n codec.parameters['level-asymmetry-allowed'] === undefined) {\n codec.parameters['level-asymmetry-allowed'] = '1';\n }\n\n // for subsequent offers, we might have to re-use the payload\n // type of the last offer.\n if (transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.codecs) {\n transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) {\n if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() &&\n codec.clockRate === remoteCodec.clockRate) {\n codec.preferredPayloadType = remoteCodec.payloadType;\n }\n });\n }\n });\n localCapabilities.headerExtensions.forEach(function(hdrExt) {\n var remoteExtensions = transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.headerExtensions || [];\n remoteExtensions.forEach(function(rHdrExt) {\n if (hdrExt.uri === rHdrExt.uri) {\n hdrExt.id = rHdrExt.id;\n }\n });\n });\n\n // generate an ssrc now, to be used later in rtpSender.send\n var sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 1) * 1001\n }];\n if (track) {\n // add RTX\n if (edgeVersion >= 15019 && kind === 'video' &&\n !sendEncodingParameters[0].rtx) {\n sendEncodingParameters[0].rtx = {\n ssrc: sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n\n if (transceiver.wantReceive) {\n transceiver.rtpReceiver = new window.RTCRtpReceiver(\n transceiver.dtlsTransport, kind);\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n });\n\n // always offer BUNDLE and dispose on return if not supported.\n if (pc._config.bundlePolicy !== 'max-compat') {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n sdp += writeMediaSection(transceiver, transceiver.localCapabilities,\n 'offer', transceiver.stream, pc._dtlsRole);\n sdp += 'a=rtcp-rsize\\r\\n';\n\n if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' &&\n (sdpMLineIndex === 0 || !pc.usingBundle)) {\n transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) {\n cand.component = 1;\n sdp += 'a=' + SDPUtils.writeCandidate(cand) + '\\r\\n';\n });\n\n if (transceiver.iceGatherer.state === 'completed') {\n sdp += 'a=end-of-candidates\\r\\n';\n }\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'offer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.createAnswer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer after close'));\n }\n\n if (!(pc.signalingState === 'have-remote-offer' ||\n pc.signalingState === 'have-local-pranswer')) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer in signalingState ' + pc.signalingState));\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n if (pc.usingBundle) {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n var mediaSectionsInOffer = SDPUtils.getMediaSections(\n pc._remoteDescription.sdp).length;\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n if (sdpMLineIndex + 1 > mediaSectionsInOffer) {\n return;\n }\n if (transceiver.rejected) {\n if (transceiver.kind === 'application') {\n if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt\n sdp += 'm=application 0 DTLS/SCTP 5000\\r\\n';\n } else {\n sdp += 'm=application 0 ' + transceiver.protocol +\n ' webrtc-datachannel\\r\\n';\n }\n } else if (transceiver.kind === 'audio') {\n sdp += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\\r\\n' +\n 'a=rtpmap:0 PCMU/8000\\r\\n';\n } else if (transceiver.kind === 'video') {\n sdp += 'm=video 0 UDP/TLS/RTP/SAVPF 120\\r\\n' +\n 'a=rtpmap:120 VP8/90000\\r\\n';\n }\n sdp += 'c=IN IP4 0.0.0.0\\r\\n' +\n 'a=inactive\\r\\n' +\n 'a=mid:' + transceiver.mid + '\\r\\n';\n return;\n }\n\n // FIXME: look at direction.\n if (transceiver.stream) {\n var localTrack;\n if (transceiver.kind === 'audio') {\n localTrack = transceiver.stream.getAudioTracks()[0];\n } else if (transceiver.kind === 'video') {\n localTrack = transceiver.stream.getVideoTracks()[0];\n }\n if (localTrack) {\n // add RTX\n if (edgeVersion >= 15019 && transceiver.kind === 'video' &&\n !transceiver.sendEncodingParameters[0].rtx) {\n transceiver.sendEncodingParameters[0].rtx = {\n ssrc: transceiver.sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n }\n\n // Calculate intersection of capabilities.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n sdp += writeMediaSection(transceiver, commonCapabilities,\n 'answer', transceiver.stream, pc._dtlsRole);\n if (transceiver.rtcpParameters &&\n transceiver.rtcpParameters.reducedSize) {\n sdp += 'a=rtcp-rsize\\r\\n';\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'answer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.addIceCandidate = function(candidate) {\n var pc = this;\n var sections;\n if (candidate && !(candidate.sdpMLineIndex !== undefined ||\n candidate.sdpMid)) {\n return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required'));\n }\n\n // TODO: needs to go into ops queue.\n return new Promise(function(resolve, reject) {\n if (!pc._remoteDescription) {\n return reject(makeError('InvalidStateError',\n 'Can not add ICE candidate without a remote description'));\n } else if (!candidate || candidate.candidate === '') {\n for (var j = 0; j < pc.transceivers.length; j++) {\n if (pc.transceivers[j].rejected) {\n continue;\n }\n pc.transceivers[j].iceTransport.addRemoteCandidate({});\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[j] += 'a=end-of-candidates\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n if (pc.usingBundle) {\n break;\n }\n }\n } else {\n var sdpMLineIndex = candidate.sdpMLineIndex;\n if (candidate.sdpMid) {\n for (var i = 0; i < pc.transceivers.length; i++) {\n if (pc.transceivers[i].mid === candidate.sdpMid) {\n sdpMLineIndex = i;\n break;\n }\n }\n }\n var transceiver = pc.transceivers[sdpMLineIndex];\n if (transceiver) {\n if (transceiver.rejected) {\n return resolve();\n }\n var cand = Object.keys(candidate.candidate).length > 0 ?\n SDPUtils.parseCandidate(candidate.candidate) : {};\n // Ignore Chrome's invalid candidates since Edge does not like them.\n if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) {\n return resolve();\n }\n // Ignore RTCP candidates, we assume RTCP-MUX.\n if (cand.component && cand.component !== 1) {\n return resolve();\n }\n // when using bundle, avoid adding candidates to the wrong\n // ice transport. And avoid adding candidates added in the SDP.\n if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 &&\n transceiver.iceTransport !== pc.transceivers[0].iceTransport)) {\n if (!maybeAddCandidate(transceiver.iceTransport, cand)) {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n\n // update the remoteDescription.\n var candidateString = candidate.candidate.trim();\n if (candidateString.indexOf('a=') === 0) {\n candidateString = candidateString.substr(2);\n }\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[sdpMLineIndex] += 'a=' +\n (cand.type ? candidateString : 'end-of-candidates')\n + '\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n } else {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n resolve();\n });\n };\n\n RTCPeerConnection.prototype.getStats = function(selector) {\n if (selector && selector instanceof window.MediaStreamTrack) {\n var senderOrReceiver = null;\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.rtpSender &&\n transceiver.rtpSender.track === selector) {\n senderOrReceiver = transceiver.rtpSender;\n } else if (transceiver.rtpReceiver &&\n transceiver.rtpReceiver.track === selector) {\n senderOrReceiver = transceiver.rtpReceiver;\n }\n });\n if (!senderOrReceiver) {\n throw makeError('InvalidAccessError', 'Invalid selector.');\n }\n return senderOrReceiver.getStats();\n }\n\n var promises = [];\n this.transceivers.forEach(function(transceiver) {\n ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport',\n 'dtlsTransport'].forEach(function(method) {\n if (transceiver[method]) {\n promises.push(transceiver[method].getStats());\n }\n });\n });\n return Promise.all(promises).then(function(allStats) {\n var results = new Map();\n allStats.forEach(function(stats) {\n stats.forEach(function(stat) {\n results.set(stat.id, stat);\n });\n });\n return results;\n });\n };\n\n // fix low-level stat names and return Map instead of object.\n var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer',\n 'RTCIceTransport', 'RTCDtlsTransport'];\n ortcObjects.forEach(function(ortcObjectName) {\n var obj = window[ortcObjectName];\n if (obj && obj.prototype && obj.prototype.getStats) {\n var nativeGetstats = obj.prototype.getStats;\n obj.prototype.getStats = function() {\n return nativeGetstats.apply(this)\n .then(function(nativeStats) {\n var mapStats = new Map();\n Object.keys(nativeStats).forEach(function(id) {\n nativeStats[id].type = fixStatsType(nativeStats[id]);\n mapStats.set(id, nativeStats[id]);\n });\n return mapStats;\n });\n };\n }\n });\n\n // legacy callback shims. Should be moved to adapter.js some days.\n var methods = ['createOffer', 'createAnswer'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[0] === 'function' ||\n typeof args[1] === 'function') { // legacy\n return nativeMethod.apply(this, [arguments[2]])\n .then(function(description) {\n if (typeof args[0] === 'function') {\n args[0].apply(null, [description]);\n }\n }, function(error) {\n if (typeof args[1] === 'function') {\n args[1].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function' ||\n typeof args[2] === 'function') { // legacy\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n }, function(error) {\n if (typeof args[2] === 'function') {\n args[2].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n // getStats is special. It doesn't have a spec legacy method yet we support\n // getStats(something, cb) without error callbacks.\n ['getStats'].forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function') {\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n return RTCPeerConnection;\n};\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n const shimError_ = function(e) {\n return {\n name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name,\n message: e.message,\n constraint: e.constraint,\n toString() {\n return this.name;\n }\n };\n };\n\n // getUserMedia error shim.\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e)));\n };\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window) {\n if (!('getDisplayMedia' in window.navigator)) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n window.navigator.getDisplayMedia.bind(window.navigator);\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nimport {filterIceServers} from './filtericeservers';\nimport shimRTCPeerConnection from 'rtcpeerconnection-shim';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimPeerConnection(window, browserDetails) {\n if (window.RTCIceGatherer) {\n if (!window.RTCIceCandidate) {\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n return args;\n };\n }\n if (!window.RTCSessionDescription) {\n window.RTCSessionDescription = function RTCSessionDescription(args) {\n return args;\n };\n }\n // this adds an additional event listener to MediaStrackTrack that signals\n // when a tracks enabled property was changed. Workaround for a bug in\n // addStream, see below. No longer required in 15025+\n if (browserDetails.version < 15025) {\n const origMSTEnabled = Object.getOwnPropertyDescriptor(\n window.MediaStreamTrack.prototype, 'enabled');\n Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', {\n set(value) {\n origMSTEnabled.set.call(this, value);\n const ev = new Event('enabled');\n ev.enabled = value;\n this.dispatchEvent(ev);\n }\n });\n }\n }\n\n // ORTC defines the DTMF sender a bit different.\n // https://github.com/w3c/ortc/issues/714\n if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) {\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = new window.RTCDtmfSender(this);\n } else if (this.track.kind === 'video') {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n // Edge currently only implements the RTCDtmfSender, not the\n // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2*\n if (window.RTCDtmfSender && !window.RTCDTMFSender) {\n window.RTCDTMFSender = window.RTCDtmfSender;\n }\n\n const RTCPeerConnectionShim = shimRTCPeerConnection(window,\n browserDetails.version);\n window.RTCPeerConnection = function RTCPeerConnection(config) {\n if (config && config.iceServers) {\n config.iceServers = filterIceServers(config.iceServers,\n browserDetails.version);\n utils.log('ICE servers after filtering:', config.iceServers);\n }\n return new RTCPeerConnectionShim(config);\n };\n window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype;\n}\n\nexport function shimReplaceTrack(window) {\n // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614\n if (window.RTCRtpSender &&\n !('replaceTrack' in window.RTCRtpSender.prototype)) {\n window.RTCRtpSender.prototype.replaceTrack =\n window.RTCRtpSender.prototype.setTrack;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n const MediaStreamTrack = window && window.MediaStreamTrack;\n\n navigator.getUserMedia = function(constraints, onSuccess, onError) {\n // Replace Firefox 44+'s deprecation warning with unprefixed version.\n utils.deprecated('navigator.getUserMedia',\n 'navigator.mediaDevices.getUserMedia');\n navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError);\n };\n\n if (!(browserDetails.version > 55 &&\n 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n\n const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n if (typeof c === 'object' && typeof c.audio === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c.audio, 'autoGainControl', 'mozAutoGainControl');\n remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeGetUserMedia(c);\n };\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) {\n const nativeGetSettings = MediaStreamTrack.prototype.getSettings;\n MediaStreamTrack.prototype.getSettings = function() {\n const obj = nativeGetSettings.apply(this, arguments);\n remap(obj, 'mozAutoGainControl', 'autoGainControl');\n remap(obj, 'mozNoiseSuppression', 'noiseSuppression');\n return obj;\n };\n }\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) {\n const nativeApplyConstraints =\n MediaStreamTrack.prototype.applyConstraints;\n MediaStreamTrack.prototype.applyConstraints = function(c) {\n if (this.kind === 'audio' && typeof c === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c, 'autoGainControl', 'mozAutoGainControl');\n remap(c, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeApplyConstraints.apply(this, [c]);\n };\n }\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window, preferredMediaSource) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n if (!(constraints && constraints.video)) {\n const err = new DOMException('getDisplayMedia without video ' +\n 'constraints is undefined');\n err.name = 'NotFoundError';\n // from https://heycam.github.io/webidl/#idl-DOMException-error-names\n err.code = 8;\n return Promise.reject(err);\n }\n if (constraints.video === true) {\n constraints.video = {mediaSource: preferredMediaSource};\n } else {\n constraints.video.mediaSource = preferredMediaSource;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCTrackEvent &&\n ('receiver' in window.RTCTrackEvent.prototype) &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (typeof window !== 'object' ||\n !(window.RTCPeerConnection || window.mozRTCPeerConnection)) {\n return; // probably media.peerconnection.enabled=false in about:config\n }\n if (!window.RTCPeerConnection && window.mozRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.mozRTCPeerConnection;\n }\n\n if (browserDetails.version < 53) {\n // shim away need for obsolete RTCIceCandidate/RTCSessionDescription.\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n\n const modernStatsTypes = {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n };\n\n const nativeGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n return nativeGetStats.apply(this, [selector || null])\n .then(stats => {\n if (browserDetails.version < 53 && !onSucc) {\n // Shim only promise getStats with spec-hyphens in type names\n // Leave callback version alone; misc old uses of forEach before Map\n try {\n stats.forEach(stat => {\n stat.type = modernStatsTypes[stat.type] || stat.type;\n });\n } catch (e) {\n if (e.name !== 'TypeError') {\n throw e;\n }\n // Avoid TypeError: \"type\" is read-only, in old versions. 34-43ish\n stats.forEach((stat, i) => {\n stats.set(i, Object.assign({}, stat, {\n type: modernStatsTypes[stat.type] || stat.type\n }));\n });\n }\n }\n return stats;\n })\n .then(onSucc, onErr);\n };\n}\n\nexport function shimSenderGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) {\n return;\n }\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n return this.track ? this._pc.getStats(this.track) :\n Promise.resolve(new Map());\n };\n}\n\nexport function shimReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) {\n return;\n }\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers = function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n return this._pc.getStats(this.track);\n };\n}\n\nexport function shimRemoveStream(window) {\n if (!window.RTCPeerConnection ||\n 'removeStream' in window.RTCPeerConnection.prototype) {\n return;\n }\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n utils.deprecated('removeStream', 'removeTrack');\n this.getSenders().forEach(sender => {\n if (sender.track && stream.getTracks().includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n}\n\nexport function shimRTCDataChannel(window) {\n // rename DataChannel to RTCDataChannel (native fix in FF60):\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851\n if (window.DataChannel && !window.RTCDataChannel) {\n window.RTCDataChannel = window.DataChannel;\n }\n}\n\nexport function shimAddTransceiver(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver;\n if (origAddTransceiver) {\n window.RTCPeerConnection.prototype.addTransceiver =\n function addTransceiver() {\n this.setParametersPromises = [];\n const initParameters = arguments[1];\n const shouldPerformCheck = initParameters &&\n 'sendEncodings' in initParameters;\n if (shouldPerformCheck) {\n // If sendEncodings params are provided, validate grammar\n initParameters.sendEncodings.forEach((encodingParam) => {\n if ('rid' in encodingParam) {\n const ridRegex = /^[a-z0-9]{0,16}$/i;\n if (!ridRegex.test(encodingParam.rid)) {\n throw new TypeError('Invalid RID value provided.');\n }\n }\n if ('scaleResolutionDownBy' in encodingParam) {\n if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) {\n throw new RangeError('scale_resolution_down_by must be >= 1.0');\n }\n }\n if ('maxFramerate' in encodingParam) {\n if (!(parseFloat(encodingParam.maxFramerate) >= 0)) {\n throw new RangeError('max_framerate must be >= 0.0');\n }\n }\n });\n }\n const transceiver = origAddTransceiver.apply(this, arguments);\n if (shouldPerformCheck) {\n // Check if the init options were applied. If not we do this in an\n // asynchronous way and save the promise reference in a global object.\n // This is an ugly hack, but at the same time is way more robust than\n // checking the sender parameters before and after the createOffer\n // Also note that after the createoffer we are not 100% sure that\n // the params were asynchronously applied so we might miss the\n // opportunity to recreate offer.\n const {sender} = transceiver;\n const params = sender.getParameters();\n if (!('encodings' in params) ||\n // Avoid being fooled by patched getParameters() below.\n (params.encodings.length === 1 &&\n Object.keys(params.encodings[0]).length === 0)) {\n params.encodings = initParameters.sendEncodings;\n sender.sendEncodings = initParameters.sendEncodings;\n this.setParametersPromises.push(sender.setParameters(params)\n .then(() => {\n delete sender.sendEncodings;\n }).catch(() => {\n delete sender.sendEncodings;\n })\n );\n }\n }\n return transceiver;\n };\n }\n}\n\nexport function shimGetParameters(window) {\n if (!(typeof window === 'object' && window.RTCRtpSender)) {\n return;\n }\n const origGetParameters = window.RTCRtpSender.prototype.getParameters;\n if (origGetParameters) {\n window.RTCRtpSender.prototype.getParameters =\n function getParameters() {\n const params = origGetParameters.apply(this, arguments);\n if (!('encodings' in params)) {\n params.encodings = [].concat(this.sendEncodings || [{}]);\n }\n return params;\n };\n }\n}\n\nexport function shimCreateOffer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer = function createOffer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateOffer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimCreateAnswer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer;\n window.RTCPeerConnection.prototype.createAnswer = function createAnswer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateAnswer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateAnswer.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n'use strict';\nimport * as utils from '../utils';\n\nexport function shimLocalStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n return this._localStreams;\n };\n }\n if (!('addStream' in window.RTCPeerConnection.prototype)) {\n const _addTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n // Try to emulate Chrome's behaviour of adding in audio-video order.\n // Safari orders by track id.\n stream.getAudioTracks().forEach(track => _addTrack.call(this, track,\n stream));\n stream.getVideoTracks().forEach(track => _addTrack.call(this, track,\n stream));\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, ...streams) {\n if (streams) {\n streams.forEach((stream) => {\n if (!this._localStreams) {\n this._localStreams = [stream];\n } else if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n });\n }\n return _addTrack.apply(this, arguments);\n };\n }\n if (!('removeStream' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n const index = this._localStreams.indexOf(stream);\n if (index === -1) {\n return;\n }\n this._localStreams.splice(index, 1);\n const tracks = stream.getTracks();\n this.getSenders().forEach(sender => {\n if (tracks.includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n }\n}\n\nexport function shimRemoteStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getRemoteStreams =\n function getRemoteStreams() {\n return this._remoteStreams ? this._remoteStreams : [];\n };\n }\n if (!('onaddstream' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', {\n get() {\n return this._onaddstream;\n },\n set(f) {\n if (this._onaddstream) {\n this.removeEventListener('addstream', this._onaddstream);\n this.removeEventListener('track', this._onaddstreampoly);\n }\n this.addEventListener('addstream', this._onaddstream = f);\n this.addEventListener('track', this._onaddstreampoly = (e) => {\n e.streams.forEach(stream => {\n if (!this._remoteStreams) {\n this._remoteStreams = [];\n }\n if (this._remoteStreams.includes(stream)) {\n return;\n }\n this._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n this.dispatchEvent(event);\n });\n });\n }\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n const pc = this;\n if (!this._onaddstreampoly) {\n this.addEventListener('track', this._onaddstreampoly = function(e) {\n e.streams.forEach(stream => {\n if (!pc._remoteStreams) {\n pc._remoteStreams = [];\n }\n if (pc._remoteStreams.indexOf(stream) >= 0) {\n return;\n }\n pc._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n pc.dispatchEvent(event);\n });\n });\n }\n return origSetRemoteDescription.apply(pc, arguments);\n };\n }\n}\n\nexport function shimCallbacksAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n const prototype = window.RTCPeerConnection.prototype;\n const origCreateOffer = prototype.createOffer;\n const origCreateAnswer = prototype.createAnswer;\n const setLocalDescription = prototype.setLocalDescription;\n const setRemoteDescription = prototype.setRemoteDescription;\n const addIceCandidate = prototype.addIceCandidate;\n\n prototype.createOffer =\n function createOffer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateOffer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n prototype.createAnswer =\n function createAnswer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateAnswer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n let withCallback = function(description, successCallback, failureCallback) {\n const promise = setLocalDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setLocalDescription = withCallback;\n\n withCallback = function(description, successCallback, failureCallback) {\n const promise = setRemoteDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setRemoteDescription = withCallback;\n\n withCallback = function(candidate, successCallback, failureCallback) {\n const promise = addIceCandidate.apply(this, [candidate]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.addIceCandidate = withCallback;\n}\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {\n // shim not needed in Safari 12.1\n const mediaDevices = navigator.mediaDevices;\n const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices);\n navigator.mediaDevices.getUserMedia = (constraints) => {\n return _getUserMedia(shimConstraints(constraints));\n };\n }\n\n if (!navigator.getUserMedia && navigator.mediaDevices &&\n navigator.mediaDevices.getUserMedia) {\n navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) {\n navigator.mediaDevices.getUserMedia(constraints)\n .then(cb, errcb);\n }.bind(navigator);\n }\n}\n\nexport function shimConstraints(constraints) {\n if (constraints && constraints.video !== undefined) {\n return Object.assign({},\n constraints,\n {video: utils.compactObject(constraints.video)}\n );\n }\n\n return constraints;\n}\n\nexport function shimRTCIceServerUrls(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // migrate from non-spec RTCIceServer.url to RTCIceServer.urls\n const OrigPeerConnection = window.RTCPeerConnection;\n window.RTCPeerConnection =\n function RTCPeerConnection(pcConfig, pcConstraints) {\n if (pcConfig && pcConfig.iceServers) {\n const newIceServers = [];\n for (let i = 0; i < pcConfig.iceServers.length; i++) {\n let server = pcConfig.iceServers[i];\n if (!server.hasOwnProperty('urls') &&\n server.hasOwnProperty('url')) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n server = JSON.parse(JSON.stringify(server));\n server.urls = server.url;\n delete server.url;\n newIceServers.push(server);\n } else {\n newIceServers.push(pcConfig.iceServers[i]);\n }\n }\n pcConfig.iceServers = newIceServers;\n }\n return new OrigPeerConnection(pcConfig, pcConstraints);\n };\n window.RTCPeerConnection.prototype = OrigPeerConnection.prototype;\n // wrap static methods. Currently just generateCertificate.\n if ('generateCertificate' in OrigPeerConnection) {\n Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', {\n get() {\n return OrigPeerConnection.generateCertificate;\n }\n });\n }\n}\n\nexport function shimTrackEventTransceiver(window) {\n // Add event.transceiver member over deprecated event.receiver\n if (typeof window === 'object' && window.RTCTrackEvent &&\n 'receiver' in window.RTCTrackEvent.prototype &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimCreateOfferLegacy(window) {\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer =\n function createOffer(offerOptions) {\n if (offerOptions) {\n if (typeof offerOptions.offerToReceiveAudio !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveAudio =\n !!offerOptions.offerToReceiveAudio;\n }\n const audioTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'audio');\n if (offerOptions.offerToReceiveAudio === false && audioTransceiver) {\n if (audioTransceiver.direction === 'sendrecv') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('sendonly');\n } else {\n audioTransceiver.direction = 'sendonly';\n }\n } else if (audioTransceiver.direction === 'recvonly') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('inactive');\n } else {\n audioTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveAudio === true &&\n !audioTransceiver) {\n this.addTransceiver('audio');\n }\n\n if (typeof offerOptions.offerToReceiveVideo !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveVideo =\n !!offerOptions.offerToReceiveVideo;\n }\n const videoTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'video');\n if (offerOptions.offerToReceiveVideo === false && videoTransceiver) {\n if (videoTransceiver.direction === 'sendrecv') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('sendonly');\n } else {\n videoTransceiver.direction = 'sendonly';\n }\n } else if (videoTransceiver.direction === 'recvonly') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('inactive');\n } else {\n videoTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveVideo === true &&\n !videoTransceiver) {\n this.addTransceiver('video');\n }\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimAudioContext(window) {\n if (typeof window !== 'object' || window.AudioContext) {\n return;\n }\n window.AudioContext = window.webkitAudioContext;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport SDPUtils from 'sdp';\nimport * as utils from './utils';\n\nexport function shimRTCIceCandidate(window) {\n // foundation is arbitrarily chosen as an indicator for full support for\n // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface\n if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in\n window.RTCIceCandidate.prototype)) {\n return;\n }\n\n const NativeRTCIceCandidate = window.RTCIceCandidate;\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n // Remove the a= which shouldn't be part of the candidate string.\n if (typeof args === 'object' && args.candidate &&\n args.candidate.indexOf('a=') === 0) {\n args = JSON.parse(JSON.stringify(args));\n args.candidate = args.candidate.substr(2);\n }\n\n if (args.candidate && args.candidate.length) {\n // Augment the native candidate with the parsed fields.\n const nativeCandidate = new NativeRTCIceCandidate(args);\n const parsedCandidate = SDPUtils.parseCandidate(args.candidate);\n const augmentedCandidate = Object.assign(nativeCandidate,\n parsedCandidate);\n\n // Add a serializer that does not serialize the extra attributes.\n augmentedCandidate.toJSON = function toJSON() {\n return {\n candidate: augmentedCandidate.candidate,\n sdpMid: augmentedCandidate.sdpMid,\n sdpMLineIndex: augmentedCandidate.sdpMLineIndex,\n usernameFragment: augmentedCandidate.usernameFragment,\n };\n };\n return augmentedCandidate;\n }\n return new NativeRTCIceCandidate(args);\n };\n window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype;\n\n // Hook up the augmented candidate in onicecandidate and\n // addEventListener('icecandidate', ...)\n utils.wrapPeerConnectionEvent(window, 'icecandidate', e => {\n if (e.candidate) {\n Object.defineProperty(e, 'candidate', {\n value: new window.RTCIceCandidate(e.candidate),\n writable: 'false'\n });\n }\n return e;\n });\n}\n\nexport function shimMaxMessageSize(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n if (!('sctp' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n }\n });\n }\n\n const sctpInDescription = function(description) {\n if (!description || !description.sdp) {\n return false;\n }\n const sections = SDPUtils.splitSections(description.sdp);\n sections.shift();\n return sections.some(mediaSection => {\n const mLine = SDPUtils.parseMLine(mediaSection);\n return mLine && mLine.kind === 'application'\n && mLine.protocol.indexOf('SCTP') !== -1;\n });\n };\n\n const getRemoteFirefoxVersion = function(description) {\n // TODO: Is there a better solution for detecting Firefox?\n const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\\d+)/);\n if (match === null || match.length < 2) {\n return -1;\n }\n const version = parseInt(match[1], 10);\n // Test for NaN (yes, this is ugly)\n return version !== version ? -1 : version;\n };\n\n const getCanSendMaxMessageSize = function(remoteIsFirefox) {\n // Every implementation we know can send at least 64 KiB.\n // Note: Although Chrome is technically able to send up to 256 KiB, the\n // data does not reach the other peer reliably.\n // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419\n let canSendMaxMessageSize = 65536;\n if (browserDetails.browser === 'firefox') {\n if (browserDetails.version < 57) {\n if (remoteIsFirefox === -1) {\n // FF < 57 will send in 16 KiB chunks using the deprecated PPID\n // fragmentation.\n canSendMaxMessageSize = 16384;\n } else {\n // However, other FF (and RAWRTC) can reassemble PPID-fragmented\n // messages. Thus, supporting ~2 GiB when sending.\n canSendMaxMessageSize = 2147483637;\n }\n } else if (browserDetails.version < 60) {\n // Currently, all FF >= 57 will reset the remote maximum message size\n // to the default value when a data channel is created at a later\n // stage. :(\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n canSendMaxMessageSize =\n browserDetails.version === 57 ? 65535 : 65536;\n } else {\n // FF >= 60 supports sending ~2 GiB\n canSendMaxMessageSize = 2147483637;\n }\n }\n return canSendMaxMessageSize;\n };\n\n const getMaxMessageSize = function(description, remoteIsFirefox) {\n // Note: 65536 bytes is the default value from the SDP spec. Also,\n // every implementation we know supports receiving 65536 bytes.\n let maxMessageSize = 65536;\n\n // FF 57 has a slightly incorrect default remote max message size, so\n // we need to adjust it here to avoid a failure when sending.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697\n if (browserDetails.browser === 'firefox'\n && browserDetails.version === 57) {\n maxMessageSize = 65535;\n }\n\n const match = SDPUtils.matchPrefix(description.sdp,\n 'a=max-message-size:');\n if (match.length > 0) {\n maxMessageSize = parseInt(match[0].substr(19), 10);\n } else if (browserDetails.browser === 'firefox' &&\n remoteIsFirefox !== -1) {\n // If the maximum message size is not present in the remote SDP and\n // both local and remote are Firefox, the remote peer can receive\n // ~2 GiB.\n maxMessageSize = 2147483637;\n }\n return maxMessageSize;\n };\n\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n this._sctp = null;\n // Chrome decided to not expose .sctp in plan-b mode.\n // As usual, adapter.js has to do an 'ugly worakaround'\n // to cover up the mess.\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) {\n const {sdpSemantics} = this.getConfiguration();\n if (sdpSemantics === 'plan-b') {\n Object.defineProperty(this, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n },\n enumerable: true,\n configurable: true,\n });\n }\n }\n\n if (sctpInDescription(arguments[0])) {\n // Check if the remote is FF.\n const isFirefox = getRemoteFirefoxVersion(arguments[0]);\n\n // Get the maximum message size the local peer is capable of sending\n const canSendMMS = getCanSendMaxMessageSize(isFirefox);\n\n // Get the maximum message size of the remote peer.\n const remoteMMS = getMaxMessageSize(arguments[0], isFirefox);\n\n // Determine final maximum message size\n let maxMessageSize;\n if (canSendMMS === 0 && remoteMMS === 0) {\n maxMessageSize = Number.POSITIVE_INFINITY;\n } else if (canSendMMS === 0 || remoteMMS === 0) {\n maxMessageSize = Math.max(canSendMMS, remoteMMS);\n } else {\n maxMessageSize = Math.min(canSendMMS, remoteMMS);\n }\n\n // Create a dummy RTCSctpTransport object and the 'maxMessageSize'\n // attribute.\n const sctp = {};\n Object.defineProperty(sctp, 'maxMessageSize', {\n get() {\n return maxMessageSize;\n }\n });\n this._sctp = sctp;\n }\n\n return origSetRemoteDescription.apply(this, arguments);\n };\n}\n\nexport function shimSendThrowTypeError(window) {\n if (!(window.RTCPeerConnection &&\n 'createDataChannel' in window.RTCPeerConnection.prototype)) {\n return;\n }\n\n // Note: Although Firefox >= 57 has a native implementation, the maximum\n // message size can be reset for all data channels at a later stage.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n\n function wrapDcSend(dc, pc) {\n const origDataChannelSend = dc.send;\n dc.send = function send() {\n const data = arguments[0];\n const length = data.length || data.size || data.byteLength;\n if (dc.readyState === 'open' &&\n pc.sctp && length > pc.sctp.maxMessageSize) {\n throw new TypeError('Message too large (can send a maximum of ' +\n pc.sctp.maxMessageSize + ' bytes)');\n }\n return origDataChannelSend.apply(dc, arguments);\n };\n }\n const origCreateDataChannel =\n window.RTCPeerConnection.prototype.createDataChannel;\n window.RTCPeerConnection.prototype.createDataChannel =\n function createDataChannel() {\n const dataChannel = origCreateDataChannel.apply(this, arguments);\n wrapDcSend(dataChannel, this);\n return dataChannel;\n };\n utils.wrapPeerConnectionEvent(window, 'datachannel', e => {\n wrapDcSend(e.channel, e.target);\n return e;\n });\n}\n\n\n/* shims RTCConnectionState by pretending it is the same as iceConnectionState.\n * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12\n * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect\n * since DTLS failures would be hidden. See\n * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827\n * for the Firefox tracking bug.\n */\nexport function shimConnectionState(window) {\n if (!window.RTCPeerConnection ||\n 'connectionState' in window.RTCPeerConnection.prototype) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n Object.defineProperty(proto, 'connectionState', {\n get() {\n return {\n completed: 'connected',\n checking: 'connecting'\n }[this.iceConnectionState] || this.iceConnectionState;\n },\n enumerable: true,\n configurable: true\n });\n Object.defineProperty(proto, 'onconnectionstatechange', {\n get() {\n return this._onconnectionstatechange || null;\n },\n set(cb) {\n if (this._onconnectionstatechange) {\n this.removeEventListener('connectionstatechange',\n this._onconnectionstatechange);\n delete this._onconnectionstatechange;\n }\n if (cb) {\n this.addEventListener('connectionstatechange',\n this._onconnectionstatechange = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n\n ['setLocalDescription', 'setRemoteDescription'].forEach((method) => {\n const origMethod = proto[method];\n proto[method] = function() {\n if (!this._connectionstatechangepoly) {\n this._connectionstatechangepoly = e => {\n const pc = e.target;\n if (pc._lastConnectionState !== pc.connectionState) {\n pc._lastConnectionState = pc.connectionState;\n const newEvent = new Event('connectionstatechange', e);\n pc.dispatchEvent(newEvent);\n }\n return e;\n };\n this.addEventListener('iceconnectionstatechange',\n this._connectionstatechangepoly);\n }\n return origMethod.apply(this, arguments);\n };\n });\n}\n\nexport function removeExtmapAllowMixed(window, browserDetails) {\n /* remove a=extmap-allow-mixed for webrtc.org < M71 */\n if (!window.RTCPeerConnection) {\n return;\n }\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) {\n return;\n }\n if (browserDetails.browser === 'safari' && browserDetails.version >= 605) {\n return;\n }\n const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription(desc) {\n if (desc && desc.sdp && desc.sdp.indexOf('\\na=extmap-allow-mixed') !== -1) {\n const sdp = desc.sdp.split('\\n').filter((line) => {\n return line.trim() !== 'a=extmap-allow-mixed';\n }).join('\\n');\n // Safari enforces read-only-ness of RTCSessionDescription fields.\n if (window.RTCSessionDescription &&\n desc instanceof window.RTCSessionDescription) {\n arguments[0] = new window.RTCSessionDescription({\n type: desc.type,\n sdp,\n });\n } else {\n desc.sdp = sdp;\n }\n }\n return nativeSRD.apply(this, arguments);\n };\n}\n\nexport function shimAddIceCandidateNullOrEmpty(window, browserDetails) {\n // Support for addIceCandidate(null or undefined)\n // as well as addIceCandidate({candidate: \"\", ...})\n // https://bugs.chromium.org/p/chromium/issues/detail?id=978582\n // Note: must be called before other polyfills which change the signature.\n if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) {\n return;\n }\n const nativeAddIceCandidate =\n window.RTCPeerConnection.prototype.addIceCandidate;\n if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) {\n return;\n }\n window.RTCPeerConnection.prototype.addIceCandidate =\n function addIceCandidate() {\n if (!arguments[0]) {\n if (arguments[1]) {\n arguments[1].apply(null);\n }\n return Promise.resolve();\n }\n // Firefox 68+ emits and processes {candidate: \"\", ...}, ignore\n // in older versions.\n // Native support for ignoring exists for Chrome M77+.\n // Safari ignores as well, exact version unknown but works in the same\n // version that also ignores addIceCandidate(null).\n if (((browserDetails.browser === 'chrome' && browserDetails.version < 78)\n || (browserDetails.browser === 'firefox'\n && browserDetails.version < 68)\n || (browserDetails.browser === 'safari'))\n && arguments[0] && arguments[0].candidate === '') {\n return Promise.resolve();\n }\n return nativeAddIceCandidate.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\nimport * as utils from './utils';\n\n // Browser shims.\nimport * as chromeShim from './chrome/chrome_shim';\nimport * as edgeShim from './edge/edge_shim';\nimport * as firefoxShim from './firefox/firefox_shim';\nimport * as safariShim from './safari/safari_shim';\nimport * as commonShim from './common_shim';\n\n// Shimming starts here.\nexport function adapterFactory({window} = {}, options = {\n shimChrome: true,\n shimFirefox: true,\n shimEdge: true,\n shimSafari: true,\n}) {\n // Utils.\n const logging = utils.log;\n const browserDetails = utils.detectBrowser(window);\n\n const adapter = {\n browserDetails,\n commonShim,\n extractVersion: utils.extractVersion,\n disableLog: utils.disableLog,\n disableWarnings: utils.disableWarnings\n };\n\n // Shim browser if found.\n switch (browserDetails.browser) {\n case 'chrome':\n if (!chromeShim || !chromeShim.shimPeerConnection ||\n !options.shimChrome) {\n logging('Chrome shim is not included in this adapter release.');\n return adapter;\n }\n if (browserDetails.version === null) {\n logging('Chrome shim can not determine version, not shimming.');\n return adapter;\n }\n logging('adapter.js shimming chrome.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = chromeShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n chromeShim.shimGetUserMedia(window, browserDetails);\n chromeShim.shimMediaStream(window, browserDetails);\n chromeShim.shimPeerConnection(window, browserDetails);\n chromeShim.shimOnTrack(window, browserDetails);\n chromeShim.shimAddTrackRemoveTrack(window, browserDetails);\n chromeShim.shimGetSendersWithDtmf(window, browserDetails);\n chromeShim.shimGetStats(window, browserDetails);\n chromeShim.shimSenderReceiverGetStats(window, browserDetails);\n chromeShim.fixNegotiationNeeded(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n case 'firefox':\n if (!firefoxShim || !firefoxShim.shimPeerConnection ||\n !options.shimFirefox) {\n logging('Firefox shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming firefox.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = firefoxShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n firefoxShim.shimGetUserMedia(window, browserDetails);\n firefoxShim.shimPeerConnection(window, browserDetails);\n firefoxShim.shimOnTrack(window, browserDetails);\n firefoxShim.shimRemoveStream(window, browserDetails);\n firefoxShim.shimSenderGetStats(window, browserDetails);\n firefoxShim.shimReceiverGetStats(window, browserDetails);\n firefoxShim.shimRTCDataChannel(window, browserDetails);\n firefoxShim.shimAddTransceiver(window, browserDetails);\n firefoxShim.shimGetParameters(window, browserDetails);\n firefoxShim.shimCreateOffer(window, browserDetails);\n firefoxShim.shimCreateAnswer(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'edge':\n if (!edgeShim || !edgeShim.shimPeerConnection || !options.shimEdge) {\n logging('MS edge shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming edge.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = edgeShim;\n\n edgeShim.shimGetUserMedia(window, browserDetails);\n edgeShim.shimGetDisplayMedia(window, browserDetails);\n edgeShim.shimPeerConnection(window, browserDetails);\n edgeShim.shimReplaceTrack(window, browserDetails);\n\n // the edge shim implements the full RTCIceCandidate object.\n\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'safari':\n if (!safariShim || !options.shimSafari) {\n logging('Safari shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming safari.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = safariShim;\n\n // Must be called before shimCallbackAPI.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n safariShim.shimRTCIceServerUrls(window, browserDetails);\n safariShim.shimCreateOfferLegacy(window, browserDetails);\n safariShim.shimCallbacksAPI(window, browserDetails);\n safariShim.shimLocalStreamsAPI(window, browserDetails);\n safariShim.shimRemoteStreamsAPI(window, browserDetails);\n safariShim.shimTrackEventTransceiver(window, browserDetails);\n safariShim.shimGetUserMedia(window, browserDetails);\n safariShim.shimAudioContext(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n default:\n logging('Unsupported browser!');\n break;\n }\n\n return adapter;\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n\n'use strict';\n\nimport {adapterFactory} from './adapter_factory.js';\n\nconst adapter =\n adapterFactory({window: typeof window === 'undefined' ? undefined : window});\nexport default adapter;\n","\n'use strict';\nimport * as utils from './utils.js';\nimport * as MediaFormatModule from './mediaformat.js';\nimport adapter from 'webrtc-adapter';\n\n/**\n * @class AudioTrackConstraints\n * @classDesc Constraints for creating an audio MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track.\n */\nexport class AudioTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.AudioSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Values could be \"mic\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Do not provide deviceId if source is not \"mic\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n this.deviceId = undefined;\n }\n}\n\n/**\n * @class VideoTrackConstraints\n * @classDesc Constraints for creating a video MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.VideoSourceInfo} source Source info of this video track.\n */\nexport class VideoTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.VideoSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Values could be \"camera\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Do not provide deviceId if source is not \"camera\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n\n this.deviceId = undefined;\n\n /**\n * @member {Owt.Base.Resolution} resolution\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.resolution = undefined;\n\n /**\n * @member {number} frameRate\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.frameRate = undefined;\n }\n}\n/**\n * @class StreamConstraints\n * @classDesc Constraints for creating a MediaStream from screen mic and camera.\n * @memberof Owt.Base\n * @constructor\n * @param {?Owt.Base.AudioTrackConstraints} audioConstraints\n * @param {?Owt.Base.VideoTrackConstraints} videoConstraints\n */\nexport class StreamConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(audioConstraints = false, videoConstraints = false) {\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.audio = audioConstraints;\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.video = videoConstraints;\n }\n}\n\n// eslint-disable-next-line require-jsdoc\nfunction isVideoConstrainsForScreenCast(constraints) {\n return (typeof constraints.video === 'object' && constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST);\n}\n\n/**\n * @class MediaStreamFactory\n * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself.\n * @memberof Owt.Base\n */\nexport class MediaStreamFactory {\n /**\n * @function createMediaStream\n * @static\n * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are \"screen-cast\".\n * @memberof Owt.Base.MediaStreamFactory\n * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened:\n * - One or more parameters cannot be satisfied.\n * - Specified device is busy.\n * - Cannot obtain necessary permission or operation is canceled by user.\n * - Video source is screen cast, while audio source is not.\n * - Audio source is screen cast, while video source is disabled.\n * @param {Owt.Base.StreamConstraints} constraints\n */\n static createMediaStream(constraints) {\n if (typeof constraints !== 'object' ||\n (!constraints.audio && !constraints.video)) {\n return Promise.reject(new TypeError('Invalid constrains'));\n }\n if (!isVideoConstrainsForScreenCast(constraints) &&\n (typeof constraints.audio === 'object') &&\n constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(\n new TypeError('Cannot share screen without video.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) && !utils.isChrome() &&\n !utils.isFirefox()) {\n return Promise.reject(\n new TypeError('Screen sharing only supports Chrome and Firefox.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) &&\n typeof constraints.audio === 'object' &&\n constraints.audio.source !==\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(new TypeError(\n 'Cannot capture video from screen cast while capture audio from'\n + ' other source.'));\n }\n\n // Check and convert constraints.\n if (!constraints.audio && !constraints.video) {\n return Promise.reject(new TypeError(\n 'At least one of audio and video must be requested.'));\n }\n const mediaConstraints = Object.create({});\n if (typeof constraints.audio === 'object' &&\n constraints.audio.source === MediaFormatModule.AudioSourceInfo.MIC) {\n mediaConstraints.audio = Object.create({});\n if (utils.isEdge()) {\n mediaConstraints.audio.deviceId = constraints.audio.deviceId;\n } else {\n mediaConstraints.audio.deviceId = {\n exact: constraints.audio.deviceId,\n };\n }\n } else {\n if (constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n mediaConstraints.audio = true;\n } else {\n mediaConstraints.audio = constraints.audio;\n }\n }\n if (typeof constraints.video === 'object') {\n mediaConstraints.video = Object.create({});\n if (typeof constraints.video.frameRate === 'number') {\n mediaConstraints.video.frameRate = constraints.video.frameRate;\n }\n if (constraints.video.resolution &&\n constraints.video.resolution.width &&\n constraints.video.resolution.height) {\n if (constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.width = constraints.video.resolution.width;\n mediaConstraints.video.height = constraints.video.resolution.height;\n } else {\n mediaConstraints.video.width = Object.create({});\n mediaConstraints.video.width.exact =\n constraints.video.resolution.width;\n mediaConstraints.video.height = Object.create({});\n mediaConstraints.video.height.exact =\n constraints.video.resolution.height;\n }\n }\n if (typeof constraints.video.deviceId === 'string') {\n mediaConstraints.video.deviceId = {exact: constraints.video.deviceId};\n }\n if (utils.isFirefox() &&\n constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.mediaSource = 'screen';\n }\n } else {\n mediaConstraints.video = constraints.video;\n }\n\n if (isVideoConstrainsForScreenCast(constraints)) {\n return navigator.mediaDevices.getDisplayMedia(mediaConstraints);\n } else {\n return navigator.mediaDevices.getUserMedia(mediaConstraints);\n }\n }\n}\n","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n\nexport * from './mediastream-factory.js';\nexport * from './mediaformat.js';","let logger;\nlet errorLogger;\n\nexport function setLogger() {\n /*eslint-disable */\n logger = console.log;\n errorLogger = console.error;\n /*eslint-enable */\n}\n\nexport function isEnable() {\n return logger != null;\n}\n\nexport function log(message, ...optionalParams) {\n if (logger) {\n logger(message, ...optionalParams);\n }\n}\nexport function error(message, ...optionalParams) {\n if (errorLogger) {\n errorLogger(message, ...optionalParams);\n }\n}\n","export default class Event {\n constructor(type) {\n this.listener = {};\n this.type = type | '';\n }\n\n on(event, fn) {\n if (!this.listener[event]) {\n this.listener[event] = [];\n }\n this.listener[event].push(fn);\n return true;\n }\n\n off(event, fn) {\n if (this.listener[event]) {\n var index = this.listener[event].indexOf(fn);\n if (index > -1) {\n this.listener[event].splice(index, 1);\n }\n return true;\n }\n return false;\n }\n\n offAll() {\n this.listener = {};\n }\n\n dispatch(event, data) {\n if (this.listener[event]) {\n this.listener[event].map((each) => {\n each.apply(null, [data]);\n });\n return true;\n }\n return false;\n }\n}\n","'use strict';\n\nmodule.exports = function bind(fn, thisArg) {\n return function wrap() {\n var args = new Array(arguments.length);\n for (var i = 0; i < args.length; i++) {\n args[i] = arguments[i];\n }\n return fn.apply(thisArg, args);\n };\n};\n","'use strict';\n\nvar bind = require('./helpers/bind');\n\n/*global toString:true*/\n\n// utils is a library of generic helper functions non-specific to axios\n\nvar toString = Object.prototype.toString;\n\n/**\n * Determine if a value is an Array\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Array, otherwise false\n */\nfunction isArray(val) {\n return toString.call(val) === '[object Array]';\n}\n\n/**\n * Determine if a value is undefined\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if the value is undefined, otherwise false\n */\nfunction isUndefined(val) {\n return typeof val === 'undefined';\n}\n\n/**\n * Determine if a value is a Buffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Buffer, otherwise false\n */\nfunction isBuffer(val) {\n return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor)\n && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val);\n}\n\n/**\n * Determine if a value is an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an ArrayBuffer, otherwise false\n */\nfunction isArrayBuffer(val) {\n return toString.call(val) === '[object ArrayBuffer]';\n}\n\n/**\n * Determine if a value is a FormData\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an FormData, otherwise false\n */\nfunction isFormData(val) {\n return (typeof FormData !== 'undefined') && (val instanceof FormData);\n}\n\n/**\n * Determine if a value is a view on an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false\n */\nfunction isArrayBufferView(val) {\n var result;\n if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) {\n result = ArrayBuffer.isView(val);\n } else {\n result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer);\n }\n return result;\n}\n\n/**\n * Determine if a value is a String\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a String, otherwise false\n */\nfunction isString(val) {\n return typeof val === 'string';\n}\n\n/**\n * Determine if a value is a Number\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Number, otherwise false\n */\nfunction isNumber(val) {\n return typeof val === 'number';\n}\n\n/**\n * Determine if a value is an Object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Object, otherwise false\n */\nfunction isObject(val) {\n return val !== null && typeof val === 'object';\n}\n\n/**\n * Determine if a value is a plain Object\n *\n * @param {Object} val The value to test\n * @return {boolean} True if value is a plain Object, otherwise false\n */\nfunction isPlainObject(val) {\n if (toString.call(val) !== '[object Object]') {\n return false;\n }\n\n var prototype = Object.getPrototypeOf(val);\n return prototype === null || prototype === Object.prototype;\n}\n\n/**\n * Determine if a value is a Date\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Date, otherwise false\n */\nfunction isDate(val) {\n return toString.call(val) === '[object Date]';\n}\n\n/**\n * Determine if a value is a File\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a File, otherwise false\n */\nfunction isFile(val) {\n return toString.call(val) === '[object File]';\n}\n\n/**\n * Determine if a value is a Blob\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Blob, otherwise false\n */\nfunction isBlob(val) {\n return toString.call(val) === '[object Blob]';\n}\n\n/**\n * Determine if a value is a Function\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Function, otherwise false\n */\nfunction isFunction(val) {\n return toString.call(val) === '[object Function]';\n}\n\n/**\n * Determine if a value is a Stream\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Stream, otherwise false\n */\nfunction isStream(val) {\n return isObject(val) && isFunction(val.pipe);\n}\n\n/**\n * Determine if a value is a URLSearchParams object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a URLSearchParams object, otherwise false\n */\nfunction isURLSearchParams(val) {\n return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams;\n}\n\n/**\n * Trim excess whitespace off the beginning and end of a string\n *\n * @param {String} str The String to trim\n * @returns {String} The String freed of excess whitespace\n */\nfunction trim(str) {\n return str.replace(/^\\s*/, '').replace(/\\s*$/, '');\n}\n\n/**\n * Determine if we're running in a standard browser environment\n *\n * This allows axios to run in a web worker, and react-native.\n * Both environments support XMLHttpRequest, but not fully standard globals.\n *\n * web workers:\n * typeof window -> undefined\n * typeof document -> undefined\n *\n * react-native:\n * navigator.product -> 'ReactNative'\n * nativescript\n * navigator.product -> 'NativeScript' or 'NS'\n */\nfunction isStandardBrowserEnv() {\n if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' ||\n navigator.product === 'NativeScript' ||\n navigator.product === 'NS')) {\n return false;\n }\n return (\n typeof window !== 'undefined' &&\n typeof document !== 'undefined'\n );\n}\n\n/**\n * Iterate over an Array or an Object invoking a function for each item.\n *\n * If `obj` is an Array callback will be called passing\n * the value, index, and complete array for each item.\n *\n * If 'obj' is an Object callback will be called passing\n * the value, key, and complete object for each property.\n *\n * @param {Object|Array} obj The object to iterate\n * @param {Function} fn The callback to invoke for each item\n */\nfunction forEach(obj, fn) {\n // Don't bother if no value provided\n if (obj === null || typeof obj === 'undefined') {\n return;\n }\n\n // Force an array if not already something iterable\n if (typeof obj !== 'object') {\n /*eslint no-param-reassign:0*/\n obj = [obj];\n }\n\n if (isArray(obj)) {\n // Iterate over array values\n for (var i = 0, l = obj.length; i < l; i++) {\n fn.call(null, obj[i], i, obj);\n }\n } else {\n // Iterate over object keys\n for (var key in obj) {\n if (Object.prototype.hasOwnProperty.call(obj, key)) {\n fn.call(null, obj[key], key, obj);\n }\n }\n }\n}\n\n/**\n * Accepts varargs expecting each argument to be an object, then\n * immutably merges the properties of each object and returns result.\n *\n * When multiple objects contain the same key the later object in\n * the arguments list will take precedence.\n *\n * Example:\n *\n * ```js\n * var result = merge({foo: 123}, {foo: 456});\n * console.log(result.foo); // outputs 456\n * ```\n *\n * @param {Object} obj1 Object to merge\n * @returns {Object} Result of all merge properties\n */\nfunction merge(/* obj1, obj2, obj3, ... */) {\n var result = {};\n function assignValue(val, key) {\n if (isPlainObject(result[key]) && isPlainObject(val)) {\n result[key] = merge(result[key], val);\n } else if (isPlainObject(val)) {\n result[key] = merge({}, val);\n } else if (isArray(val)) {\n result[key] = val.slice();\n } else {\n result[key] = val;\n }\n }\n\n for (var i = 0, l = arguments.length; i < l; i++) {\n forEach(arguments[i], assignValue);\n }\n return result;\n}\n\n/**\n * Extends object a by mutably adding to it the properties of object b.\n *\n * @param {Object} a The object to be extended\n * @param {Object} b The object to copy properties from\n * @param {Object} thisArg The object to bind function to\n * @return {Object} The resulting value of object a\n */\nfunction extend(a, b, thisArg) {\n forEach(b, function assignValue(val, key) {\n if (thisArg && typeof val === 'function') {\n a[key] = bind(val, thisArg);\n } else {\n a[key] = val;\n }\n });\n return a;\n}\n\n/**\n * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM)\n *\n * @param {string} content with BOM\n * @return {string} content value without BOM\n */\nfunction stripBOM(content) {\n if (content.charCodeAt(0) === 0xFEFF) {\n content = content.slice(1);\n }\n return content;\n}\n\nmodule.exports = {\n isArray: isArray,\n isArrayBuffer: isArrayBuffer,\n isBuffer: isBuffer,\n isFormData: isFormData,\n isArrayBufferView: isArrayBufferView,\n isString: isString,\n isNumber: isNumber,\n isObject: isObject,\n isPlainObject: isPlainObject,\n isUndefined: isUndefined,\n isDate: isDate,\n isFile: isFile,\n isBlob: isBlob,\n isFunction: isFunction,\n isStream: isStream,\n isURLSearchParams: isURLSearchParams,\n isStandardBrowserEnv: isStandardBrowserEnv,\n forEach: forEach,\n merge: merge,\n extend: extend,\n trim: trim,\n stripBOM: stripBOM\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction encode(val) {\n return encodeURIComponent(val).\n replace(/%3A/gi, ':').\n replace(/%24/g, '$').\n replace(/%2C/gi, ',').\n replace(/%20/g, '+').\n replace(/%5B/gi, '[').\n replace(/%5D/gi, ']');\n}\n\n/**\n * Build a URL by appending params to the end\n *\n * @param {string} url The base of the url (e.g., http://www.google.com)\n * @param {object} [params] The params to be appended\n * @returns {string} The formatted url\n */\nmodule.exports = function buildURL(url, params, paramsSerializer) {\n /*eslint no-param-reassign:0*/\n if (!params) {\n return url;\n }\n\n var serializedParams;\n if (paramsSerializer) {\n serializedParams = paramsSerializer(params);\n } else if (utils.isURLSearchParams(params)) {\n serializedParams = params.toString();\n } else {\n var parts = [];\n\n utils.forEach(params, function serialize(val, key) {\n if (val === null || typeof val === 'undefined') {\n return;\n }\n\n if (utils.isArray(val)) {\n key = key + '[]';\n } else {\n val = [val];\n }\n\n utils.forEach(val, function parseValue(v) {\n if (utils.isDate(v)) {\n v = v.toISOString();\n } else if (utils.isObject(v)) {\n v = JSON.stringify(v);\n }\n parts.push(encode(key) + '=' + encode(v));\n });\n });\n\n serializedParams = parts.join('&');\n }\n\n if (serializedParams) {\n var hashmarkIndex = url.indexOf('#');\n if (hashmarkIndex !== -1) {\n url = url.slice(0, hashmarkIndex);\n }\n\n url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams;\n }\n\n return url;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction InterceptorManager() {\n this.handlers = [];\n}\n\n/**\n * Add a new interceptor to the stack\n *\n * @param {Function} fulfilled The function to handle `then` for a `Promise`\n * @param {Function} rejected The function to handle `reject` for a `Promise`\n *\n * @return {Number} An ID used to remove interceptor later\n */\nInterceptorManager.prototype.use = function use(fulfilled, rejected) {\n this.handlers.push({\n fulfilled: fulfilled,\n rejected: rejected\n });\n return this.handlers.length - 1;\n};\n\n/**\n * Remove an interceptor from the stack\n *\n * @param {Number} id The ID that was returned by `use`\n */\nInterceptorManager.prototype.eject = function eject(id) {\n if (this.handlers[id]) {\n this.handlers[id] = null;\n }\n};\n\n/**\n * Iterate over all the registered interceptors\n *\n * This method is particularly useful for skipping over any\n * interceptors that may have become `null` calling `eject`.\n *\n * @param {Function} fn The function to call for each interceptor\n */\nInterceptorManager.prototype.forEach = function forEach(fn) {\n utils.forEach(this.handlers, function forEachHandler(h) {\n if (h !== null) {\n fn(h);\n }\n });\n};\n\nmodule.exports = InterceptorManager;\n","'use strict';\n\nvar utils = require('./../utils');\n\n/**\n * Transform the data for a request or a response\n *\n * @param {Object|String} data The data to be transformed\n * @param {Array} headers The headers for the request or response\n * @param {Array|Function} fns A single function or Array of functions\n * @returns {*} The resulting transformed data\n */\nmodule.exports = function transformData(data, headers, fns) {\n /*eslint no-param-reassign:0*/\n utils.forEach(fns, function transform(fn) {\n data = fn(data, headers);\n });\n\n return data;\n};\n","'use strict';\n\nmodule.exports = function isCancel(value) {\n return !!(value && value.__CANCEL__);\n};\n","'use strict';\n\nvar utils = require('../utils');\n\nmodule.exports = function normalizeHeaderName(headers, normalizedName) {\n utils.forEach(headers, function processHeader(value, name) {\n if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) {\n headers[normalizedName] = value;\n delete headers[name];\n }\n });\n};\n","'use strict';\n\n/**\n * Update an Error with the specified config, error code, and response.\n *\n * @param {Error} error The error to update.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The error.\n */\nmodule.exports = function enhanceError(error, config, code, request, response) {\n error.config = config;\n if (code) {\n error.code = code;\n }\n\n error.request = request;\n error.response = response;\n error.isAxiosError = true;\n\n error.toJSON = function toJSON() {\n return {\n // Standard\n message: this.message,\n name: this.name,\n // Microsoft\n description: this.description,\n number: this.number,\n // Mozilla\n fileName: this.fileName,\n lineNumber: this.lineNumber,\n columnNumber: this.columnNumber,\n stack: this.stack,\n // Axios\n config: this.config,\n code: this.code\n };\n };\n return error;\n};\n","'use strict';\n\nvar enhanceError = require('./enhanceError');\n\n/**\n * Create an Error with the specified message, config, error code, request and response.\n *\n * @param {string} message The error message.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The created error.\n */\nmodule.exports = function createError(message, config, code, request, response) {\n var error = new Error(message);\n return enhanceError(error, config, code, request, response);\n};\n","'use strict';\n\nvar createError = require('./createError');\n\n/**\n * Resolve or reject a Promise based on response status.\n *\n * @param {Function} resolve A function that resolves the promise.\n * @param {Function} reject A function that rejects the promise.\n * @param {object} response The response.\n */\nmodule.exports = function settle(resolve, reject, response) {\n var validateStatus = response.config.validateStatus;\n if (!response.status || !validateStatus || validateStatus(response.status)) {\n resolve(response);\n } else {\n reject(createError(\n 'Request failed with status code ' + response.status,\n response.config,\n null,\n response.request,\n response\n ));\n }\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs support document.cookie\n (function standardBrowserEnv() {\n return {\n write: function write(name, value, expires, path, domain, secure) {\n var cookie = [];\n cookie.push(name + '=' + encodeURIComponent(value));\n\n if (utils.isNumber(expires)) {\n cookie.push('expires=' + new Date(expires).toGMTString());\n }\n\n if (utils.isString(path)) {\n cookie.push('path=' + path);\n }\n\n if (utils.isString(domain)) {\n cookie.push('domain=' + domain);\n }\n\n if (secure === true) {\n cookie.push('secure');\n }\n\n document.cookie = cookie.join('; ');\n },\n\n read: function read(name) {\n var match = document.cookie.match(new RegExp('(^|;\\\\s*)(' + name + ')=([^;]*)'));\n return (match ? decodeURIComponent(match[3]) : null);\n },\n\n remove: function remove(name) {\n this.write(name, '', Date.now() - 86400000);\n }\n };\n })() :\n\n // Non standard browser env (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return {\n write: function write() {},\n read: function read() { return null; },\n remove: function remove() {}\n };\n })()\n);\n","'use strict';\n\n/**\n * Determines whether the specified URL is absolute\n *\n * @param {string} url The URL to test\n * @returns {boolean} True if the specified URL is absolute, otherwise false\n */\nmodule.exports = function isAbsoluteURL(url) {\n // A URL is considered absolute if it begins with \"://\" or \"//\" (protocol-relative URL).\n // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed\n // by any combination of letters, digits, plus, period, or hyphen.\n return /^([a-z][a-z\\d\\+\\-\\.]*:)?\\/\\//i.test(url);\n};\n","'use strict';\n\n/**\n * Creates a new URL by combining the specified URLs\n *\n * @param {string} baseURL The base URL\n * @param {string} relativeURL The relative URL\n * @returns {string} The combined URL\n */\nmodule.exports = function combineURLs(baseURL, relativeURL) {\n return relativeURL\n ? baseURL.replace(/\\/+$/, '') + '/' + relativeURL.replace(/^\\/+/, '')\n : baseURL;\n};\n","'use strict';\n\nvar isAbsoluteURL = require('../helpers/isAbsoluteURL');\nvar combineURLs = require('../helpers/combineURLs');\n\n/**\n * Creates a new URL by combining the baseURL with the requestedURL,\n * only when the requestedURL is not already an absolute URL.\n * If the requestURL is absolute, this function returns the requestedURL untouched.\n *\n * @param {string} baseURL The base URL\n * @param {string} requestedURL Absolute or relative URL to combine\n * @returns {string} The combined full path\n */\nmodule.exports = function buildFullPath(baseURL, requestedURL) {\n if (baseURL && !isAbsoluteURL(requestedURL)) {\n return combineURLs(baseURL, requestedURL);\n }\n return requestedURL;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\n// Headers whose duplicates are ignored by node\n// c.f. https://nodejs.org/api/http.html#http_message_headers\nvar ignoreDuplicateOf = [\n 'age', 'authorization', 'content-length', 'content-type', 'etag',\n 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since',\n 'last-modified', 'location', 'max-forwards', 'proxy-authorization',\n 'referer', 'retry-after', 'user-agent'\n];\n\n/**\n * Parse headers into an object\n *\n * ```\n * Date: Wed, 27 Aug 2014 08:58:49 GMT\n * Content-Type: application/json\n * Connection: keep-alive\n * Transfer-Encoding: chunked\n * ```\n *\n * @param {String} headers Headers needing to be parsed\n * @returns {Object} Headers parsed into an object\n */\nmodule.exports = function parseHeaders(headers) {\n var parsed = {};\n var key;\n var val;\n var i;\n\n if (!headers) { return parsed; }\n\n utils.forEach(headers.split('\\n'), function parser(line) {\n i = line.indexOf(':');\n key = utils.trim(line.substr(0, i)).toLowerCase();\n val = utils.trim(line.substr(i + 1));\n\n if (key) {\n if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) {\n return;\n }\n if (key === 'set-cookie') {\n parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]);\n } else {\n parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val;\n }\n }\n });\n\n return parsed;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs have full support of the APIs needed to test\n // whether the request URL is of the same origin as current location.\n (function standardBrowserEnv() {\n var msie = /(msie|trident)/i.test(navigator.userAgent);\n var urlParsingNode = document.createElement('a');\n var originURL;\n\n /**\n * Parse a URL to discover it's components\n *\n * @param {String} url The URL to be parsed\n * @returns {Object}\n */\n function resolveURL(url) {\n var href = url;\n\n if (msie) {\n // IE needs attribute set twice to normalize properties\n urlParsingNode.setAttribute('href', href);\n href = urlParsingNode.href;\n }\n\n urlParsingNode.setAttribute('href', href);\n\n // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils\n return {\n href: urlParsingNode.href,\n protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '',\n host: urlParsingNode.host,\n search: urlParsingNode.search ? urlParsingNode.search.replace(/^\\?/, '') : '',\n hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '',\n hostname: urlParsingNode.hostname,\n port: urlParsingNode.port,\n pathname: (urlParsingNode.pathname.charAt(0) === '/') ?\n urlParsingNode.pathname :\n '/' + urlParsingNode.pathname\n };\n }\n\n originURL = resolveURL(window.location.href);\n\n /**\n * Determine if a URL shares the same origin as the current location\n *\n * @param {String} requestURL The URL to test\n * @returns {boolean} True if URL shares the same origin, otherwise false\n */\n return function isURLSameOrigin(requestURL) {\n var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL;\n return (parsed.protocol === originURL.protocol &&\n parsed.host === originURL.host);\n };\n })() :\n\n // Non standard browser envs (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return function isURLSameOrigin() {\n return true;\n };\n })()\n);\n","'use strict';\n\nvar utils = require('./../utils');\nvar settle = require('./../core/settle');\nvar cookies = require('./../helpers/cookies');\nvar buildURL = require('./../helpers/buildURL');\nvar buildFullPath = require('../core/buildFullPath');\nvar parseHeaders = require('./../helpers/parseHeaders');\nvar isURLSameOrigin = require('./../helpers/isURLSameOrigin');\nvar createError = require('../core/createError');\n\nmodule.exports = function xhrAdapter(config) {\n return new Promise(function dispatchXhrRequest(resolve, reject) {\n var requestData = config.data;\n var requestHeaders = config.headers;\n\n if (utils.isFormData(requestData)) {\n delete requestHeaders['Content-Type']; // Let the browser set it\n }\n\n var request = new XMLHttpRequest();\n\n // HTTP basic authentication\n if (config.auth) {\n var username = config.auth.username || '';\n var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : '';\n requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password);\n }\n\n var fullPath = buildFullPath(config.baseURL, config.url);\n request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true);\n\n // Set the request timeout in MS\n request.timeout = config.timeout;\n\n // Listen for ready state\n request.onreadystatechange = function handleLoad() {\n if (!request || request.readyState !== 4) {\n return;\n }\n\n // The request errored out and we didn't get a response, this will be\n // handled by onerror instead\n // With one exception: request that using file: protocol, most browsers\n // will return status as 0 even though it's a successful request\n if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) {\n return;\n }\n\n // Prepare the response\n var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null;\n var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response;\n var response = {\n data: responseData,\n status: request.status,\n statusText: request.statusText,\n headers: responseHeaders,\n config: config,\n request: request\n };\n\n settle(resolve, reject, response);\n\n // Clean up request\n request = null;\n };\n\n // Handle browser request cancellation (as opposed to a manual cancellation)\n request.onabort = function handleAbort() {\n if (!request) {\n return;\n }\n\n reject(createError('Request aborted', config, 'ECONNABORTED', request));\n\n // Clean up request\n request = null;\n };\n\n // Handle low level network errors\n request.onerror = function handleError() {\n // Real errors are hidden from us by the browser\n // onerror should only fire if it's a network error\n reject(createError('Network Error', config, null, request));\n\n // Clean up request\n request = null;\n };\n\n // Handle timeout\n request.ontimeout = function handleTimeout() {\n var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded';\n if (config.timeoutErrorMessage) {\n timeoutErrorMessage = config.timeoutErrorMessage;\n }\n reject(createError(timeoutErrorMessage, config, 'ECONNABORTED',\n request));\n\n // Clean up request\n request = null;\n };\n\n // Add xsrf header\n // This is only done if running in a standard browser environment.\n // Specifically not if we're in a web worker, or react-native.\n if (utils.isStandardBrowserEnv()) {\n // Add xsrf header\n var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ?\n cookies.read(config.xsrfCookieName) :\n undefined;\n\n if (xsrfValue) {\n requestHeaders[config.xsrfHeaderName] = xsrfValue;\n }\n }\n\n // Add headers to the request\n if ('setRequestHeader' in request) {\n utils.forEach(requestHeaders, function setRequestHeader(val, key) {\n if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') {\n // Remove Content-Type if data is undefined\n delete requestHeaders[key];\n } else {\n // Otherwise add header to the request\n request.setRequestHeader(key, val);\n }\n });\n }\n\n // Add withCredentials to request if needed\n if (!utils.isUndefined(config.withCredentials)) {\n request.withCredentials = !!config.withCredentials;\n }\n\n // Add responseType to request if needed\n if (config.responseType) {\n try {\n request.responseType = config.responseType;\n } catch (e) {\n // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2.\n // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function.\n if (config.responseType !== 'json') {\n throw e;\n }\n }\n }\n\n // Handle progress if needed\n if (typeof config.onDownloadProgress === 'function') {\n request.addEventListener('progress', config.onDownloadProgress);\n }\n\n // Not all browsers support upload events\n if (typeof config.onUploadProgress === 'function' && request.upload) {\n request.upload.addEventListener('progress', config.onUploadProgress);\n }\n\n if (config.cancelToken) {\n // Handle cancellation\n config.cancelToken.promise.then(function onCanceled(cancel) {\n if (!request) {\n return;\n }\n\n request.abort();\n reject(cancel);\n // Clean up request\n request = null;\n });\n }\n\n if (!requestData) {\n requestData = null;\n }\n\n // Send the request\n request.send(requestData);\n });\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar normalizeHeaderName = require('./helpers/normalizeHeaderName');\n\nvar DEFAULT_CONTENT_TYPE = {\n 'Content-Type': 'application/x-www-form-urlencoded'\n};\n\nfunction setContentTypeIfUnset(headers, value) {\n if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) {\n headers['Content-Type'] = value;\n }\n}\n\nfunction getDefaultAdapter() {\n var adapter;\n if (typeof XMLHttpRequest !== 'undefined') {\n // For browsers use XHR adapter\n adapter = require('./adapters/xhr');\n } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') {\n // For node use HTTP adapter\n adapter = require('./adapters/http');\n }\n return adapter;\n}\n\nvar defaults = {\n adapter: getDefaultAdapter(),\n\n transformRequest: [function transformRequest(data, headers) {\n normalizeHeaderName(headers, 'Accept');\n normalizeHeaderName(headers, 'Content-Type');\n if (utils.isFormData(data) ||\n utils.isArrayBuffer(data) ||\n utils.isBuffer(data) ||\n utils.isStream(data) ||\n utils.isFile(data) ||\n utils.isBlob(data)\n ) {\n return data;\n }\n if (utils.isArrayBufferView(data)) {\n return data.buffer;\n }\n if (utils.isURLSearchParams(data)) {\n setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8');\n return data.toString();\n }\n if (utils.isObject(data)) {\n setContentTypeIfUnset(headers, 'application/json;charset=utf-8');\n return JSON.stringify(data);\n }\n return data;\n }],\n\n transformResponse: [function transformResponse(data) {\n /*eslint no-param-reassign:0*/\n if (typeof data === 'string') {\n try {\n data = JSON.parse(data);\n } catch (e) { /* Ignore */ }\n }\n return data;\n }],\n\n /**\n * A timeout in milliseconds to abort a request. If set to 0 (default) a\n * timeout is not created.\n */\n timeout: 0,\n\n xsrfCookieName: 'XSRF-TOKEN',\n xsrfHeaderName: 'X-XSRF-TOKEN',\n\n maxContentLength: -1,\n maxBodyLength: -1,\n\n validateStatus: function validateStatus(status) {\n return status >= 200 && status < 300;\n }\n};\n\ndefaults.headers = {\n common: {\n 'Accept': 'application/json, text/plain, */*'\n }\n};\n\nutils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) {\n defaults.headers[method] = {};\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE);\n});\n\nmodule.exports = defaults;\n","'use strict';\n\nvar utils = require('./../utils');\nvar transformData = require('./transformData');\nvar isCancel = require('../cancel/isCancel');\nvar defaults = require('../defaults');\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nfunction throwIfCancellationRequested(config) {\n if (config.cancelToken) {\n config.cancelToken.throwIfRequested();\n }\n}\n\n/**\n * Dispatch a request to the server using the configured adapter.\n *\n * @param {object} config The config that is to be used for the request\n * @returns {Promise} The Promise to be fulfilled\n */\nmodule.exports = function dispatchRequest(config) {\n throwIfCancellationRequested(config);\n\n // Ensure headers exist\n config.headers = config.headers || {};\n\n // Transform request data\n config.data = transformData(\n config.data,\n config.headers,\n config.transformRequest\n );\n\n // Flatten headers\n config.headers = utils.merge(\n config.headers.common || {},\n config.headers[config.method] || {},\n config.headers\n );\n\n utils.forEach(\n ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'],\n function cleanHeaderConfig(method) {\n delete config.headers[method];\n }\n );\n\n var adapter = config.adapter || defaults.adapter;\n\n return adapter(config).then(function onAdapterResolution(response) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n response.data = transformData(\n response.data,\n response.headers,\n config.transformResponse\n );\n\n return response;\n }, function onAdapterRejection(reason) {\n if (!isCancel(reason)) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n if (reason && reason.response) {\n reason.response.data = transformData(\n reason.response.data,\n reason.response.headers,\n config.transformResponse\n );\n }\n }\n\n return Promise.reject(reason);\n });\n};\n","'use strict';\n\nvar utils = require('../utils');\n\n/**\n * Config-specific merge-function which creates a new config-object\n * by merging two configuration objects together.\n *\n * @param {Object} config1\n * @param {Object} config2\n * @returns {Object} New object resulting from merging config2 to config1\n */\nmodule.exports = function mergeConfig(config1, config2) {\n // eslint-disable-next-line no-param-reassign\n config2 = config2 || {};\n var config = {};\n\n var valueFromConfig2Keys = ['url', 'method', 'data'];\n var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params'];\n var defaultToConfig2Keys = [\n 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer',\n 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName',\n 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress',\n 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent',\n 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding'\n ];\n var directMergeKeys = ['validateStatus'];\n\n function getMergedValue(target, source) {\n if (utils.isPlainObject(target) && utils.isPlainObject(source)) {\n return utils.merge(target, source);\n } else if (utils.isPlainObject(source)) {\n return utils.merge({}, source);\n } else if (utils.isArray(source)) {\n return source.slice();\n }\n return source;\n }\n\n function mergeDeepProperties(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n }\n\n utils.forEach(valueFromConfig2Keys, function valueFromConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n }\n });\n\n utils.forEach(mergeDeepPropertiesKeys, mergeDeepProperties);\n\n utils.forEach(defaultToConfig2Keys, function defaultToConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n utils.forEach(directMergeKeys, function merge(prop) {\n if (prop in config2) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (prop in config1) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n var axiosKeys = valueFromConfig2Keys\n .concat(mergeDeepPropertiesKeys)\n .concat(defaultToConfig2Keys)\n .concat(directMergeKeys);\n\n var otherKeys = Object\n .keys(config1)\n .concat(Object.keys(config2))\n .filter(function filterAxiosKeys(key) {\n return axiosKeys.indexOf(key) === -1;\n });\n\n utils.forEach(otherKeys, mergeDeepProperties);\n\n return config;\n};\n","'use strict';\n\nvar utils = require('./../utils');\nvar buildURL = require('../helpers/buildURL');\nvar InterceptorManager = require('./InterceptorManager');\nvar dispatchRequest = require('./dispatchRequest');\nvar mergeConfig = require('./mergeConfig');\n\n/**\n * Create a new instance of Axios\n *\n * @param {Object} instanceConfig The default config for the instance\n */\nfunction Axios(instanceConfig) {\n this.defaults = instanceConfig;\n this.interceptors = {\n request: new InterceptorManager(),\n response: new InterceptorManager()\n };\n}\n\n/**\n * Dispatch a request\n *\n * @param {Object} config The config specific for this request (merged with this.defaults)\n */\nAxios.prototype.request = function request(config) {\n /*eslint no-param-reassign:0*/\n // Allow for axios('example/url'[, config]) a la fetch API\n if (typeof config === 'string') {\n config = arguments[1] || {};\n config.url = arguments[0];\n } else {\n config = config || {};\n }\n\n config = mergeConfig(this.defaults, config);\n\n // Set config.method\n if (config.method) {\n config.method = config.method.toLowerCase();\n } else if (this.defaults.method) {\n config.method = this.defaults.method.toLowerCase();\n } else {\n config.method = 'get';\n }\n\n // Hook up interceptors middleware\n var chain = [dispatchRequest, undefined];\n var promise = Promise.resolve(config);\n\n this.interceptors.request.forEach(function unshiftRequestInterceptors(interceptor) {\n chain.unshift(interceptor.fulfilled, interceptor.rejected);\n });\n\n this.interceptors.response.forEach(function pushResponseInterceptors(interceptor) {\n chain.push(interceptor.fulfilled, interceptor.rejected);\n });\n\n while (chain.length) {\n promise = promise.then(chain.shift(), chain.shift());\n }\n\n return promise;\n};\n\nAxios.prototype.getUri = function getUri(config) {\n config = mergeConfig(this.defaults, config);\n return buildURL(config.url, config.params, config.paramsSerializer).replace(/^\\?/, '');\n};\n\n// Provide aliases for supported request methods\nutils.forEach(['delete', 'get', 'head', 'options'], function forEachMethodNoData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: (config || {}).data\n }));\n };\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, data, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: data\n }));\n };\n});\n\nmodule.exports = Axios;\n","'use strict';\n\n/**\n * A `Cancel` is an object that is thrown when an operation is canceled.\n *\n * @class\n * @param {string=} message The message.\n */\nfunction Cancel(message) {\n this.message = message;\n}\n\nCancel.prototype.toString = function toString() {\n return 'Cancel' + (this.message ? ': ' + this.message : '');\n};\n\nCancel.prototype.__CANCEL__ = true;\n\nmodule.exports = Cancel;\n","'use strict';\n\nvar Cancel = require('./Cancel');\n\n/**\n * A `CancelToken` is an object that can be used to request cancellation of an operation.\n *\n * @class\n * @param {Function} executor The executor function.\n */\nfunction CancelToken(executor) {\n if (typeof executor !== 'function') {\n throw new TypeError('executor must be a function.');\n }\n\n var resolvePromise;\n this.promise = new Promise(function promiseExecutor(resolve) {\n resolvePromise = resolve;\n });\n\n var token = this;\n executor(function cancel(message) {\n if (token.reason) {\n // Cancellation has already been requested\n return;\n }\n\n token.reason = new Cancel(message);\n resolvePromise(token.reason);\n });\n}\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nCancelToken.prototype.throwIfRequested = function throwIfRequested() {\n if (this.reason) {\n throw this.reason;\n }\n};\n\n/**\n * Returns an object that contains a new `CancelToken` and a function that, when called,\n * cancels the `CancelToken`.\n */\nCancelToken.source = function source() {\n var cancel;\n var token = new CancelToken(function executor(c) {\n cancel = c;\n });\n return {\n token: token,\n cancel: cancel\n };\n};\n\nmodule.exports = CancelToken;\n","'use strict';\n\n/**\n * Syntactic sugar for invoking a function and expanding an array for arguments.\n *\n * Common use case would be to use `Function.prototype.apply`.\n *\n * ```js\n * function f(x, y, z) {}\n * var args = [1, 2, 3];\n * f.apply(null, args);\n * ```\n *\n * With `spread` this example can be re-written.\n *\n * ```js\n * spread(function(x, y, z) {})([1, 2, 3]);\n * ```\n *\n * @param {Function} callback\n * @returns {Function}\n */\nmodule.exports = function spread(callback) {\n return function wrap(arr) {\n return callback.apply(null, arr);\n };\n};\n","'use strict';\n\n/**\n * Determines whether the payload is an error thrown by Axios\n *\n * @param {*} payload The value to test\n * @returns {boolean} True if the payload is an error thrown by Axios, otherwise false\n */\nmodule.exports = function isAxiosError(payload) {\n return (typeof payload === 'object') && (payload.isAxiosError === true);\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar bind = require('./helpers/bind');\nvar Axios = require('./core/Axios');\nvar mergeConfig = require('./core/mergeConfig');\nvar defaults = require('./defaults');\n\n/**\n * Create an instance of Axios\n *\n * @param {Object} defaultConfig The default config for the instance\n * @return {Axios} A new instance of Axios\n */\nfunction createInstance(defaultConfig) {\n var context = new Axios(defaultConfig);\n var instance = bind(Axios.prototype.request, context);\n\n // Copy axios.prototype to instance\n utils.extend(instance, Axios.prototype, context);\n\n // Copy context to instance\n utils.extend(instance, context);\n\n return instance;\n}\n\n// Create the default instance to be exported\nvar axios = createInstance(defaults);\n\n// Expose Axios class to allow class inheritance\naxios.Axios = Axios;\n\n// Factory for creating new instances\naxios.create = function create(instanceConfig) {\n return createInstance(mergeConfig(axios.defaults, instanceConfig));\n};\n\n// Expose Cancel & CancelToken\naxios.Cancel = require('./cancel/Cancel');\naxios.CancelToken = require('./cancel/CancelToken');\naxios.isCancel = require('./cancel/isCancel');\n\n// Expose all/spread\naxios.all = function all(promises) {\n return Promise.all(promises);\n};\naxios.spread = require('./helpers/spread');\n\n// Expose isAxiosError\naxios.isAxiosError = require('./helpers/isAxiosError');\n\nmodule.exports = axios;\n\n// Allow use of default import syntax in TypeScript\nmodule.exports.default = axios;\n","module.exports = require('./lib/axios');","\nimport { setLogger } from '../ulity/debug';\nimport * as debug from '../ulity/debug';\nimport Event from '../ulity/event';\nimport Events from '../base/event';\nimport axios from 'axios';\nimport * as Base from '../base/export';\n\nexport default class RTCEndpoint extends Event\n{\n constructor(options)\n {\n super('RTCPusherPlayer');\n this.TAG = '[RTCPusherPlayer]';\n\n let defaults = {\n element: '',// html video element\n debug: false,// if output debug log\n zlmsdpUrl:'',\n simulecast:false,\n useCamera:true,\n audioEnable:true,\n videoEnable:true,\n recvOnly:false,\n resolution:{w:0,h:0}\n };\n \n this.options = Object.assign({}, defaults, options);\n\n if(this.options.debug)\n {\n setLogger();\n }\n\n this.e = {\n onicecandidate:this._onIceCandidate.bind(this),\n ontrack:this._onTrack.bind(this),\n onicecandidateerror:this._onIceCandidateError.bind(this)\n };\n\n this._remoteStream = null;\n this._localStream = null;\n\n this.pc = new RTCPeerConnection(null);\n\n this.pc.onicecandidate = this.e.onicecandidate;\n this.pc.onicecandidateerror = this.e.onicecandidateerror;\n this.pc.ontrack = this.e.ontrack;\n\n if(!this.options.recvOnly && (this.options.audioEnable || this.options.videoEnable))\n this.start();\n else\n this.receive();\n \n }\n\n receive()\n {\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n //debug.error(this.TAG,'this not implement');\n const AudioTransceiverInit = {\n direction: 'recvonly',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'recvonly',\n sendEncodings:[],\n };\n\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n videoTransceiver = this.pc.addTransceiver('video',VideoTransceiverInit);\n \n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/anwser exchange \n this.dispatch(Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED,ret);\n return;\n }\n let anwser = {};\n anwser.sdp = ret.sdp;\n anwser.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n\n this.pc.setRemoteDescription(anwser).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n }\n\n start()\n {\n let videoConstraints = false;\n let audioConstraints = false;\n\n if(this.options.useCamera)\n {\n if(this.options.videoEnable)\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.CAMERA);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n }\n else\n {\n if(this.options.videoEnable)\n {\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.SCREENCAST);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.SCREENCAST);\n }\n else\n {\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n else\n {// error shared display media not only audio\n debug.error(this.TAG,'error paramter');\n }\n }\n \n }\n\n if(this.options.resolution.w !=0 && this.options.resolution.h!=0 && typeof videoConstraints == 'object'){\n videoConstraints.resolution = new Base.Resolution(this.options.resolution.w ,this.options.resolution.h);\n }\n\n Base.MediaStreamFactory.createMediaStream(new Base.StreamConstraints(\n audioConstraints, videoConstraints)).then(stream => {\n\n this._localStream = stream;\n\n this.dispatch(Events.WEBRTC_ON_LOCAL_STREAM,stream);\n\n const AudioTransceiverInit = {\n direction: 'sendrecv',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'sendrecv',\n sendEncodings:[],\n };\n \n if(this.options.simulecast && stream.getVideoTracks().length>0)\n {\n VideoTransceiverInit.sendEncodings = [\n {rid: 'q', active: true, scaleResolutionDownBy: 4.0},\n {rid: 'h', active: true, scaleResolutionDownBy: 2.0},\n {rid: 'f', active: true}\n ];\n }\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n if(stream.getAudioTracks().length>0)\n {\n audioTransceiver = this.pc.addTransceiver(stream.getAudioTracks()[0],\n AudioTransceiverInit);\n }\n else\n {\n AudioTransceiverInit.direction ='recvonly';\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n }\n \n if(stream.getVideoTracks().length>0)\n {\n videoTransceiver = this.pc.addTransceiver(stream.getVideoTracks()[0],\n VideoTransceiverInit);\n }\n else\n {\n VideoTransceiverInit.direction = 'recvonly';\n videoTransceiver = this.pc.addTransceiver('video',\n VideoTransceiverInit);\n }\n\n /*\n stream.getTracks().forEach((track,idx)=>{\n debug.log(this.TAG,track);\n this.pc.addTrack(track);\n });\n */\n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/anwser exchange \n this.dispatch(Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED,ret);\n return;\n }\n let anwser = {};\n anwser.sdp = ret.sdp;\n anwser.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n \n this.pc.setRemoteDescription(anwser).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n\n }).catch(e=>{\n this.dispatch(Events.CAPTURE_STREAM_FAILED);\n //debug.error(this.TAG,e);\n });\n \n //const offerOptions = {};\n /*\n if (typeof this.pc.addTransceiver === 'function') {\n // |direction| seems not working on Safari.\n this.pc.addTransceiver('audio', { direction: 'recvonly' });\n this.pc.addTransceiver('video', { direction: 'recvonly' });\n } else {\n offerOptions.offerToReceiveAudio = true;\n offerOptions.offerToReceiveVideo = true;\n }\n */\n\n\n\n }\n _onIceCandidate(event) {\n if (event.candidate) { \n debug.log('Remote ICE candidate: \\n ' + event.candidate.candidate);\n // Send the candidate to the remote peer\n }\n else {\n // All ICE candidates have been sent\n }\n }\n\n _onTrack(event){\n if(this.options.element && event.streams && event.streams.length>0)\n {\n this.options.element.srcObject = event.streams[0];\n this._remoteStream = event.streams[0];\n\n this.dispatch(Events.WEBRTC_ON_REMOTE_STREAMS,event);\n }\n else\n {\n debug.error('element pararm is failed');\n }\n }\n\n _onIceCandidateError(event){\n this.dispatch(Events.WEBRTC_ICE_CANDIDATE_ERROR,event);\n }\n\n close()\n {\n if(this.pc)\n {\n this.pc.close();\n this.pc=null;\n }\n\n if(this.options)\n {\n this.options=null;\n }\n\n if(this._localStream)\n {\n this._localStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n\n if(this._remoteStream)\n {\n this._remoteStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n }\n\n get remoteStream()\n {\n return this._remoteStream;\n }\n \n get localStream()\n {\n return this._localStream;\n }\n}\n","import * as mediaformat from './mediaformat';\nimport * as MediaFactory from './mediastream-factory';\n\n\nconst quickScan=[\n {\n 'label': '4K(UHD)',\n 'width': 3840,\n 'height': 2160\n },\n {\n 'label': '1080p(FHD)',\n 'width': 1920,\n 'height': 1080\n },\n {\n 'label': 'UXGA',\n 'width': 1600,\n 'height': 1200,\n 'ratio': '4:3'\n },\n {\n 'label': '720p(HD)',\n 'width': 1280,\n 'height': 720\n },\n {\n 'label': 'SVGA',\n 'width': 800,\n 'height': 600\n },\n {\n 'label': 'VGA',\n 'width': 640,\n 'height': 480\n },\n {\n 'label': '360p(nHD)',\n 'width': 640,\n 'height': 360\n },\n {\n 'label': 'CIF',\n 'width': 352,\n 'height': 288\n },\n {\n 'label': 'QVGA',\n 'width': 320,\n 'height': 240\n },\n {\n 'label': 'QCIF',\n 'width': 176,\n 'height': 144\n },\n {\n 'label': 'QQVGA',\n 'width': 160,\n 'height': 120\n }\n];\n\n\n\n\nexport default function GetSupportCameraResolutions(){\n return new Promise(function (resolve, reject) {\n let resolutions = [];\n let ok = 0;\n let err = 0;\n for (let i = 0; i < quickScan.length; ++i) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(quickScan[i].width, quickScan[i].height);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolutions.push(quickScan[i]);\n ok++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n }).catch(e => {\n err++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n });\n }\n });\n}\n\nexport function GetAllScanResolution()\n{\n return quickScan;\n}\nexport function isSupportResolution(w,h)\n{\n return new Promise(function (resolve, reject) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(w,h);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolve();\n }).catch(e => {\n reject(e);\n });\n });\n}","import * as events from './base/event';\nimport * as compile from './ulity/version';\nimport * as media from './base/export';\nimport * as endpoint from './endpoint/endpoint';\nimport * as resolution from './base/resolutionfind';\n\n\n\nconsole.log('build date:',compile.BUILD_DATE);\nconsole.log('version:',compile.VERSION);\n\nexport const Events = events.default;\nexport const Media = media;\nexport const Endpoint = endpoint.default;\nexport const GetSupportCameraResolutions = resolution.default;\nexport const GetAllScanResolution = resolution.GetAllScanResolution;\nexport const isSupportResolution = 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- - - -
-
- - - - - - - - \ No newline at end of file From 8721c5a62b5378ed273c47ff00da875c485f958d Mon Sep 17 00:00:00 2001 From: xgj Date: Tue, 8 Jun 2021 10:07:51 +0800 Subject: [PATCH 202/218] for joy4 rtmp push onMetaData compatibility --- src/Rtmp/RtmpSession.cpp | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/Rtmp/RtmpSession.cpp b/src/Rtmp/RtmpSession.cpp index 7775c553..a1e0caa9 100644 --- a/src/Rtmp/RtmpSession.cpp +++ b/src/Rtmp/RtmpSession.cpp @@ -463,7 +463,10 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) { std::string type = dec.load(); if (type == "@setDataFrame") { setMetaData(dec); - } else { + } else if(type == "onMetaData"){ + //setMetaData(dec); + _publisher_metadata = dec.load(); + }else { TraceP(this) << "unknown notify:" << type; } break; From 89d0a9dedf39ba617d40e666629b7cb9b5ea7843 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 8 Jun 2021 14:49:32 +0800 Subject: [PATCH 203/218] =?UTF-8?q?=E5=B1=8F=E8=94=BD=E5=8A=A8=E6=80=81?= =?UTF-8?q?=E6=B7=BB=E5=8A=A0=E9=85=8D=E7=BD=AE=E6=96=87=E4=BB=B6=E7=9A=84?= =?UTF-8?q?=E7=89=B9=E6=80=A7?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- server/WebApi.cpp | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/server/WebApi.cpp b/server/WebApi.cpp index c6644c7b..5990a8fe 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -401,11 +401,15 @@ void installWebApi() { int changed = API::Success; for (auto &pr : allArgs) { if (ini.find(pr.first) == ini.end()) { +#if 1 //没有这个key - //continue; + continue; +#else // 新增配置选项,为了动态添加多个ffmpeg cmd 模板 ini[pr.first] = pr.second; - continue;// 防止changed变化 + // 防止changed变化 + continue; +#endif } if (ini[pr.first] == pr.second) { continue; From e88281b2ffd46e9e091a73507474c0d0d68a8d76 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 8 Jun 2021 14:49:48 +0800 Subject: [PATCH 204/218] =?UTF-8?q?=E5=88=A0=E9=99=A4=E6=97=A0=E6=95=88?= =?UTF-8?q?=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/H264Rtmp.cpp | 6 ------ 1 file changed, 6 deletions(-) diff --git a/src/Extension/H264Rtmp.cpp b/src/Extension/H264Rtmp.cpp index b196cb6e..19152a6f 100644 --- a/src/Extension/H264Rtmp.cpp +++ b/src/Extension/H264Rtmp.cpp @@ -214,12 +214,6 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) { _lastPacket->buffer.append((char *) &size, 4); _lastPacket->buffer.append(pcData, iLen); _lastPacket->body_size = _lastPacket->buffer.size(); - /* - if (type == H264Frame::NAL_B_P) { - RtmpCodec::inputRtmp(_lastPacket); - _lastPacket = nullptr; - } - */ } void H264RtmpEncoder::makeVideoConfigPkt() { From 0593f65fcb54ad7ab5a3687651a3b6fc3a0bd6fd Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Tue, 8 Jun 2021 14:50:12 +0800 Subject: [PATCH 205/218] =?UTF-8?q?=E5=AE=8C=E5=96=84=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Rtmp/RtmpSession.cpp | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/Rtmp/RtmpSession.cpp b/src/Rtmp/RtmpSession.cpp index a1e0caa9..b061cb29 100644 --- a/src/Rtmp/RtmpSession.cpp +++ b/src/Rtmp/RtmpSession.cpp @@ -463,10 +463,10 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) { std::string type = dec.load(); if (type == "@setDataFrame") { setMetaData(dec); - } else if(type == "onMetaData"){ - //setMetaData(dec); - _publisher_metadata = dec.load(); - }else { + } else if (type == "onMetaData") { + //兼容某些不规范的推流器 + _publisher_metadata = dec.load(); + } else { TraceP(this) << "unknown notify:" << type; } break; From 5b8b5c0ed56caa7371cd7208b87607b783e557e1 Mon Sep 17 00:00:00 2001 From: xgj Date: Fri, 11 Jun 2021 11:04:43 +0800 Subject: [PATCH 206/218] for ssl static link add ld library link --- CMakeLists.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/CMakeLists.txt b/CMakeLists.txt index 54bfed8a..3329bdf4 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -87,6 +87,10 @@ if (OPENSSL_FOUND AND ENABLE_OPENSSL) include_directories(${OPENSSL_INCLUDE_DIR}) add_definitions(-DENABLE_OPENSSL) list(APPEND LINK_LIB_LIST ${OPENSSL_LIBRARIES}) + if (CMAKE_SYSTEM_NAME MATCHES "Linux") + list(APPEND LINK_LIB_LIST dl) + endif() + else() message(WARNING "openssl未找到，rtmp将不支持flash播放器，https/wss/rtsps/rtmps也将失效") endif () From 98d9dcd0098afe8d67e46925a9c3bc88cb15c305 Mon Sep 17 00:00:00 2001 From: xgj Date: Fri, 11 Jun 2021 11:57:55 +0800 Subject: [PATCH 207/218] when link ssl with static library add ld link like this cmake .. -DCMAKE_BUILD_TYPE=Release -DOPENSSL_ROOT_DIR=/usr/local/ssl -DOPENSSL_USE_STATIC_LIBS=TRUE --- CMakeLists.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/CMakeLists.txt b/CMakeLists.txt index 3329bdf4..b3a53a41 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -87,7 +87,7 @@ if (OPENSSL_FOUND AND ENABLE_OPENSSL) include_directories(${OPENSSL_INCLUDE_DIR}) add_definitions(-DENABLE_OPENSSL) list(APPEND LINK_LIB_LIST ${OPENSSL_LIBRARIES}) - if (CMAKE_SYSTEM_NAME MATCHES "Linux") + if (CMAKE_SYSTEM_NAME MATCHES "Linux" AND OPENSSL_USE_STATIC_LIBS) list(APPEND LINK_LIB_LIST dl) endif() From 807f9626d5c65171b792b74684d6b90e8274b2cd Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 16 Jun 2021 11:14:10 +0800 Subject: [PATCH 208/218] =?UTF-8?q?webrtc=E6=96=B0=E5=A2=9E=E5=AF=B9av1?= =?UTF-8?q?=E7=9A=84=E6=94=AF=E6=8C=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- src/Extension/Frame.cpp | 48 ++++++++++------------------------ src/Extension/Frame.h | 34 ++++++++++++++---------- webrtc/Sdp.cpp | 2 +- webrtc/offer.sdp | 58 +++++++++++++++++++++++------------------ 4 files changed, 68 insertions(+), 74 deletions(-) diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index 9d826543..b43206db 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -106,47 +106,27 @@ Frame::Ptr Frame::getCacheAbleFrame(const Frame::Ptr &frame){ return std::make_shared(frame); } -TrackType getTrackType(CodecId codecId){ - switch (codecId){ - case CodecVP8: - case CodecVP9: - case CodecH264: - case CodecH265: return TrackVideo; - case CodecAAC: - case CodecG711A: - case CodecG711U: - case CodecOpus: - case CodecL16: return TrackAudio; - default: return TrackInvalid; +TrackType getTrackType(CodecId codecId) { + switch (codecId) { +#define XX(name, type, value, str) case name : return type; + CODEC_MAP(XX) +#undef XX + default : return TrackInvalid; } } -const char* getCodecName(CodecId codec){ +const char *getCodecName(CodecId codec) { switch (codec) { - case CodecH264 : return "H264"; - case CodecH265 : return "H265"; - case CodecAAC : return "mpeg4-generic"; - case CodecG711A : return "PCMA"; - case CodecG711U : return "PCMU"; - case CodecOpus : return "opus"; - case CodecVP8 : return "VP8"; - case CodecVP9 : return "VP9"; - case CodecL16 : return "L16"; - default: return "invalid"; +#define XX(name, type, value, str) case name : return str; + CODEC_MAP(XX) +#undef XX + default : return "invalid"; } } -static map codec_map = { - {"H264", CodecH264}, - {"H265", CodecH265}, - {"mpeg4-generic", CodecAAC}, - {"PCMA", CodecG711A}, - {"PCMU", CodecG711U}, - {"opus", CodecOpus}, - {"VP8", CodecVP8}, - {"VP9", CodecVP9}, - {"L16", CodecL16} -}; +#define XX(name, type, value, str) {str, name}, +static map codec_map = {CODEC_MAP(XX)}; +#undef XX CodecId getCodecId(const string &str){ auto it = codec_map.find(str); diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index ce242f71..ca8a2b39 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -21,20 +21,6 @@ using namespace toolkit; namespace mediakit{ -typedef enum { - CodecInvalid = -1, - CodecH264 = 0, - CodecH265, - CodecAAC, - CodecG711A, - CodecG711U, - CodecOpus, - CodecL16, - CodecVP8, - CodecVP9, - CodecMax -} CodecId; - typedef enum { TrackInvalid = -1, TrackVideo = 0, @@ -44,6 +30,26 @@ typedef enum { TrackMax } TrackType; +#define CODEC_MAP(XX) \ + XX(CodecH264, TrackVideo, 0, "H264") \ + XX(CodecH265, TrackVideo, 1, "H265") \ + XX(CodecAAC, TrackAudio, 2, "mpeg4-generic") \ + XX(CodecG711A, TrackAudio, 3, "PCMA") \ + XX(CodecG711U, TrackAudio, 4, "PCMU") \ + XX(CodecOpus, TrackAudio, 5, "opus") \ + XX(CodecL16, TrackAudio, 6, "L16") \ + XX(CodecVP8, TrackVideo, 7, "VP8") \ + XX(CodecVP9, TrackVideo, 8, "VP9") \ + XX(CodecAV1, TrackVideo, 9, "AV1X") + +typedef enum { + CodecInvalid = -1, +#define XX(name, type, value, str) name = value, + CODEC_MAP(XX) +#undef XX + CodecMax +} CodecId; + /** * 字符串转媒体类型转 */ diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 5bd993b3..369b3ca8 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -1421,7 +1421,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){ } case TrackVideo: { //此处调整偏好的编码格式优先级 - preferred_codec = {CodecH264, CodecH265}; + preferred_codec = {CodecH264, CodecH265, CodecAV1}; rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli"}; extmap = { RtpExtType::abs_send_time, diff --git a/webrtc/offer.sdp b/webrtc/offer.sdp index 762150d8..c3b09302 100644 --- a/webrtc/offer.sdp +++ b/webrtc/offer.sdp @@ -1,5 +1,5 @@ v=0 -o=- 257973874652185302 2 IN IP4 127.0.0.1 +o=- 8056465047193717905 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 @@ -8,10 +8,10 @@ a=msid-semantic: WMS m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 -a=ice-ufrag:w2IN -a=ice-pwd:X7kCoPoI2NqW8kxuV9LHRR78 +a=ice-ufrag:LtFR +a=ice-pwd:sUVVlvhNoL2g/GL36TyfZGwP a=ice-options:trickle -a=fingerprint:sha-256 7A:A7:A4:9A:BC:37:64:68:9C:48:E5:E9:9B:97:BD:88:17:3E:E5:44:29:4D:6D:BB:AB:2C:85:B8:DE:7A:15:B1 +a=fingerprint:sha-256 21:21:07:E8:3C:D0:3B:45:87:9A:31:86:DE:4F:C1:BA:E1:0E:96:BA:41:36:6E:3A:3F:C6:C8:92:95:5B:71:5F a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level @@ -21,7 +21,7 @@ a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv -a=msid:- 56049f63-4b19-45c4-aa0a-8895049b5430 +a=msid:- 2ebeb64c-2eb3-4c4f-b5d5-d578245b969e a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc @@ -38,17 +38,17 @@ a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 -a=ssrc:3304267696 cname:sCv+hHL1+2UbfMTB -a=ssrc:3304267696 msid:- 56049f63-4b19-45c4-aa0a-8895049b5430 -a=ssrc:3304267696 mslabel:- -a=ssrc:3304267696 label:56049f63-4b19-45c4-aa0a-8895049b5430 -m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 +a=ssrc:905965261 cname:7iEkMV0/MMfqSEce +a=ssrc:905965261 msid:- 2ebeb64c-2eb3-4c4f-b5d5-d578245b969e +a=ssrc:905965261 mslabel:- +a=ssrc:905965261 label:2ebeb64c-2eb3-4c4f-b5d5-d578245b969e +m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 -a=ice-ufrag:w2IN -a=ice-pwd:X7kCoPoI2NqW8kxuV9LHRR78 +a=ice-ufrag:LtFR +a=ice-pwd:sUVVlvhNoL2g/GL36TyfZGwP a=ice-options:trickle -a=fingerprint:sha-256 7A:A7:A4:9A:BC:37:64:68:9C:48:E5:E9:9B:97:BD:88:17:3E:E5:44:29:4D:6D:BB:AB:2C:85:B8:DE:7A:15:B1 +a=fingerprint:sha-256 21:21:07:E8:3C:D0:3B:45:87:9A:31:86:DE:4F:C1:BA:E1:0E:96:BA:41:36:6E:3A:3F:C6:C8:92:95:5B:71:5F a=setup:actpass a=mid:1 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset @@ -63,7 +63,7 @@ a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv -a=msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 +a=msid:- f36bb41d-d05d-4310-b05b-7913d0029b18 a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 @@ -128,13 +128,21 @@ a=rtcp-fb:108 nack pli a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 +a=rtpmap:35 AV1X/90000 +a=rtcp-fb:35 goog-remb +a=rtcp-fb:35 transport-cc +a=rtcp-fb:35 ccm fir +a=rtcp-fb:35 nack +a=rtcp-fb:35 nack pli +a=rtpmap:36 rtx/90000 +a=fmtp:36 apt=35 a=rtpmap:124 H264/90000 a=rtcp-fb:124 goog-remb a=rtcp-fb:124 transport-cc a=rtcp-fb:124 ccm fir a=rtcp-fb:124 nack a=rtcp-fb:124 nack pli -a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f +a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:119 rtx/90000 a=fmtp:119 apt=124 a=rtpmap:123 H264/90000 @@ -143,19 +151,19 @@ a=rtcp-fb:123 transport-cc a=rtcp-fb:123 ccm fir a=rtcp-fb:123 nack a=rtcp-fb:123 nack pli -a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f +a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:118 rtx/90000 a=fmtp:118 apt=123 a=rtpmap:114 red/90000 a=rtpmap:115 rtx/90000 a=fmtp:115 apt=114 a=rtpmap:116 ulpfec/90000 -a=ssrc-group:FID 1128910219 3552306261 -a=ssrc:1128910219 cname:sCv+hHL1+2UbfMTB -a=ssrc:1128910219 msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 -a=ssrc:1128910219 mslabel:- -a=ssrc:1128910219 label:a73b3f5f-007a-4a46-ac8b-582ca7fee460 -a=ssrc:3552306261 cname:sCv+hHL1+2UbfMTB -a=ssrc:3552306261 msid:- a73b3f5f-007a-4a46-ac8b-582ca7fee460 -a=ssrc:3552306261 mslabel:- -a=ssrc:3552306261 label:a73b3f5f-007a-4a46-ac8b-582ca7fee460 \ No newline at end of file +a=ssrc-group:FID 2678501654 361960375 +a=ssrc:2678501654 cname:7iEkMV0/MMfqSEce +a=ssrc:2678501654 msid:- f36bb41d-d05d-4310-b05b-7913d0029b18 +a=ssrc:2678501654 mslabel:- +a=ssrc:2678501654 label:f36bb41d-d05d-4310-b05b-7913d0029b18 +a=ssrc:361960375 cname:7iEkMV0/MMfqSEce +a=ssrc:361960375 msid:- f36bb41d-d05d-4310-b05b-7913d0029b18 +a=ssrc:361960375 mslabel:- +a=ssrc:361960375 label:f36bb41d-d05d-4310-b05b-7913d0029b18 \ No newline at end of file From 1920802764d7eb50e2cb87c4cfea79b314c31635 Mon Sep 17 00:00:00 2001 From: xgj Date: Thu, 17 Jun 2021 18:08:35 +0800 Subject: [PATCH 209/218] for h264 multi slice on frame compatibility --- src/Extension/Frame.cpp | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index b43206db..61b6d613 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -191,9 +191,12 @@ bool FrameMerger::willFlush(const Frame::Ptr &frame) const{ } switch (frame->getCodecId()) { case CodecH264 : { - if (H264_TYPE(frame->data()[frame->prefixSize()]) == H264Frame::NAL_B_P) { - //如果是264的b/p帧，那么也刷新输出 + if (frame->data()[frame->prefixSize()+1]&0x80 !=0) { + //264 新一帧的开始，刷新输出 return true; + }else{ + // 不刷新输出 + return false; } break; } From 3429690d75cb0702eec6255bf0ec661d59f27bdd Mon Sep 17 00:00:00 2001 From: xgj Date: Thu, 17 Jun 2021 19:26:58 +0800 Subject: [PATCH 210/218] for h264 pps sps aud sei not check first_mb_in_slice --- src/Extension/Frame.cpp | 3 ++- src/Extension/H264Rtmp.cpp | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index 61b6d613..5864908b 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -191,7 +191,8 @@ bool FrameMerger::willFlush(const Frame::Ptr &frame) const{ } switch (frame->getCodecId()) { case CodecH264 : { - if (frame->data()[frame->prefixSize()+1]&0x80 !=0) { + auto type = H264_TYPE(frame->data()[frame->prefixSize()]); + if (frame->data()[frame->prefixSize()+1]&0x80 !=0 && type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR ) {// sei aud pps sps 不判断 //264 新一帧的开始，刷新输出 return true; }else{ diff --git a/src/Extension/H264Rtmp.cpp b/src/Extension/H264Rtmp.cpp index 19152a6f..47f5192e 100644 --- a/src/Extension/H264Rtmp.cpp +++ b/src/Extension/H264Rtmp.cpp @@ -183,7 +183,7 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) { } } - if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || pcData[1]&0x80 != 0)) { + if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || (pcData[1]&0x80 != 0 && type>=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR))) { RtmpCodec::inputRtmp(_lastPacket); _lastPacket = nullptr; } From a6e9652da843f111f0232ff9494a40f12b62e9b0 Mon Sep 17 00:00:00 2001 From: xgj Date: Fri, 18 Jun 2021 12:40:42 +0800 Subject: [PATCH 211/218] for h264/h265 judge a frame start more safe(h264 is first_mb_in_slice h265 is first_slice_segment_in_pic_flag) and for h265 only idr is key frame --- src/Extension/Frame.cpp | 9 ++++++--- src/Extension/H264Rtmp.cpp | 2 +- src/Extension/H265.cpp | 7 ++++--- src/Extension/H265Rtmp.cpp | 2 +- 4 files changed, 12 insertions(+), 8 deletions(-) diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index 5864908b..f51f5945 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -192,7 +192,7 @@ bool FrameMerger::willFlush(const Frame::Ptr &frame) const{ switch (frame->getCodecId()) { case CodecH264 : { auto type = H264_TYPE(frame->data()[frame->prefixSize()]); - if (frame->data()[frame->prefixSize()+1]&0x80 !=0 && type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR ) {// sei aud pps sps 不判断 + if ((frame->data()[frame->prefixSize()+1]&0x80) !=0 && type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR ) {// sei aud pps sps 不判断 //264 新一帧的开始，刷新输出 return true; }else{ @@ -202,9 +202,12 @@ bool FrameMerger::willFlush(const Frame::Ptr &frame) const{ break; } case CodecH265 : { - if (H265_TYPE(frame->data()[frame->prefixSize()]) == H265Frame::NAL_TRAIL_R) { - //如果是265的TRAIL_R帧，那么也刷新输出 + auto type = H265_TYPE(frame->data()[frame->prefixSize()]); + if ((type>=H265Frame::NAL_TRAIL_R &&type<= H265Frame::NAL_RSV_IRAP_VCL23) && ( (frame->data()[frame->prefixSize()+2]>>7 & 0x01) != 0)) { + //first_slice_segment_in_pic_flag is frame start return true; + }else{ + return false; } break; } diff --git a/src/Extension/H264Rtmp.cpp b/src/Extension/H264Rtmp.cpp index 47f5192e..b3426fe8 100644 --- a/src/Extension/H264Rtmp.cpp +++ b/src/Extension/H264Rtmp.cpp @@ -183,7 +183,7 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) { } } - if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || (pcData[1]&0x80 != 0 && type>=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR))) { + if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || ((pcData[1]&0x80) != 0 && type>=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR))) { RtmpCodec::inputRtmp(_lastPacket); _lastPacket = nullptr; } diff --git a/src/Extension/H265.cpp b/src/Extension/H265.cpp index 76df36a7..a4b2dde6 100644 --- a/src/Extension/H265.cpp +++ b/src/Extension/H265.cpp @@ -64,10 +64,11 @@ bool H265Frame::configFrame() const { } bool H265Frame::isKeyFrame(int type, const char *ptr) { - if (!ptr || type != NAL_IDR_W_RADL) { - return type >= NAL_BLA_W_LP && type <= NAL_RSV_IRAP_VCL23; + if(ptr){ + return (((*((uint8_t *) ptr + 2)) >> 7) & 0x01) == 1 && (type == NAL_IDR_N_LP || type == NAL_IDR_W_RADL); } - return (((*((uint8_t *) ptr + 2)) >> 7) & 0x01) == 1; + return false; + } H265Frame::H265Frame(){ diff --git a/src/Extension/H265Rtmp.cpp b/src/Extension/H265Rtmp.cpp index 3a8c32ab..f863b502 100644 --- a/src/Extension/H265Rtmp.cpp +++ b/src/Extension/H265Rtmp.cpp @@ -169,7 +169,7 @@ void H265RtmpEncoder::inputFrame(const Frame::Ptr &frame) { return; } - if (_lastPacket && (_lastPacket->time_stamp != frame->dts() || type == H265Frame::NAL_TRAIL_R)) { + if (_lastPacket && (_lastPacket->time_stamp != frame->dts() || (type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR && (pcData[2]>>7 &0x01) !=0))) { RtmpCodec::inputRtmp(_lastPacket); _lastPacket = nullptr; } From 5fc5a9ef5fa4e443c4783095459f2ccf38381fe7 Mon Sep 17 00:00:00 2001 From: wxf Date: Fri, 18 Jun 2021 13:36:25 +0800 Subject: [PATCH 212/218] =?UTF-8?q?webrtc:=20=E4=BF=AE=E6=AD=A3=20sdp=20?= =?UTF-8?q?=E4=B8=AD=20candidate=20=E8=A7=A3=E6=9E=90=E9=97=AE=E9=A2=98?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 见 https://datatracker.ietf.org/doc/html/rfc5245#section-15.1 > foundation = 1*32ice-char --- webrtc/Sdp.cpp | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 369b3ca8..892a9afb 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -704,12 +704,13 @@ string SdpAttrSctpMap::toString() const { } void SdpAttrCandidate::parse(const string &str) { - char foundation_buf[32] = {0}; + char foundation_buf[40] = {0}; char transport_buf[16] = {0}; char address_buf[32] = {0}; char type_buf[16] = {0}; - if (7 != sscanf(str.data(), "%31[^ ] %" SCNu32 " %15[^ ] %" SCNu32 " %31[^ ] %" SCNu16 " typ %15[^ ]", + // https://datatracker.ietf.org/doc/html/rfc5245#section-15.1 + if (7 != sscanf(str.data(), "%32[^ ] %" SCNu32 " %15[^ ] %" SCNu32 " %31[^ ] %" SCNu16 " typ %15[^ ]", foundation_buf, &component, transport_buf, &priority, address_buf, &port, type_buf)) { SDP_THROW(); } From 2335043cdf2e6d2db0b98e484568ed669469e565 Mon Sep 17 00:00:00 2001 From: xgj Date: Fri, 18 Jun 2021 15:19:21 +0800 Subject: [PATCH 213/218] drop SEI and AUD for h264 and h265 avoid sei as a frame --- src/Extension/Frame.cpp | 27 ++++++++++++++++++++++++++- src/Extension/Frame.h | 1 + src/Extension/H265Rtmp.cpp | 4 ++-- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/src/Extension/Frame.cpp b/src/Extension/Frame.cpp index f51f5945..62753f59 100644 --- a/src/Extension/Frame.cpp +++ b/src/Extension/Frame.cpp @@ -244,8 +244,33 @@ void FrameMerger::doMerge(BufferLikeString &merged, const Frame::Ptr &frame) con default: /*不可达*/ assert(0); break; } } - +bool FrameMerger::shouldDrop(const Frame::Ptr &frame) const{ + switch (frame->getCodecId()) { + case CodecH264:{ + auto type = H264_TYPE(frame->data()[frame->prefixSize()]); + if(type == H264Frame::NAL_SEI || type == H264Frame::NAL_AUD){ + // 防止吧AUD或者SEI当成一帧 + return true; + } + break; + } + case CodecH265: { + //如果是新的一帧，前面的缓存需要输出 + auto type = H265_TYPE(frame->data()[frame->prefixSize()]); + if(type == H265Frame::NAL_AUD || type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX){ + // 防止吧AUD或者SEI当成一帧 + return true; + } + break; + } + default: break; + } + return false; +} void FrameMerger::inputFrame(const Frame::Ptr &frame, const onOutput &cb) { + if(shouldDrop(frame)){ + return; + } if (willFlush(frame)) { Frame::Ptr back = _frameCached.back(); Buffer::Ptr merged_frame = back; diff --git a/src/Extension/Frame.h b/src/Extension/Frame.h index ca8a2b39..95a5ec6d 100644 --- a/src/Extension/Frame.h +++ b/src/Extension/Frame.h @@ -475,6 +475,7 @@ public: private: bool willFlush(const Frame::Ptr &frame) const; void doMerge(BufferLikeString &buffer, const Frame::Ptr &frame) const; + bool shouldDrop(const Frame::Ptr &frame) const; private: int _type; diff --git a/src/Extension/H265Rtmp.cpp b/src/Extension/H265Rtmp.cpp index f863b502..84dddad6 100644 --- a/src/Extension/H265Rtmp.cpp +++ b/src/Extension/H265Rtmp.cpp @@ -165,8 +165,8 @@ void H265RtmpEncoder::inputFrame(const Frame::Ptr &frame) { } } - if(type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX){ - return; + if(type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX || type == H265Frame::NAL_AUD){ + return;// 防止sei aud 作为一帧 } if (_lastPacket && (_lastPacket->time_stamp != frame->dts() || (type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR && (pcData[2]>>7 &0x01) !=0))) { From 89d8f005f8bdad0c74afaa574f15a18272958665 Mon Sep 17 00:00:00 2001 From: Johnny Date: Sat, 19 Jun 2021 23:56:17 +0800 Subject: [PATCH 214/218] fix: fix typo. --- webrtc/RtpExt.cpp | 2 +- webrtc/readme.md | 2 +- www/webrtc/ZLMRTCClient.js | 30 +++++++++++++++--------------- www/webrtc/ZLMRTCClient.js.map | 2 +- www/webrtc/index.html | 32 ++++++++++++++++---------------- 5 files changed, 34 insertions(+), 34 deletions(-) diff --git a/webrtc/RtpExt.cpp b/webrtc/RtpExt.cpp index ecc39c5d..dd9a1e2e 100644 --- a/webrtc/RtpExt.cpp +++ b/webrtc/RtpExt.cpp @@ -390,7 +390,7 @@ string RtpExt::getSdesMid() const { //https://tools.ietf.org/html/draft-ietf-avtext-rid-06 -//用于simulecast +//用于simulcast //3.1. RTCP 'RtpStreamId' SDES Extension // // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 diff --git a/webrtc/readme.md b/webrtc/readme.md index ff832278..87d21b85 100644 --- a/webrtc/readme.md +++ b/webrtc/readme.md @@ -31,7 +31,7 @@ ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟 - 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。 - 2、实现rtp重传等相关功能。 -- 3、实现simulecast相关功能。 +- 3、实现simulcast相关功能。 - 4、fec、rtp扩展等其他功能。 - 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。 diff --git a/www/webrtc/ZLMRTCClient.js b/www/webrtc/ZLMRTCClient.js index c495dc1a..a92ed4a4 100644 --- a/www/webrtc/ZLMRTCClient.js +++ b/www/webrtc/ZLMRTCClient.js @@ -4,7 +4,7 @@ var ZLMRTCClient = (function (exports) { const Events$1 = { WEBRTC_NOT_SUPPORT: 'WEBRTC_NOT_SUPPORT', WEBRTC_ICE_CANDIDATE_ERROR: 'WEBRTC_ICE_CANDIDATE_ERROR', - WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED: 'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED', + WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED: 'WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED', WEBRTC_ON_REMOTE_STREAMS: 'WEBRTC_ON_REMOTE_STREAMS', WEBRTC_ON_LOCAL_STREAM: 'WEBRTC_ON_LOCAL_STREAM', CAPTURE_STREAM_FAILED: 'CAPTURE_STREAM_FAILED' @@ -7285,7 +7285,7 @@ var ZLMRTCClient = (function (exports) { debug: false, // if output debug log zlmsdpUrl: '', - simulecast: false, + simulcast: false, useCamera: true, audioEnable: true, videoEnable: true, @@ -7342,16 +7342,16 @@ var ZLMRTCClient = (function (exports) { let ret = response.data; //JSON.parse(response.data); if (ret.code != 0) { - // mean failed for offer/anwser exchange - this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); + // mean failed for offer/answer exchange + this.dispatch(Events$1.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED, ret); return; } - let anwser = {}; - anwser.sdp = ret.sdp; - anwser.type = 'answer'; + let answer = {}; + answer.sdp = ret.sdp; + answer.type = 'answer'; log(this.TAG, 'answer:', ret.sdp); - this.pc.setRemoteDescription(anwser).then(() => { + this.pc.setRemoteDescription(answer).then(() => { log(this.TAG, 'set remote sucess'); }).catch(e => { error(this.TAG, e); @@ -7398,7 +7398,7 @@ var ZLMRTCClient = (function (exports) { sendEncodings: [] }; - if (this.options.simulecast && stream.getVideoTracks().length > 0) { + if (this.options.simulcast && stream.getVideoTracks().length > 0) { VideoTransceiverInit.sendEncodings = [{ rid: 'q', active: true, @@ -7449,16 +7449,16 @@ var ZLMRTCClient = (function (exports) { let ret = response.data; //JSON.parse(response.data); if (ret.code != 0) { - // mean failed for offer/anwser exchange - this.dispatch(Events$1.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, ret); + // mean failed for offer/answer exchange + this.dispatch(Events$1.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED, ret); return; } - let anwser = {}; - anwser.sdp = ret.sdp; - anwser.type = 'answer'; + let answer = {}; + answer.sdp = ret.sdp; + answer.type = 'answer'; log(this.TAG, 'answer:', ret.sdp); - this.pc.setRemoteDescription(anwser).then(() => { + this.pc.setRemoteDescription(answer).then(() => { log(this.TAG, 'set remote sucess'); }).catch(e => { error(this.TAG, e); diff --git a/www/webrtc/ZLMRTCClient.js.map b/www/webrtc/ZLMRTCClient.js.map index 28184d8d..fa9114c4 100644 --- a/www/webrtc/ZLMRTCClient.js.map +++ b/www/webrtc/ZLMRTCClient.js.map @@ -1 +1 @@ -{"version":3,"file":"ZLMRTCClient.js","sources":["../src/base/event.js","../src/ulity/version.js","../src/base/utils.js","../src/base/mediaformat.js","../node_modules/webrtc-adapter/src/js/utils.js","../node_modules/webrtc-adapter/src/js/chrome/getusermedia.js","../node_modules/webrtc-adapter/src/js/chrome/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/chrome/chrome_shim.js","../node_modules/webrtc-adapter/src/js/edge/filtericeservers.js","../node_modules/sdp/sdp.js","../node_modules/rtcpeerconnection-shim/rtcpeerconnection.js","../node_modules/webrtc-adapter/src/js/edge/getusermedia.js","../node_modules/webrtc-adapter/src/js/edge/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/edge/edge_shim.js","../node_modules/webrtc-adapter/src/js/firefox/getusermedia.js","../node_modules/webrtc-adapter/src/js/firefox/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/firefox/firefox_shim.js","../node_modules/webrtc-adapter/src/js/safari/safari_shim.js","../node_modules/webrtc-adapter/src/js/common_shim.js","../node_modules/webrtc-adapter/src/js/adapter_factory.js","../node_modules/webrtc-adapter/src/js/adapter_core.js","../src/base/mediastream-factory.js","../src/base/export.js","../src/ulity/debug.js","../src/ulity/event.js","../node_modules/axios/lib/helpers/bind.js","../node_modules/axios/lib/utils.js","../node_modules/axios/lib/helpers/buildURL.js","../node_modules/axios/lib/core/InterceptorManager.js","../node_modules/axios/lib/core/transformData.js","../node_modules/axios/lib/cancel/isCancel.js","../node_modules/axios/lib/helpers/normalizeHeaderName.js","../node_modules/axios/lib/core/enhanceError.js","../node_modules/axios/lib/core/createError.js","../node_modules/axios/lib/core/settle.js","../node_modules/axios/lib/helpers/cookies.js","../node_modules/axios/lib/helpers/isAbsoluteURL.js","../node_modules/axios/lib/helpers/combineURLs.js","../node_modules/axios/lib/core/buildFullPath.js","../node_modules/axios/lib/helpers/parseHeaders.js","../node_modules/axios/lib/helpers/isURLSameOrigin.js","../node_modules/axios/lib/adapters/xhr.js","../node_modules/axios/lib/defaults.js","../node_modules/axios/lib/core/dispatchRequest.js","../node_modules/axios/lib/core/mergeConfig.js","../node_modules/axios/lib/core/Axios.js","../node_modules/axios/lib/cancel/Cancel.js","../node_modules/axios/lib/cancel/CancelToken.js","../node_modules/axios/lib/helpers/spread.js","../node_modules/axios/lib/helpers/isAxiosError.js","../node_modules/axios/lib/axios.js","../node_modules/axios/index.js","../src/endpoint/endpoint.js","../src/base/resolutionfind.js","../src/export.js"],"sourcesContent":["const Events = {\n\tWEBRTC_NOT_SUPPORT : 'WEBRTC_NOT_SUPPORT',\n\tWEBRTC_ICE_CANDIDATE_ERROR : 'WEBRTC_ICE_CANDIDATE_ERROR',\n\tWEBRTC_OFFER_ANWSER_EXCHANGE_FAILED:'WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED',\n\tWEBRTC_ON_REMOTE_STREAMS:'WEBRTC_ON_REMOTE_STREAMS',\n\tWEBRTC_ON_LOCAL_STREAM:'WEBRTC_ON_LOCAL_STREAM',\n\tCAPTURE_STREAM_FAILED:'CAPTURE_STREAM_FAILED'\n};\n\nexport default Events;","export const VERSION = '__VERSION__';\nexport const BUILD_DATE = '__BUILD_DATE__';","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n\n// eslint-disable-next-line require-jsdoc\nexport function isFirefox() {\n return window.navigator.userAgent.match('Firefox') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isChrome() {\n return window.navigator.userAgent.match('Chrome') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isSafari() {\n return /^((?!chrome|android).)*safari/i.test(window.navigator.userAgent);\n}\n// eslint-disable-next-line require-jsdoc\nexport function isEdge() {\n return window.navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/) !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function createUuid() {\n return 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx'.replace(/[xy]/g, function(c) {\n const r = Math.random() * 16 | 0;\n const v = c === 'x' ? r : (r & 0x3 | 0x8);\n return v.toString(16);\n });\n}","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n/**\n * @class AudioSourceInfo\n * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const AudioSourceInfo = {\n MIC: 'mic',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class VideoSourceInfo\n * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const VideoSourceInfo = {\n CAMERA: 'camera',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class TrackKind\n * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const TrackKind = {\n /**\n * Audio tracks.\n * @type string\n */\n AUDIO: 'audio',\n /**\n * Video tracks.\n * @type string\n */\n VIDEO: 'video',\n /**\n * Both audio and video tracks.\n * @type string\n */\n AUDIO_AND_VIDEO: 'av',\n};\n/**\n * @class Resolution\n * @memberOf Owt.Base\n * @classDesc The Resolution defines the size of a rectangle.\n * @constructor\n * @param {number} width\n * @param {number} height\n */\nexport class Resolution {\n // eslint-disable-next-line require-jsdoc\n constructor(width, height) {\n /**\n * @member {number} width\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.width = width;\n /**\n * @member {number} height\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.height = height;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nlet logDisabled_ = true;\nlet deprecationWarnings_ = true;\n\n/**\n * Extract browser version out of the provided user agent string.\n *\n * @param {!string} uastring userAgent string.\n * @param {!string} expr Regular expression used as match criteria.\n * @param {!number} pos position in the version string to be returned.\n * @return {!number} browser version.\n */\nexport function extractVersion(uastring, expr, pos) {\n const match = uastring.match(expr);\n return match && match.length >= pos && parseInt(match[pos], 10);\n}\n\n// Wraps the peerconnection event eventNameToWrap in a function\n// which returns the modified event object (or false to prevent\n// the event).\nexport function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) {\n if (!window.RTCPeerConnection) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n const nativeAddEventListener = proto.addEventListener;\n proto.addEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap) {\n return nativeAddEventListener.apply(this, arguments);\n }\n const wrappedCallback = (e) => {\n const modifiedEvent = wrapper(e);\n if (modifiedEvent) {\n if (cb.handleEvent) {\n cb.handleEvent(modifiedEvent);\n } else {\n cb(modifiedEvent);\n }\n }\n };\n this._eventMap = this._eventMap || {};\n if (!this._eventMap[eventNameToWrap]) {\n this._eventMap[eventNameToWrap] = new Map();\n }\n this._eventMap[eventNameToWrap].set(cb, wrappedCallback);\n return nativeAddEventListener.apply(this, [nativeEventName,\n wrappedCallback]);\n };\n\n const nativeRemoveEventListener = proto.removeEventListener;\n proto.removeEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap || !this._eventMap\n || !this._eventMap[eventNameToWrap]) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n if (!this._eventMap[eventNameToWrap].has(cb)) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n const unwrappedCb = this._eventMap[eventNameToWrap].get(cb);\n this._eventMap[eventNameToWrap].delete(cb);\n if (this._eventMap[eventNameToWrap].size === 0) {\n delete this._eventMap[eventNameToWrap];\n }\n if (Object.keys(this._eventMap).length === 0) {\n delete this._eventMap;\n }\n return nativeRemoveEventListener.apply(this, [nativeEventName,\n unwrappedCb]);\n };\n\n Object.defineProperty(proto, 'on' + eventNameToWrap, {\n get() {\n return this['_on' + eventNameToWrap];\n },\n set(cb) {\n if (this['_on' + eventNameToWrap]) {\n this.removeEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap]);\n delete this['_on' + eventNameToWrap];\n }\n if (cb) {\n this.addEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap] = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n}\n\nexport function disableLog(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n logDisabled_ = bool;\n return (bool) ? 'adapter.js logging disabled' :\n 'adapter.js logging enabled';\n}\n\n/**\n * Disable or enable deprecation warnings\n * @param {!boolean} bool set to true to disable warnings.\n */\nexport function disableWarnings(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n deprecationWarnings_ = !bool;\n return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled');\n}\n\nexport function log() {\n if (typeof window === 'object') {\n if (logDisabled_) {\n return;\n }\n if (typeof console !== 'undefined' && typeof console.log === 'function') {\n console.log.apply(console, arguments);\n }\n }\n}\n\n/**\n * Shows a deprecation warning suggesting the modern and spec-compatible API.\n */\nexport function deprecated(oldMethod, newMethod) {\n if (!deprecationWarnings_) {\n return;\n }\n console.warn(oldMethod + ' is deprecated, please use ' + newMethod +\n ' instead.');\n}\n\n/**\n * Browser detector.\n *\n * @return {object} result containing browser and version\n * properties.\n */\nexport function detectBrowser(window) {\n // Returned result object.\n const result = {browser: null, version: null};\n\n // Fail early if it's not a browser\n if (typeof window === 'undefined' || !window.navigator) {\n result.browser = 'Not a browser.';\n return result;\n }\n\n const {navigator} = window;\n\n if (navigator.mozGetUserMedia) { // Firefox.\n result.browser = 'firefox';\n result.version = extractVersion(navigator.userAgent,\n /Firefox\\/(\\d+)\\./, 1);\n } else if (navigator.webkitGetUserMedia ||\n (window.isSecureContext === false && window.webkitRTCPeerConnection &&\n !window.RTCIceGatherer)) {\n // Chrome, Chromium, Webview, Opera.\n // Version matches Chrome/WebRTC version.\n // Chrome 74 removed webkitGetUserMedia on http as well so we need the\n // more complicated fallback to webkitRTCPeerConnection.\n result.browser = 'chrome';\n result.version = extractVersion(navigator.userAgent,\n /Chrom(e|ium)\\/(\\d+)\\./, 2);\n } else if (navigator.mediaDevices &&\n navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/)) { // Edge.\n result.browser = 'edge';\n result.version = extractVersion(navigator.userAgent,\n /Edge\\/(\\d+).(\\d+)$/, 2);\n } else if (window.RTCPeerConnection &&\n navigator.userAgent.match(/AppleWebKit\\/(\\d+)\\./)) { // Safari.\n result.browser = 'safari';\n result.version = extractVersion(navigator.userAgent,\n /AppleWebKit\\/(\\d+)\\./, 1);\n result.supportsUnifiedPlan = window.RTCRtpTransceiver &&\n 'currentDirection' in window.RTCRtpTransceiver.prototype;\n } else { // Default fallthrough: not supported.\n result.browser = 'Not a supported browser.';\n return result;\n }\n\n return result;\n}\n\n/**\n * Checks if something is an object.\n *\n * @param {*} val The something you want to check.\n * @return true if val is an object, false otherwise.\n */\nfunction isObject(val) {\n return Object.prototype.toString.call(val) === '[object Object]';\n}\n\n/**\n * Remove all empty objects and undefined values\n * from a nested object -- an enhanced and vanilla version\n * of Lodash's `compact`.\n */\nexport function compactObject(data) {\n if (!isObject(data)) {\n return data;\n }\n\n return Object.keys(data).reduce(function(accumulator, key) {\n const isObj = isObject(data[key]);\n const value = isObj ? compactObject(data[key]) : data[key];\n const isEmptyObject = isObj && !Object.keys(value).length;\n if (value === undefined || isEmptyObject) {\n return accumulator;\n }\n return Object.assign(accumulator, {[key]: value});\n }, {});\n}\n\n/* iterates the stats graph recursively. */\nexport function walkStats(stats, base, resultSet) {\n if (!base || resultSet.has(base.id)) {\n return;\n }\n resultSet.set(base.id, base);\n Object.keys(base).forEach(name => {\n if (name.endsWith('Id')) {\n walkStats(stats, stats.get(base[name]), resultSet);\n } else if (name.endsWith('Ids')) {\n base[name].forEach(id => {\n walkStats(stats, stats.get(id), resultSet);\n });\n }\n });\n}\n\n/* filter getStats for a sender/receiver track. */\nexport function filterStats(result, track, outbound) {\n const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp';\n const filteredResult = new Map();\n if (track === null) {\n return filteredResult;\n }\n const trackStats = [];\n result.forEach(value => {\n if (value.type === 'track' &&\n value.trackIdentifier === track.id) {\n trackStats.push(value);\n }\n });\n trackStats.forEach(trackStat => {\n result.forEach(stats => {\n if (stats.type === streamStatsType && stats.trackId === trackStat.id) {\n walkStats(result, stats, filteredResult);\n }\n });\n });\n return filteredResult;\n}\n\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\nconst logging = utils.log;\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n\n if (!navigator.mediaDevices) {\n return;\n }\n\n const constraintsToChrome_ = function(c) {\n if (typeof c !== 'object' || c.mandatory || c.optional) {\n return c;\n }\n const cc = {};\n Object.keys(c).forEach(key => {\n if (key === 'require' || key === 'advanced' || key === 'mediaSource') {\n return;\n }\n const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]};\n if (r.exact !== undefined && typeof r.exact === 'number') {\n r.min = r.max = r.exact;\n }\n const oldname_ = function(prefix, name) {\n if (prefix) {\n return prefix + name.charAt(0).toUpperCase() + name.slice(1);\n }\n return (name === 'deviceId') ? 'sourceId' : name;\n };\n if (r.ideal !== undefined) {\n cc.optional = cc.optional || [];\n let oc = {};\n if (typeof r.ideal === 'number') {\n oc[oldname_('min', key)] = r.ideal;\n cc.optional.push(oc);\n oc = {};\n oc[oldname_('max', key)] = r.ideal;\n cc.optional.push(oc);\n } else {\n oc[oldname_('', key)] = r.ideal;\n cc.optional.push(oc);\n }\n }\n if (r.exact !== undefined && typeof r.exact !== 'number') {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_('', key)] = r.exact;\n } else {\n ['min', 'max'].forEach(mix => {\n if (r[mix] !== undefined) {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_(mix, key)] = r[mix];\n }\n });\n }\n });\n if (c.advanced) {\n cc.optional = (cc.optional || []).concat(c.advanced);\n }\n return cc;\n };\n\n const shimConstraints_ = function(constraints, func) {\n if (browserDetails.version >= 61) {\n return func(constraints);\n }\n constraints = JSON.parse(JSON.stringify(constraints));\n if (constraints && typeof constraints.audio === 'object') {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n constraints = JSON.parse(JSON.stringify(constraints));\n remap(constraints.audio, 'autoGainControl', 'googAutoGainControl');\n remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression');\n constraints.audio = constraintsToChrome_(constraints.audio);\n }\n if (constraints && typeof constraints.video === 'object') {\n // Shim facingMode for mobile & surface pro.\n let face = constraints.video.facingMode;\n face = face && ((typeof face === 'object') ? face : {ideal: face});\n const getSupportedFacingModeLies = browserDetails.version < 66;\n\n if ((face && (face.exact === 'user' || face.exact === 'environment' ||\n face.ideal === 'user' || face.ideal === 'environment')) &&\n !(navigator.mediaDevices.getSupportedConstraints &&\n navigator.mediaDevices.getSupportedConstraints().facingMode &&\n !getSupportedFacingModeLies)) {\n delete constraints.video.facingMode;\n let matches;\n if (face.exact === 'environment' || face.ideal === 'environment') {\n matches = ['back', 'rear'];\n } else if (face.exact === 'user' || face.ideal === 'user') {\n matches = ['front'];\n }\n if (matches) {\n // Look for matches in label, or use last cam for back (typical).\n return navigator.mediaDevices.enumerateDevices()\n .then(devices => {\n devices = devices.filter(d => d.kind === 'videoinput');\n let dev = devices.find(d => matches.some(match =>\n d.label.toLowerCase().includes(match)));\n if (!dev && devices.length && matches.includes('back')) {\n dev = devices[devices.length - 1]; // more likely the back cam\n }\n if (dev) {\n constraints.video.deviceId = face.exact ? {exact: dev.deviceId} :\n {ideal: dev.deviceId};\n }\n constraints.video = constraintsToChrome_(constraints.video);\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n });\n }\n }\n constraints.video = constraintsToChrome_(constraints.video);\n }\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n };\n\n const shimError_ = function(e) {\n if (browserDetails.version >= 64) {\n return e;\n }\n return {\n name: {\n PermissionDeniedError: 'NotAllowedError',\n PermissionDismissedError: 'NotAllowedError',\n InvalidStateError: 'NotAllowedError',\n DevicesNotFoundError: 'NotFoundError',\n ConstraintNotSatisfiedError: 'OverconstrainedError',\n TrackStartError: 'NotReadableError',\n MediaDeviceFailedDueToShutdown: 'NotAllowedError',\n MediaDeviceKillSwitchOn: 'NotAllowedError',\n TabCaptureError: 'AbortError',\n ScreenCaptureError: 'AbortError',\n DeviceCaptureError: 'AbortError'\n }[e.name] || e.name,\n message: e.message,\n constraint: e.constraint || e.constraintName,\n toString() {\n return this.name + (this.message && ': ') + this.message;\n }\n };\n };\n\n const getUserMedia_ = function(constraints, onSuccess, onError) {\n shimConstraints_(constraints, c => {\n navigator.webkitGetUserMedia(c, onSuccess, e => {\n if (onError) {\n onError(shimError_(e));\n }\n });\n });\n };\n navigator.getUserMedia = getUserMedia_.bind(navigator);\n\n // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia\n // function which returns a Promise, it does not accept spec-style\n // constraints.\n if (navigator.mediaDevices.getUserMedia) {\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(cs) {\n return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => {\n if (c.audio && !stream.getAudioTracks().length ||\n c.video && !stream.getVideoTracks().length) {\n stream.getTracks().forEach(track => {\n track.stop();\n });\n throw new DOMException('', 'NotFoundError');\n }\n return stream;\n }, e => Promise.reject(shimError_(e))));\n };\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nexport function shimGetDisplayMedia(window, getSourceId) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n // getSourceId is a function that returns a promise resolving with\n // the sourceId of the screen/window/tab to be shared.\n if (typeof getSourceId !== 'function') {\n console.error('shimGetDisplayMedia: getSourceId argument is not ' +\n 'a function');\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n return getSourceId(constraints)\n .then(sourceId => {\n const widthSpecified = constraints.video && constraints.video.width;\n const heightSpecified = constraints.video &&\n constraints.video.height;\n const frameRateSpecified = constraints.video &&\n constraints.video.frameRate;\n constraints.video = {\n mandatory: {\n chromeMediaSource: 'desktop',\n chromeMediaSourceId: sourceId,\n maxFrameRate: frameRateSpecified || 3\n }\n };\n if (widthSpecified) {\n constraints.video.mandatory.maxWidth = widthSpecified;\n }\n if (heightSpecified) {\n constraints.video.mandatory.maxHeight = heightSpecified;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n });\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimMediaStream(window) {\n window.MediaStream = window.MediaStream || window.webkitMediaStream;\n}\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in\n window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', {\n get() {\n return this._ontrack;\n },\n set(f) {\n if (this._ontrack) {\n this.removeEventListener('track', this._ontrack);\n }\n this.addEventListener('track', this._ontrack = f);\n },\n enumerable: true,\n configurable: true\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n if (!this._ontrackpoly) {\n this._ontrackpoly = (e) => {\n // onaddstream does not fire when a track is added to an existing\n // stream. But stream.onaddtrack is implemented so we use that.\n e.stream.addEventListener('addtrack', te => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === te.track.id);\n } else {\n receiver = {track: te.track};\n }\n\n const event = new Event('track');\n event.track = te.track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n e.stream.getTracks().forEach(track => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === track.id);\n } else {\n receiver = {track};\n }\n const event = new Event('track');\n event.track = track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n };\n this.addEventListener('addstream', this._ontrackpoly);\n }\n return origSetRemoteDescription.apply(this, arguments);\n };\n } else {\n // even if RTCRtpTransceiver is in window, it is only used and\n // emitted in unified-plan. Unfortunately this means we need\n // to unconditionally wrap the event.\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n if (!e.transceiver) {\n Object.defineProperty(e, 'transceiver',\n {value: {receiver: e.receiver}});\n }\n return e;\n });\n }\n}\n\nexport function shimGetSendersWithDtmf(window) {\n // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack.\n if (typeof window === 'object' && window.RTCPeerConnection &&\n !('getSenders' in window.RTCPeerConnection.prototype) &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype) {\n const shimSenderWithDtmf = function(pc, track) {\n return {\n track,\n get dtmf() {\n if (this._dtmf === undefined) {\n if (track.kind === 'audio') {\n this._dtmf = pc.createDTMFSender(track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n },\n _pc: pc\n };\n };\n\n // augment addTrack when getSenders is not available.\n if (!window.RTCPeerConnection.prototype.getSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n this._senders = this._senders || [];\n return this._senders.slice(); // return a copy of the internal state.\n };\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n let sender = origAddTrack.apply(this, arguments);\n if (!sender) {\n sender = shimSenderWithDtmf(this, track);\n this._senders.push(sender);\n }\n return sender;\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n origRemoveTrack.apply(this, arguments);\n const idx = this._senders.indexOf(sender);\n if (idx !== -1) {\n this._senders.splice(idx, 1);\n }\n };\n }\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._senders = this._senders || [];\n origAddStream.apply(this, [stream]);\n stream.getTracks().forEach(track => {\n this._senders.push(shimSenderWithDtmf(this, track));\n });\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._senders = this._senders || [];\n origRemoveStream.apply(this, [stream]);\n\n stream.getTracks().forEach(track => {\n const sender = this._senders.find(s => s.track === track);\n if (sender) { // remove sender\n this._senders.splice(this._senders.indexOf(sender), 1);\n }\n });\n };\n } else if (typeof window === 'object' && window.RTCPeerConnection &&\n 'getSenders' in window.RTCPeerConnection.prototype &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype &&\n window.RTCRtpSender &&\n !('dtmf' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = this._pc.createDTMFSender(this.track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n}\n\nexport function shimGetStats(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n\n // If selector is a function then we are in the old style stats so just\n // pass back the original getStats format to avoid breaking old users.\n if (arguments.length > 0 && typeof selector === 'function') {\n return origGetStats.apply(this, arguments);\n }\n\n // When spec-style getStats is supported, return those when called with\n // either no arguments or the selector argument is null.\n if (origGetStats.length === 0 && (arguments.length === 0 ||\n typeof selector !== 'function')) {\n return origGetStats.apply(this, []);\n }\n\n const fixChromeStats_ = function(response) {\n const standardReport = {};\n const reports = response.result();\n reports.forEach(report => {\n const standardStats = {\n id: report.id,\n timestamp: report.timestamp,\n type: {\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[report.type] || report.type\n };\n report.names().forEach(name => {\n standardStats[name] = report.stat(name);\n });\n standardReport[standardStats.id] = standardStats;\n });\n\n return standardReport;\n };\n\n // shim getStats with maplike support\n const makeMapStats = function(stats) {\n return new Map(Object.keys(stats).map(key => [key, stats[key]]));\n };\n\n if (arguments.length >= 2) {\n const successCallbackWrapper_ = function(response) {\n onSucc(makeMapStats(fixChromeStats_(response)));\n };\n\n return origGetStats.apply(this, [successCallbackWrapper_,\n selector]);\n }\n\n // promise-support\n return new Promise((resolve, reject) => {\n origGetStats.apply(this, [\n function(response) {\n resolve(makeMapStats(fixChromeStats_(response)));\n }, reject]);\n }).then(onSucc, onErr);\n };\n}\n\nexport function shimSenderReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender && window.RTCRtpReceiver)) {\n return;\n }\n\n // shim sender stats.\n if (!('getStats' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n const sender = this;\n return this._pc.getStats().then(result =>\n /* Note: this will include stats of all senders that\n * send a track with the same id as sender.track as\n * it is not possible to identify the RTCRtpSender.\n */\n utils.filterStats(result, sender.track, true));\n };\n }\n\n // shim receiver stats.\n if (!('getStats' in window.RTCRtpReceiver.prototype)) {\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers =\n function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n const receiver = this;\n return this._pc.getStats().then(result =>\n utils.filterStats(result, receiver.track, false));\n };\n }\n\n if (!('getStats' in window.RTCRtpSender.prototype &&\n 'getStats' in window.RTCRtpReceiver.prototype)) {\n return;\n }\n\n // shim RTCPeerConnection.getStats(track).\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n if (arguments.length > 0 &&\n arguments[0] instanceof window.MediaStreamTrack) {\n const track = arguments[0];\n let sender;\n let receiver;\n let err;\n this.getSenders().forEach(s => {\n if (s.track === track) {\n if (sender) {\n err = true;\n } else {\n sender = s;\n }\n }\n });\n this.getReceivers().forEach(r => {\n if (r.track === track) {\n if (receiver) {\n err = true;\n } else {\n receiver = r;\n }\n }\n return r.track === track;\n });\n if (err || (sender && receiver)) {\n return Promise.reject(new DOMException(\n 'There are more than one sender or receiver for the track.',\n 'InvalidAccessError'));\n } else if (sender) {\n return sender.getStats();\n } else if (receiver) {\n return receiver.getStats();\n }\n return Promise.reject(new DOMException(\n 'There is no sender or receiver for the track.',\n 'InvalidAccessError'));\n }\n return origGetStats.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrackWithNative(window) {\n // shim addTrack/removeTrack with native variants in order to make\n // the interactions with legacy getLocalStreams behave as in other browsers.\n // Keeps a mapping stream.id => [stream, rtpsenders...]\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n return Object.keys(this._shimmedLocalStreams)\n .map(streamId => this._shimmedLocalStreams[streamId][0]);\n };\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (!stream) {\n return origAddTrack.apply(this, arguments);\n }\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n const sender = origAddTrack.apply(this, arguments);\n if (!this._shimmedLocalStreams[stream.id]) {\n this._shimmedLocalStreams[stream.id] = [stream, sender];\n } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) {\n this._shimmedLocalStreams[stream.id].push(sender);\n }\n return sender;\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n const existingSenders = this.getSenders();\n origAddStream.apply(this, arguments);\n const newSenders = this.getSenders()\n .filter(newSender => existingSenders.indexOf(newSender) === -1);\n this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n delete this._shimmedLocalStreams[stream.id];\n return origRemoveStream.apply(this, arguments);\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n if (sender) {\n Object.keys(this._shimmedLocalStreams).forEach(streamId => {\n const idx = this._shimmedLocalStreams[streamId].indexOf(sender);\n if (idx !== -1) {\n this._shimmedLocalStreams[streamId].splice(idx, 1);\n }\n if (this._shimmedLocalStreams[streamId].length === 1) {\n delete this._shimmedLocalStreams[streamId];\n }\n });\n }\n return origRemoveTrack.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrack(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // shim addTrack and removeTrack.\n if (window.RTCPeerConnection.prototype.addTrack &&\n browserDetails.version >= 65) {\n return shimAddTrackRemoveTrackWithNative(window);\n }\n\n // also shim pc.getLocalStreams when addTrack is shimmed\n // to return the original streams.\n const origGetLocalStreams = window.RTCPeerConnection.prototype\n .getLocalStreams;\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n const nativeStreams = origGetLocalStreams.apply(this);\n this._reverseStreams = this._reverseStreams || {};\n return nativeStreams.map(stream => this._reverseStreams[stream.id]);\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n // Add identity mapping for consistency with addTrack.\n // Unless this is being used with a stream from addTrack.\n if (!this._reverseStreams[stream.id]) {\n const newStream = new window.MediaStream(stream.getTracks());\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n stream = newStream;\n }\n origAddStream.apply(this, [stream]);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]);\n delete this._reverseStreams[(this._streams[stream.id] ?\n this._streams[stream.id].id : stream.id)];\n delete this._streams[stream.id];\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n const streams = [].slice.call(arguments, 1);\n if (streams.length !== 1 ||\n !streams[0].getTracks().find(t => t === track)) {\n // this is not fully correct but all we can manage without\n // [[associated MediaStreams]] internal slot.\n throw new DOMException(\n 'The adapter.js addTrack polyfill only supports a single ' +\n ' stream which is associated with the specified track.',\n 'NotSupportedError');\n }\n\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n const oldStream = this._streams[stream.id];\n if (oldStream) {\n // this is using odd Chrome behaviour, use with caution:\n // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815\n // Note: we rely on the high-level addTrack/dtmf shim to\n // create the sender with a dtmf sender.\n oldStream.addTrack(track);\n\n // Trigger ONN async.\n Promise.resolve().then(() => {\n this.dispatchEvent(new Event('negotiationneeded'));\n });\n } else {\n const newStream = new window.MediaStream([track]);\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n this.addStream(newStream);\n }\n return this.getSenders().find(s => s.track === track);\n };\n\n // replace the internal stream id with the external one and\n // vice versa.\n function replaceInternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(internalStream.id, 'g'),\n externalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n function replaceExternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(externalStream.id, 'g'),\n internalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n ['createOffer', 'createAnswer'].forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n const args = arguments;\n const isLegacyCall = arguments.length &&\n typeof arguments[0] === 'function';\n if (isLegacyCall) {\n return nativeMethod.apply(this, [\n (description) => {\n const desc = replaceInternalStreamId(this, description);\n args[0].apply(null, [desc]);\n },\n (err) => {\n if (args[1]) {\n args[1].apply(null, err);\n }\n }, arguments[2]\n ]);\n }\n return nativeMethod.apply(this, arguments)\n .then(description => replaceInternalStreamId(this, description));\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n\n const origSetLocalDescription =\n window.RTCPeerConnection.prototype.setLocalDescription;\n window.RTCPeerConnection.prototype.setLocalDescription =\n function setLocalDescription() {\n if (!arguments.length || !arguments[0].type) {\n return origSetLocalDescription.apply(this, arguments);\n }\n arguments[0] = replaceExternalStreamId(this, arguments[0]);\n return origSetLocalDescription.apply(this, arguments);\n };\n\n // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier\n\n const origLocalDescription = Object.getOwnPropertyDescriptor(\n window.RTCPeerConnection.prototype, 'localDescription');\n Object.defineProperty(window.RTCPeerConnection.prototype,\n 'localDescription', {\n get() {\n const description = origLocalDescription.get.apply(this);\n if (description.type === '') {\n return description;\n }\n return replaceInternalStreamId(this, description);\n }\n });\n\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n // We can not yet check for sender instanceof RTCRtpSender\n // since we shim RTPSender. So we check if sender._pc is set.\n if (!sender._pc) {\n throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.', 'TypeError');\n }\n const isLocal = sender._pc === this;\n if (!isLocal) {\n throw new DOMException('Sender was not created by this connection.',\n 'InvalidAccessError');\n }\n\n // Search for the native stream the senders track belongs to.\n this._streams = this._streams || {};\n let stream;\n Object.keys(this._streams).forEach(streamid => {\n const hasTrack = this._streams[streamid].getTracks()\n .find(track => sender.track === track);\n if (hasTrack) {\n stream = this._streams[streamid];\n }\n });\n\n if (stream) {\n if (stream.getTracks().length === 1) {\n // if this is the last track of the stream, remove the stream. This\n // takes care of any shimmed _senders.\n this.removeStream(this._reverseStreams[stream.id]);\n } else {\n // relying on the same odd chrome behaviour as above.\n stream.removeTrack(sender.track);\n }\n this.dispatchEvent(new Event('negotiationneeded'));\n }\n };\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.webkitRTCPeerConnection;\n }\n if (!window.RTCPeerConnection) {\n return;\n }\n\n // shim implicit creation of RTCSessionDescription/RTCIceCandidate\n if (browserDetails.version < 53) {\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n}\n\n// Attempt to fix ONN in plan-b mode.\nexport function fixNegotiationNeeded(window, browserDetails) {\n utils.wrapPeerConnectionEvent(window, 'negotiationneeded', e => {\n const pc = e.target;\n if (browserDetails.version < 72 || (pc.getConfiguration &&\n pc.getConfiguration().sdpSemantics === 'plan-b')) {\n if (pc.signalingState !== 'stable') {\n return;\n }\n }\n return e;\n });\n}\n","/*\n * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nexport function filterIceServers(iceServers, edgeVersion) {\n let hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(server => {\n if (server && (server.urls || server.url)) {\n let urls = server.urls || server.url;\n if (server.url && !server.urls) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n }\n const isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(url => {\n // filter STUN unconditionally.\n if (url.indexOf('stun:') === 0) {\n return false;\n }\n\n const validTurn = url.startsWith('turn') &&\n !url.startsWith('turn:[') &&\n url.includes('transport=udp');\n if (validTurn && !hasTurn) {\n hasTurn = true;\n return true;\n }\n return validTurn && !hasTurn;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n","/* eslint-env node */\n'use strict';\n\n// SDP helpers.\nvar SDPUtils = {};\n\n// Generate an alphanumeric identifier for cname or mids.\n// TODO: use UUIDs instead? https://gist.github.com/jed/982883\nSDPUtils.generateIdentifier = function() {\n return Math.random().toString(36).substr(2, 10);\n};\n\n// The RTCP CNAME used by all peerconnections from the same JS.\nSDPUtils.localCName = SDPUtils.generateIdentifier();\n\n// Splits SDP into lines, dealing with both CRLF and LF.\nSDPUtils.splitLines = function(blob) {\n return blob.trim().split('\\n').map(function(line) {\n return line.trim();\n });\n};\n// Splits SDP into sessionpart and mediasections. Ensures CRLF.\nSDPUtils.splitSections = function(blob) {\n var parts = blob.split('\\nm=');\n return parts.map(function(part, index) {\n return (index > 0 ? 'm=' + part : part).trim() + '\\r\\n';\n });\n};\n\n// returns the session description.\nSDPUtils.getDescription = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n return sections && sections[0];\n};\n\n// returns the individual media sections.\nSDPUtils.getMediaSections = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n sections.shift();\n return sections;\n};\n\n// Returns lines that start with a certain prefix.\nSDPUtils.matchPrefix = function(blob, prefix) {\n return SDPUtils.splitLines(blob).filter(function(line) {\n return line.indexOf(prefix) === 0;\n });\n};\n\n// Parses an ICE candidate line. Sample input:\n// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8\n// rport 55996\"\nSDPUtils.parseCandidate = function(line) {\n var parts;\n // Parse both variants.\n if (line.indexOf('a=candidate:') === 0) {\n parts = line.substring(12).split(' ');\n } else {\n parts = line.substring(10).split(' ');\n }\n\n var candidate = {\n foundation: parts[0],\n component: parseInt(parts[1], 10),\n protocol: parts[2].toLowerCase(),\n priority: parseInt(parts[3], 10),\n ip: parts[4],\n address: parts[4], // address is an alias for ip.\n port: parseInt(parts[5], 10),\n // skip parts[6] == 'typ'\n type: parts[7]\n };\n\n for (var i = 8; i < parts.length; i += 2) {\n switch (parts[i]) {\n case 'raddr':\n candidate.relatedAddress = parts[i + 1];\n break;\n case 'rport':\n candidate.relatedPort = parseInt(parts[i + 1], 10);\n break;\n case 'tcptype':\n candidate.tcpType = parts[i + 1];\n break;\n case 'ufrag':\n candidate.ufrag = parts[i + 1]; // for backward compability.\n candidate.usernameFragment = parts[i + 1];\n break;\n default: // extension handling, in particular ufrag\n candidate[parts[i]] = parts[i + 1];\n break;\n }\n }\n return candidate;\n};\n\n// Translates a candidate object into SDP candidate attribute.\nSDPUtils.writeCandidate = function(candidate) {\n var sdp = [];\n sdp.push(candidate.foundation);\n sdp.push(candidate.component);\n sdp.push(candidate.protocol.toUpperCase());\n sdp.push(candidate.priority);\n sdp.push(candidate.address || candidate.ip);\n sdp.push(candidate.port);\n\n var type = candidate.type;\n sdp.push('typ');\n sdp.push(type);\n if (type !== 'host' && candidate.relatedAddress &&\n candidate.relatedPort) {\n sdp.push('raddr');\n sdp.push(candidate.relatedAddress);\n sdp.push('rport');\n sdp.push(candidate.relatedPort);\n }\n if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') {\n sdp.push('tcptype');\n sdp.push(candidate.tcpType);\n }\n if (candidate.usernameFragment || candidate.ufrag) {\n sdp.push('ufrag');\n sdp.push(candidate.usernameFragment || candidate.ufrag);\n }\n return 'candidate:' + sdp.join(' ');\n};\n\n// Parses an ice-options line, returns an array of option tags.\n// a=ice-options:foo bar\nSDPUtils.parseIceOptions = function(line) {\n return line.substr(14).split(' ');\n};\n\n// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:\n// a=rtpmap:111 opus/48000/2\nSDPUtils.parseRtpMap = function(line) {\n var parts = line.substr(9).split(' ');\n var parsed = {\n payloadType: parseInt(parts.shift(), 10) // was: id\n };\n\n parts = parts[0].split('/');\n\n parsed.name = parts[0];\n parsed.clockRate = parseInt(parts[1], 10); // was: clockrate\n parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;\n // legacy alias, got renamed back to channels in ORTC.\n parsed.numChannels = parsed.channels;\n return parsed;\n};\n\n// Generate an a=rtpmap line from RTCRtpCodecCapability or\n// RTCRtpCodecParameters.\nSDPUtils.writeRtpMap = function(codec) {\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n var channels = codec.channels || codec.numChannels || 1;\n return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate +\n (channels !== 1 ? '/' + channels : '') + '\\r\\n';\n};\n\n// Parses an a=extmap line (headerextension from RFC 5285). Sample input:\n// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset\nSDPUtils.parseExtmap = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n id: parseInt(parts[0], 10),\n direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv',\n uri: parts[1]\n };\n};\n\n// Generates a=extmap line from RTCRtpHeaderExtensionParameters or\n// RTCRtpHeaderExtension.\nSDPUtils.writeExtmap = function(headerExtension) {\n return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) +\n (headerExtension.direction && headerExtension.direction !== 'sendrecv'\n ? '/' + headerExtension.direction\n : '') +\n ' ' + headerExtension.uri + '\\r\\n';\n};\n\n// Parses an ftmp line, returns dictionary. Sample input:\n// a=fmtp:96 vbr=on;cng=on\n// Also deals with vbr=on; cng=on\nSDPUtils.parseFmtp = function(line) {\n var parsed = {};\n var kv;\n var parts = line.substr(line.indexOf(' ') + 1).split(';');\n for (var j = 0; j < parts.length; j++) {\n kv = parts[j].trim().split('=');\n parsed[kv[0].trim()] = kv[1];\n }\n return parsed;\n};\n\n// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeFmtp = function(codec) {\n var line = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.parameters && Object.keys(codec.parameters).length) {\n var params = [];\n Object.keys(codec.parameters).forEach(function(param) {\n if (codec.parameters[param]) {\n params.push(param + '=' + codec.parameters[param]);\n } else {\n params.push(param);\n }\n });\n line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\\r\\n';\n }\n return line;\n};\n\n// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:\n// a=rtcp-fb:98 nack rpsi\nSDPUtils.parseRtcpFb = function(line) {\n var parts = line.substr(line.indexOf(' ') + 1).split(' ');\n return {\n type: parts.shift(),\n parameter: parts.join(' ')\n };\n};\n// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeRtcpFb = function(codec) {\n var lines = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.rtcpFeedback && codec.rtcpFeedback.length) {\n // FIXME: special handling for trr-int?\n codec.rtcpFeedback.forEach(function(fb) {\n lines += 'a=rtcp-fb:' + pt + ' ' + fb.type +\n (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') +\n '\\r\\n';\n });\n }\n return lines;\n};\n\n// Parses an RFC 5576 ssrc media attribute. Sample input:\n// a=ssrc:3735928559 cname:something\nSDPUtils.parseSsrcMedia = function(line) {\n var sp = line.indexOf(' ');\n var parts = {\n ssrc: parseInt(line.substr(7, sp - 7), 10)\n };\n var colon = line.indexOf(':', sp);\n if (colon > -1) {\n parts.attribute = line.substr(sp + 1, colon - sp - 1);\n parts.value = line.substr(colon + 1);\n } else {\n parts.attribute = line.substr(sp + 1);\n }\n return parts;\n};\n\nSDPUtils.parseSsrcGroup = function(line) {\n var parts = line.substr(13).split(' ');\n return {\n semantics: parts.shift(),\n ssrcs: parts.map(function(ssrc) {\n return parseInt(ssrc, 10);\n })\n };\n};\n\n// Extracts the MID (RFC 5888) from a media section.\n// returns the MID or undefined if no mid line was found.\nSDPUtils.getMid = function(mediaSection) {\n var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0];\n if (mid) {\n return mid.substr(6);\n }\n};\n\nSDPUtils.parseFingerprint = function(line) {\n var parts = line.substr(14).split(' ');\n return {\n algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.\n value: parts[1]\n };\n};\n\n// Extracts DTLS parameters from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the fingerprint line as input. See also getIceParameters.\nSDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=fingerprint:');\n // Note: a=setup line is ignored since we use the 'auto' role.\n // Note2: 'algorithm' is not case sensitive except in Edge.\n return {\n role: 'auto',\n fingerprints: lines.map(SDPUtils.parseFingerprint)\n };\n};\n\n// Serializes DTLS parameters to SDP.\nSDPUtils.writeDtlsParameters = function(params, setupType) {\n var sdp = 'a=setup:' + setupType + '\\r\\n';\n params.fingerprints.forEach(function(fp) {\n sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\\r\\n';\n });\n return sdp;\n};\n\n// Parses a=crypto lines into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members\nSDPUtils.parseCryptoLine = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n tag: parseInt(parts[0], 10),\n cryptoSuite: parts[1],\n keyParams: parts[2],\n sessionParams: parts.slice(3),\n };\n};\n\nSDPUtils.writeCryptoLine = function(parameters) {\n return 'a=crypto:' + parameters.tag + ' ' +\n parameters.cryptoSuite + ' ' +\n (typeof parameters.keyParams === 'object'\n ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)\n : parameters.keyParams) +\n (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') +\n '\\r\\n';\n};\n\n// Parses the crypto key parameters into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*\nSDPUtils.parseCryptoKeyParams = function(keyParams) {\n if (keyParams.indexOf('inline:') !== 0) {\n return null;\n }\n var parts = keyParams.substr(7).split('|');\n return {\n keyMethod: 'inline',\n keySalt: parts[0],\n lifeTime: parts[1],\n mkiValue: parts[2] ? parts[2].split(':')[0] : undefined,\n mkiLength: parts[2] ? parts[2].split(':')[1] : undefined,\n };\n};\n\nSDPUtils.writeCryptoKeyParams = function(keyParams) {\n return keyParams.keyMethod + ':'\n + keyParams.keySalt +\n (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') +\n (keyParams.mkiValue && keyParams.mkiLength\n ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength\n : '');\n};\n\n// Extracts all SDES paramters.\nSDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=crypto:');\n return lines.map(SDPUtils.parseCryptoLine);\n};\n\n// Parses ICE information from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the ice-ufrag and ice-pwd lines as input.\nSDPUtils.getIceParameters = function(mediaSection, sessionpart) {\n var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-ufrag:')[0];\n var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-pwd:')[0];\n if (!(ufrag && pwd)) {\n return null;\n }\n return {\n usernameFragment: ufrag.substr(12),\n password: pwd.substr(10),\n };\n};\n\n// Serializes ICE parameters to SDP.\nSDPUtils.writeIceParameters = function(params) {\n return 'a=ice-ufrag:' + params.usernameFragment + '\\r\\n' +\n 'a=ice-pwd:' + params.password + '\\r\\n';\n};\n\n// Parses the SDP media section and returns RTCRtpParameters.\nSDPUtils.parseRtpParameters = function(mediaSection) {\n var description = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: [],\n rtcp: []\n };\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..]\n var pt = mline[i];\n var rtpmapline = SDPUtils.matchPrefix(\n mediaSection, 'a=rtpmap:' + pt + ' ')[0];\n if (rtpmapline) {\n var codec = SDPUtils.parseRtpMap(rtpmapline);\n var fmtps = SDPUtils.matchPrefix(\n mediaSection, 'a=fmtp:' + pt + ' ');\n // Only the first a=fmtp: is considered.\n codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};\n codec.rtcpFeedback = SDPUtils.matchPrefix(\n mediaSection, 'a=rtcp-fb:' + pt + ' ')\n .map(SDPUtils.parseRtcpFb);\n description.codecs.push(codec);\n // parse FEC mechanisms from rtpmap lines.\n switch (codec.name.toUpperCase()) {\n case 'RED':\n case 'ULPFEC':\n description.fecMechanisms.push(codec.name.toUpperCase());\n break;\n default: // only RED and ULPFEC are recognized as FEC mechanisms.\n break;\n }\n }\n }\n SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) {\n description.headerExtensions.push(SDPUtils.parseExtmap(line));\n });\n // FIXME: parse rtcp.\n return description;\n};\n\n// Generates parts of the SDP media section describing the capabilities /\n// parameters.\nSDPUtils.writeRtpDescription = function(kind, caps) {\n var sdp = '';\n\n // Build the mline.\n sdp += 'm=' + kind + ' ';\n sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs.\n sdp += ' UDP/TLS/RTP/SAVPF ';\n sdp += caps.codecs.map(function(codec) {\n if (codec.preferredPayloadType !== undefined) {\n return codec.preferredPayloadType;\n }\n return codec.payloadType;\n }).join(' ') + '\\r\\n';\n\n sdp += 'c=IN IP4 0.0.0.0\\r\\n';\n sdp += 'a=rtcp:9 IN IP4 0.0.0.0\\r\\n';\n\n // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.\n caps.codecs.forEach(function(codec) {\n sdp += SDPUtils.writeRtpMap(codec);\n sdp += SDPUtils.writeFmtp(codec);\n sdp += SDPUtils.writeRtcpFb(codec);\n });\n var maxptime = 0;\n caps.codecs.forEach(function(codec) {\n if (codec.maxptime > maxptime) {\n maxptime = codec.maxptime;\n }\n });\n if (maxptime > 0) {\n sdp += 'a=maxptime:' + maxptime + '\\r\\n';\n }\n sdp += 'a=rtcp-mux\\r\\n';\n\n if (caps.headerExtensions) {\n caps.headerExtensions.forEach(function(extension) {\n sdp += SDPUtils.writeExtmap(extension);\n });\n }\n // FIXME: write fecMechanisms.\n return sdp;\n};\n\n// Parses the SDP media section and returns an array of\n// RTCRtpEncodingParameters.\nSDPUtils.parseRtpEncodingParameters = function(mediaSection) {\n var encodingParameters = [];\n var description = SDPUtils.parseRtpParameters(mediaSection);\n var hasRed = description.fecMechanisms.indexOf('RED') !== -1;\n var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1;\n\n // filter a=ssrc:... cname:, ignore PlanB-msid\n var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(parts) {\n return parts.attribute === 'cname';\n });\n var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;\n var secondarySsrc;\n\n var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID')\n .map(function(line) {\n var parts = line.substr(17).split(' ');\n return parts.map(function(part) {\n return parseInt(part, 10);\n });\n });\n if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {\n secondarySsrc = flows[0][1];\n }\n\n description.codecs.forEach(function(codec) {\n if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) {\n var encParam = {\n ssrc: primarySsrc,\n codecPayloadType: parseInt(codec.parameters.apt, 10)\n };\n if (primarySsrc && secondarySsrc) {\n encParam.rtx = {ssrc: secondarySsrc};\n }\n encodingParameters.push(encParam);\n if (hasRed) {\n encParam = JSON.parse(JSON.stringify(encParam));\n encParam.fec = {\n ssrc: primarySsrc,\n mechanism: hasUlpfec ? 'red+ulpfec' : 'red'\n };\n encodingParameters.push(encParam);\n }\n }\n });\n if (encodingParameters.length === 0 && primarySsrc) {\n encodingParameters.push({\n ssrc: primarySsrc\n });\n }\n\n // we support both b=AS and b=TIAS but interpret AS as TIAS.\n var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b=');\n if (bandwidth.length) {\n if (bandwidth[0].indexOf('b=TIAS:') === 0) {\n bandwidth = parseInt(bandwidth[0].substr(7), 10);\n } else if (bandwidth[0].indexOf('b=AS:') === 0) {\n // use formula from JSEP to convert b=AS to TIAS value.\n bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95\n - (50 * 40 * 8);\n } else {\n bandwidth = undefined;\n }\n encodingParameters.forEach(function(params) {\n params.maxBitrate = bandwidth;\n });\n }\n return encodingParameters;\n};\n\n// parses http://draft.ortc.org/#rtcrtcpparameters*\nSDPUtils.parseRtcpParameters = function(mediaSection) {\n var rtcpParameters = {};\n\n // Gets the first SSRC. Note tha with RTX there might be multiple\n // SSRCs.\n var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(obj) {\n return obj.attribute === 'cname';\n })[0];\n if (remoteSsrc) {\n rtcpParameters.cname = remoteSsrc.value;\n rtcpParameters.ssrc = remoteSsrc.ssrc;\n }\n\n // Edge uses the compound attribute instead of reducedSize\n // compound is !reducedSize\n var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize');\n rtcpParameters.reducedSize = rsize.length > 0;\n rtcpParameters.compound = rsize.length === 0;\n\n // parses the rtcp-mux attrіbute.\n // Note that Edge does not support unmuxed RTCP.\n var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux');\n rtcpParameters.mux = mux.length > 0;\n\n return rtcpParameters;\n};\n\n// parses either a=msid: or a=ssrc:... msid lines and returns\n// the id of the MediaStream and MediaStreamTrack.\nSDPUtils.parseMsid = function(mediaSection) {\n var parts;\n var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:');\n if (spec.length === 1) {\n parts = spec[0].substr(7).split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(msidParts) {\n return msidParts.attribute === 'msid';\n });\n if (planB.length > 0) {\n parts = planB[0].value.split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n};\n\n// SCTP\n// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back\n// to draft-ietf-mmusic-sctp-sdp-05\nSDPUtils.parseSctpDescription = function(mediaSection) {\n var mline = SDPUtils.parseMLine(mediaSection);\n var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:');\n var maxMessageSize;\n if (maxSizeLine.length > 0) {\n maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);\n }\n if (isNaN(maxMessageSize)) {\n maxMessageSize = 65536;\n }\n var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:');\n if (sctpPort.length > 0) {\n return {\n port: parseInt(sctpPort[0].substr(12), 10),\n protocol: mline.fmt,\n maxMessageSize: maxMessageSize\n };\n }\n var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:');\n if (sctpMapLines.length > 0) {\n var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0]\n .substr(10)\n .split(' ');\n return {\n port: parseInt(parts[0], 10),\n protocol: parts[1],\n maxMessageSize: maxMessageSize\n };\n }\n};\n\n// SCTP\n// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers\n// support by now receiving in this format, unless we originally parsed\n// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line\n// protocol of DTLS/SCTP -- without UDP/ or TCP/)\nSDPUtils.writeSctpDescription = function(media, sctp) {\n var output = [];\n if (media.protocol !== 'DTLS/SCTP') {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctp-port:' + sctp.port + '\\r\\n'\n ];\n } else {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\\r\\n'\n ];\n }\n if (sctp.maxMessageSize !== undefined) {\n output.push('a=max-message-size:' + sctp.maxMessageSize + '\\r\\n');\n }\n return output.join('');\n};\n\n// Generate a session ID for SDP.\n// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1\n// recommends using a cryptographically random +ve 64-bit value\n// but right now this should be acceptable and within the right range\nSDPUtils.generateSessionId = function() {\n return Math.random().toString().substr(2, 21);\n};\n\n// Write boilder plate for start of SDP\n// sessId argument is optional - if not supplied it will\n// be generated randomly\n// sessVersion is optional and defaults to 2\n// sessUser is optional and defaults to 'thisisadapterortc'\nSDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {\n var sessionId;\n var version = sessVer !== undefined ? sessVer : 2;\n if (sessId) {\n sessionId = sessId;\n } else {\n sessionId = SDPUtils.generateSessionId();\n }\n var user = sessUser || 'thisisadapterortc';\n // FIXME: sess-id should be an NTP timestamp.\n return 'v=0\\r\\n' +\n 'o=' + user + ' ' + sessionId + ' ' + version +\n ' IN IP4 127.0.0.1\\r\\n' +\n 's=-\\r\\n' +\n 't=0 0\\r\\n';\n};\n\nSDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.direction) {\n sdp += 'a=' + transceiver.direction + '\\r\\n';\n } else if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n // spec.\n var msid = 'msid:' + stream.id + ' ' +\n transceiver.rtpSender.track.id + '\\r\\n';\n sdp += 'a=' + msid;\n\n // for Chrome.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n};\n\n// Gets the direction from the mediaSection or the sessionpart.\nSDPUtils.getDirection = function(mediaSection, sessionpart) {\n // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.\n var lines = SDPUtils.splitLines(mediaSection);\n for (var i = 0; i < lines.length; i++) {\n switch (lines[i]) {\n case 'a=sendrecv':\n case 'a=sendonly':\n case 'a=recvonly':\n case 'a=inactive':\n return lines[i].substr(2);\n default:\n // FIXME: What should happen here?\n }\n }\n if (sessionpart) {\n return SDPUtils.getDirection(sessionpart);\n }\n return 'sendrecv';\n};\n\nSDPUtils.getKind = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n return mline[0].substr(2);\n};\n\nSDPUtils.isRejected = function(mediaSection) {\n return mediaSection.split(' ', 2)[1] === '0';\n};\n\nSDPUtils.parseMLine = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var parts = lines[0].substr(2).split(' ');\n return {\n kind: parts[0],\n port: parseInt(parts[1], 10),\n protocol: parts[2],\n fmt: parts.slice(3).join(' ')\n };\n};\n\nSDPUtils.parseOLine = function(mediaSection) {\n var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0];\n var parts = line.substr(2).split(' ');\n return {\n username: parts[0],\n sessionId: parts[1],\n sessionVersion: parseInt(parts[2], 10),\n netType: parts[3],\n addressType: parts[4],\n address: parts[5]\n };\n};\n\n// a very naive interpretation of a valid SDP.\nSDPUtils.isValidSDP = function(blob) {\n if (typeof blob !== 'string' || blob.length === 0) {\n return false;\n }\n var lines = SDPUtils.splitLines(blob);\n for (var i = 0; i < lines.length; i++) {\n if (lines[i].length < 2 || lines[i].charAt(1) !== '=') {\n return false;\n }\n // TODO: check the modifier a bit more.\n }\n return true;\n};\n\n// Expose public methods.\nif (typeof module === 'object') {\n module.exports = SDPUtils;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nvar SDPUtils = require('sdp');\n\nfunction fixStatsType(stat) {\n return {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[stat.type] || stat.type;\n}\n\nfunction writeMediaSection(transceiver, caps, type, stream, dtlsRole) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : dtlsRole || 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n var trackId = transceiver.rtpSender._initialTrackId ||\n transceiver.rtpSender.track.id;\n transceiver.rtpSender._initialTrackId = trackId;\n // spec.\n var msid = 'msid:' + (stream ? stream.id : '-') + ' ' +\n trackId + '\\r\\n';\n sdp += 'a=' + msid;\n // for Chrome. Legacy should no longer be required.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n\n // RTX\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n}\n\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nfunction filterIceServers(iceServers, edgeVersion) {\n var hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(function(server) {\n if (server && (server.urls || server.url)) {\n var urls = server.urls || server.url;\n if (server.url && !server.urls) {\n console.warn('RTCIceServer.url is deprecated! Use urls instead.');\n }\n var isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(function(url) {\n var validTurn = url.indexOf('turn:') === 0 &&\n url.indexOf('transport=udp') !== -1 &&\n url.indexOf('turn:[') === -1 &&\n !hasTurn;\n\n if (validTurn) {\n hasTurn = true;\n return true;\n }\n return url.indexOf('stun:') === 0 && edgeVersion >= 14393 &&\n url.indexOf('?transport=udp') === -1;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n\n// Determines the intersection of local and remote capabilities.\nfunction getCommonCapabilities(localCapabilities, remoteCapabilities) {\n var commonCapabilities = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: []\n };\n\n var findCodecByPayloadType = function(pt, codecs) {\n pt = parseInt(pt, 10);\n for (var i = 0; i < codecs.length; i++) {\n if (codecs[i].payloadType === pt ||\n codecs[i].preferredPayloadType === pt) {\n return codecs[i];\n }\n }\n };\n\n var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) {\n var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs);\n var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs);\n return lCodec && rCodec &&\n lCodec.name.toLowerCase() === rCodec.name.toLowerCase();\n };\n\n localCapabilities.codecs.forEach(function(lCodec) {\n for (var i = 0; i < remoteCapabilities.codecs.length; i++) {\n var rCodec = remoteCapabilities.codecs[i];\n if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() &&\n lCodec.clockRate === rCodec.clockRate) {\n if (lCodec.name.toLowerCase() === 'rtx' &&\n lCodec.parameters && rCodec.parameters.apt) {\n // for RTX we need to find the local rtx that has a apt\n // which points to the same local codec as the remote one.\n if (!rtxCapabilityMatches(lCodec, rCodec,\n localCapabilities.codecs, remoteCapabilities.codecs)) {\n continue;\n }\n }\n rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy\n // number of channels is the highest common number of channels\n rCodec.numChannels = Math.min(lCodec.numChannels,\n rCodec.numChannels);\n // push rCodec so we reply with offerer payload type\n commonCapabilities.codecs.push(rCodec);\n\n // determine common feedback mechanisms\n rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) {\n for (var j = 0; j < lCodec.rtcpFeedback.length; j++) {\n if (lCodec.rtcpFeedback[j].type === fb.type &&\n lCodec.rtcpFeedback[j].parameter === fb.parameter) {\n return true;\n }\n }\n return false;\n });\n // FIXME: also need to determine .parameters\n // see https://github.com/openpeer/ortc/issues/569\n break;\n }\n }\n });\n\n localCapabilities.headerExtensions.forEach(function(lHeaderExtension) {\n for (var i = 0; i < remoteCapabilities.headerExtensions.length;\n i++) {\n var rHeaderExtension = remoteCapabilities.headerExtensions[i];\n if (lHeaderExtension.uri === rHeaderExtension.uri) {\n commonCapabilities.headerExtensions.push(rHeaderExtension);\n break;\n }\n }\n });\n\n // FIXME: fecMechanisms\n return commonCapabilities;\n}\n\n// is action=setLocalDescription with type allowed in signalingState\nfunction isActionAllowedInSignalingState(action, type, signalingState) {\n return {\n offer: {\n setLocalDescription: ['stable', 'have-local-offer'],\n setRemoteDescription: ['stable', 'have-remote-offer']\n },\n answer: {\n setLocalDescription: ['have-remote-offer', 'have-local-pranswer'],\n setRemoteDescription: ['have-local-offer', 'have-remote-pranswer']\n }\n }[type][action].indexOf(signalingState) !== -1;\n}\n\nfunction maybeAddCandidate(iceTransport, candidate) {\n // Edge's internal representation adds some fields therefore\n // not all fieldѕ are taken into account.\n var alreadyAdded = iceTransport.getRemoteCandidates()\n .find(function(remoteCandidate) {\n return candidate.foundation === remoteCandidate.foundation &&\n candidate.ip === remoteCandidate.ip &&\n candidate.port === remoteCandidate.port &&\n candidate.priority === remoteCandidate.priority &&\n candidate.protocol === remoteCandidate.protocol &&\n candidate.type === remoteCandidate.type;\n });\n if (!alreadyAdded) {\n iceTransport.addRemoteCandidate(candidate);\n }\n return !alreadyAdded;\n}\n\n\nfunction makeError(name, description) {\n var e = new Error(description);\n e.name = name;\n // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names\n e.code = {\n NotSupportedError: 9,\n InvalidStateError: 11,\n InvalidAccessError: 15,\n TypeError: undefined,\n OperationError: undefined\n }[name];\n return e;\n}\n\nmodule.exports = function(window, edgeVersion) {\n // https://w3c.github.io/mediacapture-main/#mediastream\n // Helper function to add the track to the stream and\n // dispatch the event ourselves.\n function addTrackToStreamAndFireEvent(track, stream) {\n stream.addTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack',\n {track: track}));\n }\n\n function removeTrackFromStreamAndFireEvent(track, stream) {\n stream.removeTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack',\n {track: track}));\n }\n\n function fireAddTrack(pc, track, receiver, streams) {\n var trackEvent = new Event('track');\n trackEvent.track = track;\n trackEvent.receiver = receiver;\n trackEvent.transceiver = {receiver: receiver};\n trackEvent.streams = streams;\n window.setTimeout(function() {\n pc._dispatchEvent('track', trackEvent);\n });\n }\n\n var RTCPeerConnection = function(config) {\n var pc = this;\n\n var _eventTarget = document.createDocumentFragment();\n ['addEventListener', 'removeEventListener', 'dispatchEvent']\n .forEach(function(method) {\n pc[method] = _eventTarget[method].bind(_eventTarget);\n });\n\n this.canTrickleIceCandidates = null;\n\n this.needNegotiation = false;\n\n this.localStreams = [];\n this.remoteStreams = [];\n\n this._localDescription = null;\n this._remoteDescription = null;\n\n this.signalingState = 'stable';\n this.iceConnectionState = 'new';\n this.connectionState = 'new';\n this.iceGatheringState = 'new';\n\n config = JSON.parse(JSON.stringify(config || {}));\n\n this.usingBundle = config.bundlePolicy === 'max-bundle';\n if (config.rtcpMuxPolicy === 'negotiate') {\n throw(makeError('NotSupportedError',\n 'rtcpMuxPolicy \\'negotiate\\' is not supported'));\n } else if (!config.rtcpMuxPolicy) {\n config.rtcpMuxPolicy = 'require';\n }\n\n switch (config.iceTransportPolicy) {\n case 'all':\n case 'relay':\n break;\n default:\n config.iceTransportPolicy = 'all';\n break;\n }\n\n switch (config.bundlePolicy) {\n case 'balanced':\n case 'max-compat':\n case 'max-bundle':\n break;\n default:\n config.bundlePolicy = 'balanced';\n break;\n }\n\n config.iceServers = filterIceServers(config.iceServers || [], edgeVersion);\n\n this._iceGatherers = [];\n if (config.iceCandidatePoolSize) {\n for (var i = config.iceCandidatePoolSize; i > 0; i--) {\n this._iceGatherers.push(new window.RTCIceGatherer({\n iceServers: config.iceServers,\n gatherPolicy: config.iceTransportPolicy\n }));\n }\n } else {\n config.iceCandidatePoolSize = 0;\n }\n\n this._config = config;\n\n // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ...\n // everything that is needed to describe a SDP m-line.\n this.transceivers = [];\n\n this._sdpSessionId = SDPUtils.generateSessionId();\n this._sdpSessionVersion = 0;\n\n this._dtlsRole = undefined; // role for a=setup to use in answers.\n\n this._isClosed = false;\n };\n\n Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', {\n configurable: true,\n get: function() {\n return this._localDescription;\n }\n });\n Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', {\n configurable: true,\n get: function() {\n return this._remoteDescription;\n }\n });\n\n // set up event handlers on prototype\n RTCPeerConnection.prototype.onicecandidate = null;\n RTCPeerConnection.prototype.onaddstream = null;\n RTCPeerConnection.prototype.ontrack = null;\n RTCPeerConnection.prototype.onremovestream = null;\n RTCPeerConnection.prototype.onsignalingstatechange = null;\n RTCPeerConnection.prototype.oniceconnectionstatechange = null;\n RTCPeerConnection.prototype.onconnectionstatechange = null;\n RTCPeerConnection.prototype.onicegatheringstatechange = null;\n RTCPeerConnection.prototype.onnegotiationneeded = null;\n RTCPeerConnection.prototype.ondatachannel = null;\n\n RTCPeerConnection.prototype._dispatchEvent = function(name, event) {\n if (this._isClosed) {\n return;\n }\n this.dispatchEvent(event);\n if (typeof this['on' + name] === 'function') {\n this['on' + name](event);\n }\n };\n\n RTCPeerConnection.prototype._emitGatheringStateChange = function() {\n var event = new Event('icegatheringstatechange');\n this._dispatchEvent('icegatheringstatechange', event);\n };\n\n RTCPeerConnection.prototype.getConfiguration = function() {\n return this._config;\n };\n\n RTCPeerConnection.prototype.getLocalStreams = function() {\n return this.localStreams;\n };\n\n RTCPeerConnection.prototype.getRemoteStreams = function() {\n return this.remoteStreams;\n };\n\n // internal helper to create a transceiver object.\n // (which is not yet the same as the WebRTC 1.0 transceiver)\n RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) {\n var hasBundleTransport = this.transceivers.length > 0;\n var transceiver = {\n track: null,\n iceGatherer: null,\n iceTransport: null,\n dtlsTransport: null,\n localCapabilities: null,\n remoteCapabilities: null,\n rtpSender: null,\n rtpReceiver: null,\n kind: kind,\n mid: null,\n sendEncodingParameters: null,\n recvEncodingParameters: null,\n stream: null,\n associatedRemoteMediaStreams: [],\n wantReceive: true\n };\n if (this.usingBundle && hasBundleTransport) {\n transceiver.iceTransport = this.transceivers[0].iceTransport;\n transceiver.dtlsTransport = this.transceivers[0].dtlsTransport;\n } else {\n var transports = this._createIceAndDtlsTransports();\n transceiver.iceTransport = transports.iceTransport;\n transceiver.dtlsTransport = transports.dtlsTransport;\n }\n if (!doNotAdd) {\n this.transceivers.push(transceiver);\n }\n return transceiver;\n };\n\n RTCPeerConnection.prototype.addTrack = function(track, stream) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call addTrack on a closed peerconnection.');\n }\n\n var alreadyExists = this.transceivers.find(function(s) {\n return s.track === track;\n });\n\n if (alreadyExists) {\n throw makeError('InvalidAccessError', 'Track already exists.');\n }\n\n var transceiver;\n for (var i = 0; i < this.transceivers.length; i++) {\n if (!this.transceivers[i].track &&\n this.transceivers[i].kind === track.kind) {\n transceiver = this.transceivers[i];\n }\n }\n if (!transceiver) {\n transceiver = this._createTransceiver(track.kind);\n }\n\n this._maybeFireNegotiationNeeded();\n\n if (this.localStreams.indexOf(stream) === -1) {\n this.localStreams.push(stream);\n }\n\n transceiver.track = track;\n transceiver.stream = stream;\n transceiver.rtpSender = new window.RTCRtpSender(track,\n transceiver.dtlsTransport);\n return transceiver.rtpSender;\n };\n\n RTCPeerConnection.prototype.addStream = function(stream) {\n var pc = this;\n if (edgeVersion >= 15025) {\n stream.getTracks().forEach(function(track) {\n pc.addTrack(track, stream);\n });\n } else {\n // Clone is necessary for local demos mostly, attaching directly\n // to two different senders does not work (build 10547).\n // Fixed in 15025 (or earlier)\n var clonedStream = stream.clone();\n stream.getTracks().forEach(function(track, idx) {\n var clonedTrack = clonedStream.getTracks()[idx];\n track.addEventListener('enabled', function(event) {\n clonedTrack.enabled = event.enabled;\n });\n });\n clonedStream.getTracks().forEach(function(track) {\n pc.addTrack(track, clonedStream);\n });\n }\n };\n\n RTCPeerConnection.prototype.removeTrack = function(sender) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call removeTrack on a closed peerconnection.');\n }\n\n if (!(sender instanceof window.RTCRtpSender)) {\n throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.');\n }\n\n var transceiver = this.transceivers.find(function(t) {\n return t.rtpSender === sender;\n });\n\n if (!transceiver) {\n throw makeError('InvalidAccessError',\n 'Sender was not created by this connection.');\n }\n var stream = transceiver.stream;\n\n transceiver.rtpSender.stop();\n transceiver.rtpSender = null;\n transceiver.track = null;\n transceiver.stream = null;\n\n // remove the stream from the set of local streams\n var localStreams = this.transceivers.map(function(t) {\n return t.stream;\n });\n if (localStreams.indexOf(stream) === -1 &&\n this.localStreams.indexOf(stream) > -1) {\n this.localStreams.splice(this.localStreams.indexOf(stream), 1);\n }\n\n this._maybeFireNegotiationNeeded();\n };\n\n RTCPeerConnection.prototype.removeStream = function(stream) {\n var pc = this;\n stream.getTracks().forEach(function(track) {\n var sender = pc.getSenders().find(function(s) {\n return s.track === track;\n });\n if (sender) {\n pc.removeTrack(sender);\n }\n });\n };\n\n RTCPeerConnection.prototype.getSenders = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpSender;\n })\n .map(function(transceiver) {\n return transceiver.rtpSender;\n });\n };\n\n RTCPeerConnection.prototype.getReceivers = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpReceiver;\n })\n .map(function(transceiver) {\n return transceiver.rtpReceiver;\n });\n };\n\n\n RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex,\n usingBundle) {\n var pc = this;\n if (usingBundle && sdpMLineIndex > 0) {\n return this.transceivers[0].iceGatherer;\n } else if (this._iceGatherers.length) {\n return this._iceGatherers.shift();\n }\n var iceGatherer = new window.RTCIceGatherer({\n iceServers: this._config.iceServers,\n gatherPolicy: this._config.iceTransportPolicy\n });\n Object.defineProperty(iceGatherer, 'state',\n {value: 'new', writable: true}\n );\n\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = [];\n this.transceivers[sdpMLineIndex].bufferCandidates = function(event) {\n var end = !event.candidate || Object.keys(event.candidate).length === 0;\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n iceGatherer.state = end ? 'completed' : 'gathering';\n if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) {\n pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event);\n }\n };\n iceGatherer.addEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n return iceGatherer;\n };\n\n // start gathering from an RTCIceGatherer.\n RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) {\n var pc = this;\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer.onlocalcandidate) {\n return;\n }\n var bufferedCandidateEvents =\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents;\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null;\n iceGatherer.removeEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n iceGatherer.onlocalcandidate = function(evt) {\n if (pc.usingBundle && sdpMLineIndex > 0) {\n // if we know that we use bundle we can drop candidates with\n // ѕdpMLineIndex > 0. If we don't do this then our state gets\n // confused since we dispose the extra ice gatherer.\n return;\n }\n var event = new Event('icecandidate');\n event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex};\n\n var cand = evt.candidate;\n // Edge emits an empty object for RTCIceCandidateComplete‥\n var end = !cand || Object.keys(cand).length === 0;\n if (end) {\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') {\n iceGatherer.state = 'completed';\n }\n } else {\n if (iceGatherer.state === 'new') {\n iceGatherer.state = 'gathering';\n }\n // RTCIceCandidate doesn't have a component, needs to be added\n cand.component = 1;\n // also the usernameFragment. TODO: update SDP to take both variants.\n cand.ufrag = iceGatherer.getLocalParameters().usernameFragment;\n\n var serializedCandidate = SDPUtils.writeCandidate(cand);\n event.candidate = Object.assign(event.candidate,\n SDPUtils.parseCandidate(serializedCandidate));\n\n event.candidate.candidate = serializedCandidate;\n event.candidate.toJSON = function() {\n return {\n candidate: event.candidate.candidate,\n sdpMid: event.candidate.sdpMid,\n sdpMLineIndex: event.candidate.sdpMLineIndex,\n usernameFragment: event.candidate.usernameFragment\n };\n };\n }\n\n // update local description.\n var sections = SDPUtils.getMediaSections(pc._localDescription.sdp);\n if (!end) {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=' + event.candidate.candidate + '\\r\\n';\n } else {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=end-of-candidates\\r\\n';\n }\n pc._localDescription.sdp =\n SDPUtils.getDescription(pc._localDescription.sdp) +\n sections.join('');\n var complete = pc.transceivers.every(function(transceiver) {\n return transceiver.iceGatherer &&\n transceiver.iceGatherer.state === 'completed';\n });\n\n if (pc.iceGatheringState !== 'gathering') {\n pc.iceGatheringState = 'gathering';\n pc._emitGatheringStateChange();\n }\n\n // Emit candidate. Also emit null candidate when all gatherers are\n // complete.\n if (!end) {\n pc._dispatchEvent('icecandidate', event);\n }\n if (complete) {\n pc._dispatchEvent('icecandidate', new Event('icecandidate'));\n pc.iceGatheringState = 'complete';\n pc._emitGatheringStateChange();\n }\n };\n\n // emit already gathered candidates.\n window.setTimeout(function() {\n bufferedCandidateEvents.forEach(function(e) {\n iceGatherer.onlocalcandidate(e);\n });\n }, 0);\n };\n\n // Create ICE transport and DTLS transport.\n RTCPeerConnection.prototype._createIceAndDtlsTransports = function() {\n var pc = this;\n var iceTransport = new window.RTCIceTransport(null);\n iceTransport.onicestatechange = function() {\n pc._updateIceConnectionState();\n pc._updateConnectionState();\n };\n\n var dtlsTransport = new window.RTCDtlsTransport(iceTransport);\n dtlsTransport.ondtlsstatechange = function() {\n pc._updateConnectionState();\n };\n dtlsTransport.onerror = function() {\n // onerror does not set state to failed by itself.\n Object.defineProperty(dtlsTransport, 'state',\n {value: 'failed', writable: true});\n pc._updateConnectionState();\n };\n\n return {\n iceTransport: iceTransport,\n dtlsTransport: dtlsTransport\n };\n };\n\n // Destroy ICE gatherer, ICE transport and DTLS transport.\n // Without triggering the callbacks.\n RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function(\n sdpMLineIndex) {\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer) {\n delete iceGatherer.onlocalcandidate;\n delete this.transceivers[sdpMLineIndex].iceGatherer;\n }\n var iceTransport = this.transceivers[sdpMLineIndex].iceTransport;\n if (iceTransport) {\n delete iceTransport.onicestatechange;\n delete this.transceivers[sdpMLineIndex].iceTransport;\n }\n var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport;\n if (dtlsTransport) {\n delete dtlsTransport.ondtlsstatechange;\n delete dtlsTransport.onerror;\n delete this.transceivers[sdpMLineIndex].dtlsTransport;\n }\n };\n\n // Start the RTP Sender and Receiver for a transceiver.\n RTCPeerConnection.prototype._transceive = function(transceiver,\n send, recv) {\n var params = getCommonCapabilities(transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n if (send && transceiver.rtpSender) {\n params.encodings = transceiver.sendEncodingParameters;\n params.rtcp = {\n cname: SDPUtils.localCName,\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.recvEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc;\n }\n transceiver.rtpSender.send(params);\n }\n if (recv && transceiver.rtpReceiver && params.codecs.length > 0) {\n // remove RTX field in Edge 14942\n if (transceiver.kind === 'video'\n && transceiver.recvEncodingParameters\n && edgeVersion < 15019) {\n transceiver.recvEncodingParameters.forEach(function(p) {\n delete p.rtx;\n });\n }\n if (transceiver.recvEncodingParameters.length) {\n params.encodings = transceiver.recvEncodingParameters;\n } else {\n params.encodings = [{}];\n }\n params.rtcp = {\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.rtcpParameters.cname) {\n params.rtcp.cname = transceiver.rtcpParameters.cname;\n }\n if (transceiver.sendEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc;\n }\n transceiver.rtpReceiver.receive(params);\n }\n };\n\n RTCPeerConnection.prototype.setLocalDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setLocalDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set local ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var sections;\n var sessionpart;\n if (description.type === 'offer') {\n // VERY limited support for SDP munging. Limited to:\n // * changing the order of codecs\n sections = SDPUtils.splitSections(description.sdp);\n sessionpart = sections.shift();\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var caps = SDPUtils.parseRtpParameters(mediaSection);\n pc.transceivers[sdpMLineIndex].localCapabilities = caps;\n });\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n });\n } else if (description.type === 'answer') {\n sections = SDPUtils.splitSections(pc._remoteDescription.sdp);\n sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var transceiver = pc.transceivers[sdpMLineIndex];\n var iceGatherer = transceiver.iceGatherer;\n var iceTransport = transceiver.iceTransport;\n var dtlsTransport = transceiver.dtlsTransport;\n var localCapabilities = transceiver.localCapabilities;\n var remoteCapabilities = transceiver.remoteCapabilities;\n\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n\n if (!rejected && !transceiver.rejected) {\n var remoteIceParameters = SDPUtils.getIceParameters(\n mediaSection, sessionpart);\n var remoteDtlsParameters = SDPUtils.getDtlsParameters(\n mediaSection, sessionpart);\n if (isIceLite) {\n remoteDtlsParameters.role = 'server';\n }\n\n if (!pc.usingBundle || sdpMLineIndex === 0) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n isIceLite ? 'controlling' : 'controlled');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // Calculate intersection of capabilities.\n var params = getCommonCapabilities(localCapabilities,\n remoteCapabilities);\n\n // Start the RTCRtpSender. The RTCRtpReceiver for this\n // transceiver has already been started in setRemoteDescription.\n pc._transceive(transceiver,\n params.codecs.length > 0,\n false);\n }\n });\n }\n\n pc._localDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-local-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.setRemoteDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setRemoteDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set remote ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var streams = {};\n pc.remoteStreams.forEach(function(stream) {\n streams[stream.id] = stream;\n });\n var receiverList = [];\n var sections = SDPUtils.splitSections(description.sdp);\n var sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n var usingBundle = SDPUtils.matchPrefix(sessionpart,\n 'a=group:BUNDLE ').length > 0;\n pc.usingBundle = usingBundle;\n var iceOptions = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-options:')[0];\n if (iceOptions) {\n pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ')\n .indexOf('trickle') >= 0;\n } else {\n pc.canTrickleIceCandidates = false;\n }\n\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var lines = SDPUtils.splitLines(mediaSection);\n var kind = SDPUtils.getKind(mediaSection);\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n var protocol = lines[0].substr(2).split(' ')[2];\n\n var direction = SDPUtils.getDirection(mediaSection, sessionpart);\n var remoteMsid = SDPUtils.parseMsid(mediaSection);\n\n var mid = SDPUtils.getMid(mediaSection) || SDPUtils.generateIdentifier();\n\n // Reject datachannels which are not implemented yet.\n if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' ||\n protocol === 'UDP/DTLS/SCTP'))) {\n // TODO: this is dangerous in the case where a non-rejected m-line\n // becomes rejected.\n pc.transceivers[sdpMLineIndex] = {\n mid: mid,\n kind: kind,\n protocol: protocol,\n rejected: true\n };\n return;\n }\n\n if (!rejected && pc.transceivers[sdpMLineIndex] &&\n pc.transceivers[sdpMLineIndex].rejected) {\n // recycle a rejected transceiver.\n pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true);\n }\n\n var transceiver;\n var iceGatherer;\n var iceTransport;\n var dtlsTransport;\n var rtpReceiver;\n var sendEncodingParameters;\n var recvEncodingParameters;\n var localCapabilities;\n\n var track;\n // FIXME: ensure the mediaSection has rtcp-mux set.\n var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection);\n var remoteIceParameters;\n var remoteDtlsParameters;\n if (!rejected) {\n remoteIceParameters = SDPUtils.getIceParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters.role = 'client';\n }\n recvEncodingParameters =\n SDPUtils.parseRtpEncodingParameters(mediaSection);\n\n var rtcpParameters = SDPUtils.parseRtcpParameters(mediaSection);\n\n var isComplete = SDPUtils.matchPrefix(mediaSection,\n 'a=end-of-candidates', sessionpart).length > 0;\n var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:')\n .map(function(cand) {\n return SDPUtils.parseCandidate(cand);\n })\n .filter(function(cand) {\n return cand.component === 1;\n });\n\n // Check if we can use BUNDLE and dispose transports.\n if ((description.type === 'offer' || description.type === 'answer') &&\n !rejected && usingBundle && sdpMLineIndex > 0 &&\n pc.transceivers[sdpMLineIndex]) {\n pc._disposeIceAndDtlsTransports(sdpMLineIndex);\n pc.transceivers[sdpMLineIndex].iceGatherer =\n pc.transceivers[0].iceGatherer;\n pc.transceivers[sdpMLineIndex].iceTransport =\n pc.transceivers[0].iceTransport;\n pc.transceivers[sdpMLineIndex].dtlsTransport =\n pc.transceivers[0].dtlsTransport;\n if (pc.transceivers[sdpMLineIndex].rtpSender) {\n pc.transceivers[sdpMLineIndex].rtpSender.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n if (pc.transceivers[sdpMLineIndex].rtpReceiver) {\n pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n }\n if (description.type === 'offer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex] ||\n pc._createTransceiver(kind);\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n usingBundle);\n }\n\n if (cands.length && transceiver.iceTransport.state === 'new') {\n if (isComplete && (!usingBundle || sdpMLineIndex === 0)) {\n transceiver.iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n localCapabilities = window.RTCRtpReceiver.getCapabilities(kind);\n\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n\n sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 2) * 1001\n }];\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n var isNewTrack = false;\n if (direction === 'sendrecv' || direction === 'sendonly') {\n isNewTrack = !transceiver.rtpReceiver;\n rtpReceiver = transceiver.rtpReceiver ||\n new window.RTCRtpReceiver(transceiver.dtlsTransport, kind);\n\n if (isNewTrack) {\n var stream;\n track = rtpReceiver.track;\n // FIXME: does not work with Plan B.\n if (remoteMsid && remoteMsid.stream === '-') {\n // no-op. a stream id of '-' means: no associated stream.\n } else if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n Object.defineProperty(streams[remoteMsid.stream], 'id', {\n get: function() {\n return remoteMsid.stream;\n }\n });\n }\n Object.defineProperty(track, 'id', {\n get: function() {\n return remoteMsid.track;\n }\n });\n stream = streams[remoteMsid.stream];\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n stream = streams.default;\n }\n if (stream) {\n addTrackToStreamAndFireEvent(track, stream);\n transceiver.associatedRemoteMediaStreams.push(stream);\n }\n receiverList.push([track, rtpReceiver, stream]);\n }\n } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) {\n transceiver.associatedRemoteMediaStreams.forEach(function(s) {\n var nativeTrack = s.getTracks().find(function(t) {\n return t.id === transceiver.rtpReceiver.track.id;\n });\n if (nativeTrack) {\n removeTrackFromStreamAndFireEvent(nativeTrack, s);\n }\n });\n transceiver.associatedRemoteMediaStreams = [];\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.remoteCapabilities = remoteCapabilities;\n transceiver.rtpReceiver = rtpReceiver;\n transceiver.rtcpParameters = rtcpParameters;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n transceiver.recvEncodingParameters = recvEncodingParameters;\n\n // Start the RTCRtpReceiver now. The RTPSender is started in\n // setLocalDescription.\n pc._transceive(pc.transceivers[sdpMLineIndex],\n false,\n isNewTrack);\n } else if (description.type === 'answer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex];\n iceGatherer = transceiver.iceGatherer;\n iceTransport = transceiver.iceTransport;\n dtlsTransport = transceiver.dtlsTransport;\n rtpReceiver = transceiver.rtpReceiver;\n sendEncodingParameters = transceiver.sendEncodingParameters;\n localCapabilities = transceiver.localCapabilities;\n\n pc.transceivers[sdpMLineIndex].recvEncodingParameters =\n recvEncodingParameters;\n pc.transceivers[sdpMLineIndex].remoteCapabilities =\n remoteCapabilities;\n pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters;\n\n if (cands.length && iceTransport.state === 'new') {\n if ((isIceLite || isComplete) &&\n (!usingBundle || sdpMLineIndex === 0)) {\n iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n if (!usingBundle || sdpMLineIndex === 0) {\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n 'controlling');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // If the offer contained RTX but the answer did not,\n // remove RTX from sendEncodingParameters.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n pc._transceive(transceiver,\n direction === 'sendrecv' || direction === 'recvonly',\n direction === 'sendrecv' || direction === 'sendonly');\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n if (rtpReceiver &&\n (direction === 'sendrecv' || direction === 'sendonly')) {\n track = rtpReceiver.track;\n if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]);\n receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]);\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams.default);\n receiverList.push([track, rtpReceiver, streams.default]);\n }\n } else {\n // FIXME: actually the receiver should be created later.\n delete transceiver.rtpReceiver;\n }\n }\n });\n\n if (pc._dtlsRole === undefined) {\n pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive';\n }\n\n pc._remoteDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-remote-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n Object.keys(streams).forEach(function(sid) {\n var stream = streams[sid];\n if (stream.getTracks().length) {\n if (pc.remoteStreams.indexOf(stream) === -1) {\n pc.remoteStreams.push(stream);\n var event = new Event('addstream');\n event.stream = stream;\n window.setTimeout(function() {\n pc._dispatchEvent('addstream', event);\n });\n }\n\n receiverList.forEach(function(item) {\n var track = item[0];\n var receiver = item[1];\n if (stream.id !== item[2].id) {\n return;\n }\n fireAddTrack(pc, track, receiver, [stream]);\n });\n }\n });\n receiverList.forEach(function(item) {\n if (item[2]) {\n return;\n }\n fireAddTrack(pc, item[0], item[1], []);\n });\n\n // check whether addIceCandidate({}) was called within four seconds after\n // setRemoteDescription.\n window.setTimeout(function() {\n if (!(pc && pc.transceivers)) {\n return;\n }\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport &&\n transceiver.iceTransport.state === 'new' &&\n transceiver.iceTransport.getRemoteCandidates().length > 0) {\n console.warn('Timeout for addRemoteCandidate. Consider sending ' +\n 'an end-of-candidates notification');\n transceiver.iceTransport.addRemoteCandidate({});\n }\n });\n }, 4000);\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.close = function() {\n this.transceivers.forEach(function(transceiver) {\n /* not yet\n if (transceiver.iceGatherer) {\n transceiver.iceGatherer.close();\n }\n */\n if (transceiver.iceTransport) {\n transceiver.iceTransport.stop();\n }\n if (transceiver.dtlsTransport) {\n transceiver.dtlsTransport.stop();\n }\n if (transceiver.rtpSender) {\n transceiver.rtpSender.stop();\n }\n if (transceiver.rtpReceiver) {\n transceiver.rtpReceiver.stop();\n }\n });\n // FIXME: clean up tracks, local streams, remote streams, etc\n this._isClosed = true;\n this._updateSignalingState('closed');\n };\n\n // Update the signaling state.\n RTCPeerConnection.prototype._updateSignalingState = function(newState) {\n this.signalingState = newState;\n var event = new Event('signalingstatechange');\n this._dispatchEvent('signalingstatechange', event);\n };\n\n // Determine whether to fire the negotiationneeded event.\n RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() {\n var pc = this;\n if (this.signalingState !== 'stable' || this.needNegotiation === true) {\n return;\n }\n this.needNegotiation = true;\n window.setTimeout(function() {\n if (pc.needNegotiation) {\n pc.needNegotiation = false;\n var event = new Event('negotiationneeded');\n pc._dispatchEvent('negotiationneeded', event);\n }\n }, 0);\n };\n\n // Update the ice connection state.\n RTCPeerConnection.prototype._updateIceConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n checking: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n }\n });\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.checking > 0) {\n newState = 'checking';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n } else if (states.completed > 0) {\n newState = 'completed';\n }\n\n if (newState !== this.iceConnectionState) {\n this.iceConnectionState = newState;\n var event = new Event('iceconnectionstatechange');\n this._dispatchEvent('iceconnectionstatechange', event);\n }\n };\n\n // Update the connection state.\n RTCPeerConnection.prototype._updateConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n connecting: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && transceiver.dtlsTransport &&\n !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n states[transceiver.dtlsTransport.state]++;\n }\n });\n // ICETransport.completed and connected are the same for this purpose.\n states.connected += states.completed;\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.connecting > 0) {\n newState = 'connecting';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n }\n\n if (newState !== this.connectionState) {\n this.connectionState = newState;\n var event = new Event('connectionstatechange');\n this._dispatchEvent('connectionstatechange', event);\n }\n };\n\n RTCPeerConnection.prototype.createOffer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createOffer after close'));\n }\n\n var numAudioTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'audio';\n }).length;\n var numVideoTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'video';\n }).length;\n\n // Determine number of audio and video tracks we need to send/recv.\n var offerOptions = arguments[0];\n if (offerOptions) {\n // Reject Chrome legacy constraints.\n if (offerOptions.mandatory || offerOptions.optional) {\n throw new TypeError(\n 'Legacy mandatory/optional constraints not supported.');\n }\n if (offerOptions.offerToReceiveAudio !== undefined) {\n if (offerOptions.offerToReceiveAudio === true) {\n numAudioTracks = 1;\n } else if (offerOptions.offerToReceiveAudio === false) {\n numAudioTracks = 0;\n } else {\n numAudioTracks = offerOptions.offerToReceiveAudio;\n }\n }\n if (offerOptions.offerToReceiveVideo !== undefined) {\n if (offerOptions.offerToReceiveVideo === true) {\n numVideoTracks = 1;\n } else if (offerOptions.offerToReceiveVideo === false) {\n numVideoTracks = 0;\n } else {\n numVideoTracks = offerOptions.offerToReceiveVideo;\n }\n }\n }\n\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.kind === 'audio') {\n numAudioTracks--;\n if (numAudioTracks < 0) {\n transceiver.wantReceive = false;\n }\n } else if (transceiver.kind === 'video') {\n numVideoTracks--;\n if (numVideoTracks < 0) {\n transceiver.wantReceive = false;\n }\n }\n });\n\n // Create M-lines for recvonly streams.\n while (numAudioTracks > 0 || numVideoTracks > 0) {\n if (numAudioTracks > 0) {\n pc._createTransceiver('audio');\n numAudioTracks--;\n }\n if (numVideoTracks > 0) {\n pc._createTransceiver('video');\n numVideoTracks--;\n }\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n // For each track, create an ice gatherer, ice transport,\n // dtls transport, potentially rtpsender and rtpreceiver.\n var track = transceiver.track;\n var kind = transceiver.kind;\n var mid = transceiver.mid || SDPUtils.generateIdentifier();\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n pc.usingBundle);\n }\n\n var localCapabilities = window.RTCRtpSender.getCapabilities(kind);\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n localCapabilities.codecs.forEach(function(codec) {\n // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552\n // by adding level-asymmetry-allowed=1\n if (codec.name === 'H264' &&\n codec.parameters['level-asymmetry-allowed'] === undefined) {\n codec.parameters['level-asymmetry-allowed'] = '1';\n }\n\n // for subsequent offers, we might have to re-use the payload\n // type of the last offer.\n if (transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.codecs) {\n transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) {\n if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() &&\n codec.clockRate === remoteCodec.clockRate) {\n codec.preferredPayloadType = remoteCodec.payloadType;\n }\n });\n }\n });\n localCapabilities.headerExtensions.forEach(function(hdrExt) {\n var remoteExtensions = transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.headerExtensions || [];\n remoteExtensions.forEach(function(rHdrExt) {\n if (hdrExt.uri === rHdrExt.uri) {\n hdrExt.id = rHdrExt.id;\n }\n });\n });\n\n // generate an ssrc now, to be used later in rtpSender.send\n var sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 1) * 1001\n }];\n if (track) {\n // add RTX\n if (edgeVersion >= 15019 && kind === 'video' &&\n !sendEncodingParameters[0].rtx) {\n sendEncodingParameters[0].rtx = {\n ssrc: sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n\n if (transceiver.wantReceive) {\n transceiver.rtpReceiver = new window.RTCRtpReceiver(\n transceiver.dtlsTransport, kind);\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n });\n\n // always offer BUNDLE and dispose on return if not supported.\n if (pc._config.bundlePolicy !== 'max-compat') {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n sdp += writeMediaSection(transceiver, transceiver.localCapabilities,\n 'offer', transceiver.stream, pc._dtlsRole);\n sdp += 'a=rtcp-rsize\\r\\n';\n\n if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' &&\n (sdpMLineIndex === 0 || !pc.usingBundle)) {\n transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) {\n cand.component = 1;\n sdp += 'a=' + SDPUtils.writeCandidate(cand) + '\\r\\n';\n });\n\n if (transceiver.iceGatherer.state === 'completed') {\n sdp += 'a=end-of-candidates\\r\\n';\n }\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'offer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.createAnswer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer after close'));\n }\n\n if (!(pc.signalingState === 'have-remote-offer' ||\n pc.signalingState === 'have-local-pranswer')) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer in signalingState ' + pc.signalingState));\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n if (pc.usingBundle) {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n var mediaSectionsInOffer = SDPUtils.getMediaSections(\n pc._remoteDescription.sdp).length;\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n if (sdpMLineIndex + 1 > mediaSectionsInOffer) {\n return;\n }\n if (transceiver.rejected) {\n if (transceiver.kind === 'application') {\n if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt\n sdp += 'm=application 0 DTLS/SCTP 5000\\r\\n';\n } else {\n sdp += 'm=application 0 ' + transceiver.protocol +\n ' webrtc-datachannel\\r\\n';\n }\n } else if (transceiver.kind === 'audio') {\n sdp += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\\r\\n' +\n 'a=rtpmap:0 PCMU/8000\\r\\n';\n } else if (transceiver.kind === 'video') {\n sdp += 'm=video 0 UDP/TLS/RTP/SAVPF 120\\r\\n' +\n 'a=rtpmap:120 VP8/90000\\r\\n';\n }\n sdp += 'c=IN IP4 0.0.0.0\\r\\n' +\n 'a=inactive\\r\\n' +\n 'a=mid:' + transceiver.mid + '\\r\\n';\n return;\n }\n\n // FIXME: look at direction.\n if (transceiver.stream) {\n var localTrack;\n if (transceiver.kind === 'audio') {\n localTrack = transceiver.stream.getAudioTracks()[0];\n } else if (transceiver.kind === 'video') {\n localTrack = transceiver.stream.getVideoTracks()[0];\n }\n if (localTrack) {\n // add RTX\n if (edgeVersion >= 15019 && transceiver.kind === 'video' &&\n !transceiver.sendEncodingParameters[0].rtx) {\n transceiver.sendEncodingParameters[0].rtx = {\n ssrc: transceiver.sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n }\n\n // Calculate intersection of capabilities.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n sdp += writeMediaSection(transceiver, commonCapabilities,\n 'answer', transceiver.stream, pc._dtlsRole);\n if (transceiver.rtcpParameters &&\n transceiver.rtcpParameters.reducedSize) {\n sdp += 'a=rtcp-rsize\\r\\n';\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'answer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.addIceCandidate = function(candidate) {\n var pc = this;\n var sections;\n if (candidate && !(candidate.sdpMLineIndex !== undefined ||\n candidate.sdpMid)) {\n return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required'));\n }\n\n // TODO: needs to go into ops queue.\n return new Promise(function(resolve, reject) {\n if (!pc._remoteDescription) {\n return reject(makeError('InvalidStateError',\n 'Can not add ICE candidate without a remote description'));\n } else if (!candidate || candidate.candidate === '') {\n for (var j = 0; j < pc.transceivers.length; j++) {\n if (pc.transceivers[j].rejected) {\n continue;\n }\n pc.transceivers[j].iceTransport.addRemoteCandidate({});\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[j] += 'a=end-of-candidates\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n if (pc.usingBundle) {\n break;\n }\n }\n } else {\n var sdpMLineIndex = candidate.sdpMLineIndex;\n if (candidate.sdpMid) {\n for (var i = 0; i < pc.transceivers.length; i++) {\n if (pc.transceivers[i].mid === candidate.sdpMid) {\n sdpMLineIndex = i;\n break;\n }\n }\n }\n var transceiver = pc.transceivers[sdpMLineIndex];\n if (transceiver) {\n if (transceiver.rejected) {\n return resolve();\n }\n var cand = Object.keys(candidate.candidate).length > 0 ?\n SDPUtils.parseCandidate(candidate.candidate) : {};\n // Ignore Chrome's invalid candidates since Edge does not like them.\n if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) {\n return resolve();\n }\n // Ignore RTCP candidates, we assume RTCP-MUX.\n if (cand.component && cand.component !== 1) {\n return resolve();\n }\n // when using bundle, avoid adding candidates to the wrong\n // ice transport. And avoid adding candidates added in the SDP.\n if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 &&\n transceiver.iceTransport !== pc.transceivers[0].iceTransport)) {\n if (!maybeAddCandidate(transceiver.iceTransport, cand)) {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n\n // update the remoteDescription.\n var candidateString = candidate.candidate.trim();\n if (candidateString.indexOf('a=') === 0) {\n candidateString = candidateString.substr(2);\n }\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[sdpMLineIndex] += 'a=' +\n (cand.type ? candidateString : 'end-of-candidates')\n + '\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n } else {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n resolve();\n });\n };\n\n RTCPeerConnection.prototype.getStats = function(selector) {\n if (selector && selector instanceof window.MediaStreamTrack) {\n var senderOrReceiver = null;\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.rtpSender &&\n transceiver.rtpSender.track === selector) {\n senderOrReceiver = transceiver.rtpSender;\n } else if (transceiver.rtpReceiver &&\n transceiver.rtpReceiver.track === selector) {\n senderOrReceiver = transceiver.rtpReceiver;\n }\n });\n if (!senderOrReceiver) {\n throw makeError('InvalidAccessError', 'Invalid selector.');\n }\n return senderOrReceiver.getStats();\n }\n\n var promises = [];\n this.transceivers.forEach(function(transceiver) {\n ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport',\n 'dtlsTransport'].forEach(function(method) {\n if (transceiver[method]) {\n promises.push(transceiver[method].getStats());\n }\n });\n });\n return Promise.all(promises).then(function(allStats) {\n var results = new Map();\n allStats.forEach(function(stats) {\n stats.forEach(function(stat) {\n results.set(stat.id, stat);\n });\n });\n return results;\n });\n };\n\n // fix low-level stat names and return Map instead of object.\n var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer',\n 'RTCIceTransport', 'RTCDtlsTransport'];\n ortcObjects.forEach(function(ortcObjectName) {\n var obj = window[ortcObjectName];\n if (obj && obj.prototype && obj.prototype.getStats) {\n var nativeGetstats = obj.prototype.getStats;\n obj.prototype.getStats = function() {\n return nativeGetstats.apply(this)\n .then(function(nativeStats) {\n var mapStats = new Map();\n Object.keys(nativeStats).forEach(function(id) {\n nativeStats[id].type = fixStatsType(nativeStats[id]);\n mapStats.set(id, nativeStats[id]);\n });\n return mapStats;\n });\n };\n }\n });\n\n // legacy callback shims. Should be moved to adapter.js some days.\n var methods = ['createOffer', 'createAnswer'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[0] === 'function' ||\n typeof args[1] === 'function') { // legacy\n return nativeMethod.apply(this, [arguments[2]])\n .then(function(description) {\n if (typeof args[0] === 'function') {\n args[0].apply(null, [description]);\n }\n }, function(error) {\n if (typeof args[1] === 'function') {\n args[1].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function' ||\n typeof args[2] === 'function') { // legacy\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n }, function(error) {\n if (typeof args[2] === 'function') {\n args[2].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n // getStats is special. It doesn't have a spec legacy method yet we support\n // getStats(something, cb) without error callbacks.\n ['getStats'].forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function') {\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n return RTCPeerConnection;\n};\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n const shimError_ = function(e) {\n return {\n name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name,\n message: e.message,\n constraint: e.constraint,\n toString() {\n return this.name;\n }\n };\n };\n\n // getUserMedia error shim.\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e)));\n };\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window) {\n if (!('getDisplayMedia' in window.navigator)) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n window.navigator.getDisplayMedia.bind(window.navigator);\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nimport {filterIceServers} from './filtericeservers';\nimport shimRTCPeerConnection from 'rtcpeerconnection-shim';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimPeerConnection(window, browserDetails) {\n if (window.RTCIceGatherer) {\n if (!window.RTCIceCandidate) {\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n return args;\n };\n }\n if (!window.RTCSessionDescription) {\n window.RTCSessionDescription = function RTCSessionDescription(args) {\n return args;\n };\n }\n // this adds an additional event listener to MediaStrackTrack that signals\n // when a tracks enabled property was changed. Workaround for a bug in\n // addStream, see below. No longer required in 15025+\n if (browserDetails.version < 15025) {\n const origMSTEnabled = Object.getOwnPropertyDescriptor(\n window.MediaStreamTrack.prototype, 'enabled');\n Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', {\n set(value) {\n origMSTEnabled.set.call(this, value);\n const ev = new Event('enabled');\n ev.enabled = value;\n this.dispatchEvent(ev);\n }\n });\n }\n }\n\n // ORTC defines the DTMF sender a bit different.\n // https://github.com/w3c/ortc/issues/714\n if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) {\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = new window.RTCDtmfSender(this);\n } else if (this.track.kind === 'video') {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n // Edge currently only implements the RTCDtmfSender, not the\n // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2*\n if (window.RTCDtmfSender && !window.RTCDTMFSender) {\n window.RTCDTMFSender = window.RTCDtmfSender;\n }\n\n const RTCPeerConnectionShim = shimRTCPeerConnection(window,\n browserDetails.version);\n window.RTCPeerConnection = function RTCPeerConnection(config) {\n if (config && config.iceServers) {\n config.iceServers = filterIceServers(config.iceServers,\n browserDetails.version);\n utils.log('ICE servers after filtering:', config.iceServers);\n }\n return new RTCPeerConnectionShim(config);\n };\n window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype;\n}\n\nexport function shimReplaceTrack(window) {\n // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614\n if (window.RTCRtpSender &&\n !('replaceTrack' in window.RTCRtpSender.prototype)) {\n window.RTCRtpSender.prototype.replaceTrack =\n window.RTCRtpSender.prototype.setTrack;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n const MediaStreamTrack = window && window.MediaStreamTrack;\n\n navigator.getUserMedia = function(constraints, onSuccess, onError) {\n // Replace Firefox 44+'s deprecation warning with unprefixed version.\n utils.deprecated('navigator.getUserMedia',\n 'navigator.mediaDevices.getUserMedia');\n navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError);\n };\n\n if (!(browserDetails.version > 55 &&\n 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n\n const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n if (typeof c === 'object' && typeof c.audio === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c.audio, 'autoGainControl', 'mozAutoGainControl');\n remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeGetUserMedia(c);\n };\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) {\n const nativeGetSettings = MediaStreamTrack.prototype.getSettings;\n MediaStreamTrack.prototype.getSettings = function() {\n const obj = nativeGetSettings.apply(this, arguments);\n remap(obj, 'mozAutoGainControl', 'autoGainControl');\n remap(obj, 'mozNoiseSuppression', 'noiseSuppression');\n return obj;\n };\n }\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) {\n const nativeApplyConstraints =\n MediaStreamTrack.prototype.applyConstraints;\n MediaStreamTrack.prototype.applyConstraints = function(c) {\n if (this.kind === 'audio' && typeof c === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c, 'autoGainControl', 'mozAutoGainControl');\n remap(c, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeApplyConstraints.apply(this, [c]);\n };\n }\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window, preferredMediaSource) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n if (!(constraints && constraints.video)) {\n const err = new DOMException('getDisplayMedia without video ' +\n 'constraints is undefined');\n err.name = 'NotFoundError';\n // from https://heycam.github.io/webidl/#idl-DOMException-error-names\n err.code = 8;\n return Promise.reject(err);\n }\n if (constraints.video === true) {\n constraints.video = {mediaSource: preferredMediaSource};\n } else {\n constraints.video.mediaSource = preferredMediaSource;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCTrackEvent &&\n ('receiver' in window.RTCTrackEvent.prototype) &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (typeof window !== 'object' ||\n !(window.RTCPeerConnection || window.mozRTCPeerConnection)) {\n return; // probably media.peerconnection.enabled=false in about:config\n }\n if (!window.RTCPeerConnection && window.mozRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.mozRTCPeerConnection;\n }\n\n if (browserDetails.version < 53) {\n // shim away need for obsolete RTCIceCandidate/RTCSessionDescription.\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n\n const modernStatsTypes = {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n };\n\n const nativeGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n return nativeGetStats.apply(this, [selector || null])\n .then(stats => {\n if (browserDetails.version < 53 && !onSucc) {\n // Shim only promise getStats with spec-hyphens in type names\n // Leave callback version alone; misc old uses of forEach before Map\n try {\n stats.forEach(stat => {\n stat.type = modernStatsTypes[stat.type] || stat.type;\n });\n } catch (e) {\n if (e.name !== 'TypeError') {\n throw e;\n }\n // Avoid TypeError: \"type\" is read-only, in old versions. 34-43ish\n stats.forEach((stat, i) => {\n stats.set(i, Object.assign({}, stat, {\n type: modernStatsTypes[stat.type] || stat.type\n }));\n });\n }\n }\n return stats;\n })\n .then(onSucc, onErr);\n };\n}\n\nexport function shimSenderGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) {\n return;\n }\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n return this.track ? this._pc.getStats(this.track) :\n Promise.resolve(new Map());\n };\n}\n\nexport function shimReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) {\n return;\n }\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers = function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n return this._pc.getStats(this.track);\n };\n}\n\nexport function shimRemoveStream(window) {\n if (!window.RTCPeerConnection ||\n 'removeStream' in window.RTCPeerConnection.prototype) {\n return;\n }\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n utils.deprecated('removeStream', 'removeTrack');\n this.getSenders().forEach(sender => {\n if (sender.track && stream.getTracks().includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n}\n\nexport function shimRTCDataChannel(window) {\n // rename DataChannel to RTCDataChannel (native fix in FF60):\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851\n if (window.DataChannel && !window.RTCDataChannel) {\n window.RTCDataChannel = window.DataChannel;\n }\n}\n\nexport function shimAddTransceiver(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver;\n if (origAddTransceiver) {\n window.RTCPeerConnection.prototype.addTransceiver =\n function addTransceiver() {\n this.setParametersPromises = [];\n const initParameters = arguments[1];\n const shouldPerformCheck = initParameters &&\n 'sendEncodings' in initParameters;\n if (shouldPerformCheck) {\n // If sendEncodings params are provided, validate grammar\n initParameters.sendEncodings.forEach((encodingParam) => {\n if ('rid' in encodingParam) {\n const ridRegex = /^[a-z0-9]{0,16}$/i;\n if (!ridRegex.test(encodingParam.rid)) {\n throw new TypeError('Invalid RID value provided.');\n }\n }\n if ('scaleResolutionDownBy' in encodingParam) {\n if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) {\n throw new RangeError('scale_resolution_down_by must be >= 1.0');\n }\n }\n if ('maxFramerate' in encodingParam) {\n if (!(parseFloat(encodingParam.maxFramerate) >= 0)) {\n throw new RangeError('max_framerate must be >= 0.0');\n }\n }\n });\n }\n const transceiver = origAddTransceiver.apply(this, arguments);\n if (shouldPerformCheck) {\n // Check if the init options were applied. If not we do this in an\n // asynchronous way and save the promise reference in a global object.\n // This is an ugly hack, but at the same time is way more robust than\n // checking the sender parameters before and after the createOffer\n // Also note that after the createoffer we are not 100% sure that\n // the params were asynchronously applied so we might miss the\n // opportunity to recreate offer.\n const {sender} = transceiver;\n const params = sender.getParameters();\n if (!('encodings' in params) ||\n // Avoid being fooled by patched getParameters() below.\n (params.encodings.length === 1 &&\n Object.keys(params.encodings[0]).length === 0)) {\n params.encodings = initParameters.sendEncodings;\n sender.sendEncodings = initParameters.sendEncodings;\n this.setParametersPromises.push(sender.setParameters(params)\n .then(() => {\n delete sender.sendEncodings;\n }).catch(() => {\n delete sender.sendEncodings;\n })\n );\n }\n }\n return transceiver;\n };\n }\n}\n\nexport function shimGetParameters(window) {\n if (!(typeof window === 'object' && window.RTCRtpSender)) {\n return;\n }\n const origGetParameters = window.RTCRtpSender.prototype.getParameters;\n if (origGetParameters) {\n window.RTCRtpSender.prototype.getParameters =\n function getParameters() {\n const params = origGetParameters.apply(this, arguments);\n if (!('encodings' in params)) {\n params.encodings = [].concat(this.sendEncodings || [{}]);\n }\n return params;\n };\n }\n}\n\nexport function shimCreateOffer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer = function createOffer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateOffer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimCreateAnswer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer;\n window.RTCPeerConnection.prototype.createAnswer = function createAnswer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateAnswer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateAnswer.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n'use strict';\nimport * as utils from '../utils';\n\nexport function shimLocalStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n return this._localStreams;\n };\n }\n if (!('addStream' in window.RTCPeerConnection.prototype)) {\n const _addTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n // Try to emulate Chrome's behaviour of adding in audio-video order.\n // Safari orders by track id.\n stream.getAudioTracks().forEach(track => _addTrack.call(this, track,\n stream));\n stream.getVideoTracks().forEach(track => _addTrack.call(this, track,\n stream));\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, ...streams) {\n if (streams) {\n streams.forEach((stream) => {\n if (!this._localStreams) {\n this._localStreams = [stream];\n } else if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n });\n }\n return _addTrack.apply(this, arguments);\n };\n }\n if (!('removeStream' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n const index = this._localStreams.indexOf(stream);\n if (index === -1) {\n return;\n }\n this._localStreams.splice(index, 1);\n const tracks = stream.getTracks();\n this.getSenders().forEach(sender => {\n if (tracks.includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n }\n}\n\nexport function shimRemoteStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getRemoteStreams =\n function getRemoteStreams() {\n return this._remoteStreams ? this._remoteStreams : [];\n };\n }\n if (!('onaddstream' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', {\n get() {\n return this._onaddstream;\n },\n set(f) {\n if (this._onaddstream) {\n this.removeEventListener('addstream', this._onaddstream);\n this.removeEventListener('track', this._onaddstreampoly);\n }\n this.addEventListener('addstream', this._onaddstream = f);\n this.addEventListener('track', this._onaddstreampoly = (e) => {\n e.streams.forEach(stream => {\n if (!this._remoteStreams) {\n this._remoteStreams = [];\n }\n if (this._remoteStreams.includes(stream)) {\n return;\n }\n this._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n this.dispatchEvent(event);\n });\n });\n }\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n const pc = this;\n if (!this._onaddstreampoly) {\n this.addEventListener('track', this._onaddstreampoly = function(e) {\n e.streams.forEach(stream => {\n if (!pc._remoteStreams) {\n pc._remoteStreams = [];\n }\n if (pc._remoteStreams.indexOf(stream) >= 0) {\n return;\n }\n pc._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n pc.dispatchEvent(event);\n });\n });\n }\n return origSetRemoteDescription.apply(pc, arguments);\n };\n }\n}\n\nexport function shimCallbacksAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n const prototype = window.RTCPeerConnection.prototype;\n const origCreateOffer = prototype.createOffer;\n const origCreateAnswer = prototype.createAnswer;\n const setLocalDescription = prototype.setLocalDescription;\n const setRemoteDescription = prototype.setRemoteDescription;\n const addIceCandidate = prototype.addIceCandidate;\n\n prototype.createOffer =\n function createOffer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateOffer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n prototype.createAnswer =\n function createAnswer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateAnswer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n let withCallback = function(description, successCallback, failureCallback) {\n const promise = setLocalDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setLocalDescription = withCallback;\n\n withCallback = function(description, successCallback, failureCallback) {\n const promise = setRemoteDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setRemoteDescription = withCallback;\n\n withCallback = function(candidate, successCallback, failureCallback) {\n const promise = addIceCandidate.apply(this, [candidate]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.addIceCandidate = withCallback;\n}\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {\n // shim not needed in Safari 12.1\n const mediaDevices = navigator.mediaDevices;\n const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices);\n navigator.mediaDevices.getUserMedia = (constraints) => {\n return _getUserMedia(shimConstraints(constraints));\n };\n }\n\n if (!navigator.getUserMedia && navigator.mediaDevices &&\n navigator.mediaDevices.getUserMedia) {\n navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) {\n navigator.mediaDevices.getUserMedia(constraints)\n .then(cb, errcb);\n }.bind(navigator);\n }\n}\n\nexport function shimConstraints(constraints) {\n if (constraints && constraints.video !== undefined) {\n return Object.assign({},\n constraints,\n {video: utils.compactObject(constraints.video)}\n );\n }\n\n return constraints;\n}\n\nexport function shimRTCIceServerUrls(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // migrate from non-spec RTCIceServer.url to RTCIceServer.urls\n const OrigPeerConnection = window.RTCPeerConnection;\n window.RTCPeerConnection =\n function RTCPeerConnection(pcConfig, pcConstraints) {\n if (pcConfig && pcConfig.iceServers) {\n const newIceServers = [];\n for (let i = 0; i < pcConfig.iceServers.length; i++) {\n let server = pcConfig.iceServers[i];\n if (!server.hasOwnProperty('urls') &&\n server.hasOwnProperty('url')) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n server = JSON.parse(JSON.stringify(server));\n server.urls = server.url;\n delete server.url;\n newIceServers.push(server);\n } else {\n newIceServers.push(pcConfig.iceServers[i]);\n }\n }\n pcConfig.iceServers = newIceServers;\n }\n return new OrigPeerConnection(pcConfig, pcConstraints);\n };\n window.RTCPeerConnection.prototype = OrigPeerConnection.prototype;\n // wrap static methods. Currently just generateCertificate.\n if ('generateCertificate' in OrigPeerConnection) {\n Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', {\n get() {\n return OrigPeerConnection.generateCertificate;\n }\n });\n }\n}\n\nexport function shimTrackEventTransceiver(window) {\n // Add event.transceiver member over deprecated event.receiver\n if (typeof window === 'object' && window.RTCTrackEvent &&\n 'receiver' in window.RTCTrackEvent.prototype &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimCreateOfferLegacy(window) {\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer =\n function createOffer(offerOptions) {\n if (offerOptions) {\n if (typeof offerOptions.offerToReceiveAudio !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveAudio =\n !!offerOptions.offerToReceiveAudio;\n }\n const audioTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'audio');\n if (offerOptions.offerToReceiveAudio === false && audioTransceiver) {\n if (audioTransceiver.direction === 'sendrecv') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('sendonly');\n } else {\n audioTransceiver.direction = 'sendonly';\n }\n } else if (audioTransceiver.direction === 'recvonly') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('inactive');\n } else {\n audioTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveAudio === true &&\n !audioTransceiver) {\n this.addTransceiver('audio');\n }\n\n if (typeof offerOptions.offerToReceiveVideo !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveVideo =\n !!offerOptions.offerToReceiveVideo;\n }\n const videoTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'video');\n if (offerOptions.offerToReceiveVideo === false && videoTransceiver) {\n if (videoTransceiver.direction === 'sendrecv') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('sendonly');\n } else {\n videoTransceiver.direction = 'sendonly';\n }\n } else if (videoTransceiver.direction === 'recvonly') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('inactive');\n } else {\n videoTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveVideo === true &&\n !videoTransceiver) {\n this.addTransceiver('video');\n }\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimAudioContext(window) {\n if (typeof window !== 'object' || window.AudioContext) {\n return;\n }\n window.AudioContext = window.webkitAudioContext;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport SDPUtils from 'sdp';\nimport * as utils from './utils';\n\nexport function shimRTCIceCandidate(window) {\n // foundation is arbitrarily chosen as an indicator for full support for\n // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface\n if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in\n window.RTCIceCandidate.prototype)) {\n return;\n }\n\n const NativeRTCIceCandidate = window.RTCIceCandidate;\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n // Remove the a= which shouldn't be part of the candidate string.\n if (typeof args === 'object' && args.candidate &&\n args.candidate.indexOf('a=') === 0) {\n args = JSON.parse(JSON.stringify(args));\n args.candidate = args.candidate.substr(2);\n }\n\n if (args.candidate && args.candidate.length) {\n // Augment the native candidate with the parsed fields.\n const nativeCandidate = new NativeRTCIceCandidate(args);\n const parsedCandidate = SDPUtils.parseCandidate(args.candidate);\n const augmentedCandidate = Object.assign(nativeCandidate,\n parsedCandidate);\n\n // Add a serializer that does not serialize the extra attributes.\n augmentedCandidate.toJSON = function toJSON() {\n return {\n candidate: augmentedCandidate.candidate,\n sdpMid: augmentedCandidate.sdpMid,\n sdpMLineIndex: augmentedCandidate.sdpMLineIndex,\n usernameFragment: augmentedCandidate.usernameFragment,\n };\n };\n return augmentedCandidate;\n }\n return new NativeRTCIceCandidate(args);\n };\n window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype;\n\n // Hook up the augmented candidate in onicecandidate and\n // addEventListener('icecandidate', ...)\n utils.wrapPeerConnectionEvent(window, 'icecandidate', e => {\n if (e.candidate) {\n Object.defineProperty(e, 'candidate', {\n value: new window.RTCIceCandidate(e.candidate),\n writable: 'false'\n });\n }\n return e;\n });\n}\n\nexport function shimMaxMessageSize(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n if (!('sctp' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n }\n });\n }\n\n const sctpInDescription = function(description) {\n if (!description || !description.sdp) {\n return false;\n }\n const sections = SDPUtils.splitSections(description.sdp);\n sections.shift();\n return sections.some(mediaSection => {\n const mLine = SDPUtils.parseMLine(mediaSection);\n return mLine && mLine.kind === 'application'\n && mLine.protocol.indexOf('SCTP') !== -1;\n });\n };\n\n const getRemoteFirefoxVersion = function(description) {\n // TODO: Is there a better solution for detecting Firefox?\n const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\\d+)/);\n if (match === null || match.length < 2) {\n return -1;\n }\n const version = parseInt(match[1], 10);\n // Test for NaN (yes, this is ugly)\n return version !== version ? -1 : version;\n };\n\n const getCanSendMaxMessageSize = function(remoteIsFirefox) {\n // Every implementation we know can send at least 64 KiB.\n // Note: Although Chrome is technically able to send up to 256 KiB, the\n // data does not reach the other peer reliably.\n // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419\n let canSendMaxMessageSize = 65536;\n if (browserDetails.browser === 'firefox') {\n if (browserDetails.version < 57) {\n if (remoteIsFirefox === -1) {\n // FF < 57 will send in 16 KiB chunks using the deprecated PPID\n // fragmentation.\n canSendMaxMessageSize = 16384;\n } else {\n // However, other FF (and RAWRTC) can reassemble PPID-fragmented\n // messages. Thus, supporting ~2 GiB when sending.\n canSendMaxMessageSize = 2147483637;\n }\n } else if (browserDetails.version < 60) {\n // Currently, all FF >= 57 will reset the remote maximum message size\n // to the default value when a data channel is created at a later\n // stage. :(\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n canSendMaxMessageSize =\n browserDetails.version === 57 ? 65535 : 65536;\n } else {\n // FF >= 60 supports sending ~2 GiB\n canSendMaxMessageSize = 2147483637;\n }\n }\n return canSendMaxMessageSize;\n };\n\n const getMaxMessageSize = function(description, remoteIsFirefox) {\n // Note: 65536 bytes is the default value from the SDP spec. Also,\n // every implementation we know supports receiving 65536 bytes.\n let maxMessageSize = 65536;\n\n // FF 57 has a slightly incorrect default remote max message size, so\n // we need to adjust it here to avoid a failure when sending.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697\n if (browserDetails.browser === 'firefox'\n && browserDetails.version === 57) {\n maxMessageSize = 65535;\n }\n\n const match = SDPUtils.matchPrefix(description.sdp,\n 'a=max-message-size:');\n if (match.length > 0) {\n maxMessageSize = parseInt(match[0].substr(19), 10);\n } else if (browserDetails.browser === 'firefox' &&\n remoteIsFirefox !== -1) {\n // If the maximum message size is not present in the remote SDP and\n // both local and remote are Firefox, the remote peer can receive\n // ~2 GiB.\n maxMessageSize = 2147483637;\n }\n return maxMessageSize;\n };\n\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n this._sctp = null;\n // Chrome decided to not expose .sctp in plan-b mode.\n // As usual, adapter.js has to do an 'ugly worakaround'\n // to cover up the mess.\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) {\n const {sdpSemantics} = this.getConfiguration();\n if (sdpSemantics === 'plan-b') {\n Object.defineProperty(this, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n },\n enumerable: true,\n configurable: true,\n });\n }\n }\n\n if (sctpInDescription(arguments[0])) {\n // Check if the remote is FF.\n const isFirefox = getRemoteFirefoxVersion(arguments[0]);\n\n // Get the maximum message size the local peer is capable of sending\n const canSendMMS = getCanSendMaxMessageSize(isFirefox);\n\n // Get the maximum message size of the remote peer.\n const remoteMMS = getMaxMessageSize(arguments[0], isFirefox);\n\n // Determine final maximum message size\n let maxMessageSize;\n if (canSendMMS === 0 && remoteMMS === 0) {\n maxMessageSize = Number.POSITIVE_INFINITY;\n } else if (canSendMMS === 0 || remoteMMS === 0) {\n maxMessageSize = Math.max(canSendMMS, remoteMMS);\n } else {\n maxMessageSize = Math.min(canSendMMS, remoteMMS);\n }\n\n // Create a dummy RTCSctpTransport object and the 'maxMessageSize'\n // attribute.\n const sctp = {};\n Object.defineProperty(sctp, 'maxMessageSize', {\n get() {\n return maxMessageSize;\n }\n });\n this._sctp = sctp;\n }\n\n return origSetRemoteDescription.apply(this, arguments);\n };\n}\n\nexport function shimSendThrowTypeError(window) {\n if (!(window.RTCPeerConnection &&\n 'createDataChannel' in window.RTCPeerConnection.prototype)) {\n return;\n }\n\n // Note: Although Firefox >= 57 has a native implementation, the maximum\n // message size can be reset for all data channels at a later stage.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n\n function wrapDcSend(dc, pc) {\n const origDataChannelSend = dc.send;\n dc.send = function send() {\n const data = arguments[0];\n const length = data.length || data.size || data.byteLength;\n if (dc.readyState === 'open' &&\n pc.sctp && length > pc.sctp.maxMessageSize) {\n throw new TypeError('Message too large (can send a maximum of ' +\n pc.sctp.maxMessageSize + ' bytes)');\n }\n return origDataChannelSend.apply(dc, arguments);\n };\n }\n const origCreateDataChannel =\n window.RTCPeerConnection.prototype.createDataChannel;\n window.RTCPeerConnection.prototype.createDataChannel =\n function createDataChannel() {\n const dataChannel = origCreateDataChannel.apply(this, arguments);\n wrapDcSend(dataChannel, this);\n return dataChannel;\n };\n utils.wrapPeerConnectionEvent(window, 'datachannel', e => {\n wrapDcSend(e.channel, e.target);\n return e;\n });\n}\n\n\n/* shims RTCConnectionState by pretending it is the same as iceConnectionState.\n * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12\n * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect\n * since DTLS failures would be hidden. See\n * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827\n * for the Firefox tracking bug.\n */\nexport function shimConnectionState(window) {\n if (!window.RTCPeerConnection ||\n 'connectionState' in window.RTCPeerConnection.prototype) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n Object.defineProperty(proto, 'connectionState', {\n get() {\n return {\n completed: 'connected',\n checking: 'connecting'\n }[this.iceConnectionState] || this.iceConnectionState;\n },\n enumerable: true,\n configurable: true\n });\n Object.defineProperty(proto, 'onconnectionstatechange', {\n get() {\n return this._onconnectionstatechange || null;\n },\n set(cb) {\n if (this._onconnectionstatechange) {\n this.removeEventListener('connectionstatechange',\n this._onconnectionstatechange);\n delete this._onconnectionstatechange;\n }\n if (cb) {\n this.addEventListener('connectionstatechange',\n this._onconnectionstatechange = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n\n ['setLocalDescription', 'setRemoteDescription'].forEach((method) => {\n const origMethod = proto[method];\n proto[method] = function() {\n if (!this._connectionstatechangepoly) {\n this._connectionstatechangepoly = e => {\n const pc = e.target;\n if (pc._lastConnectionState !== pc.connectionState) {\n pc._lastConnectionState = pc.connectionState;\n const newEvent = new Event('connectionstatechange', e);\n pc.dispatchEvent(newEvent);\n }\n return e;\n };\n this.addEventListener('iceconnectionstatechange',\n this._connectionstatechangepoly);\n }\n return origMethod.apply(this, arguments);\n };\n });\n}\n\nexport function removeExtmapAllowMixed(window, browserDetails) {\n /* remove a=extmap-allow-mixed for webrtc.org < M71 */\n if (!window.RTCPeerConnection) {\n return;\n }\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) {\n return;\n }\n if (browserDetails.browser === 'safari' && browserDetails.version >= 605) {\n return;\n }\n const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription(desc) {\n if (desc && desc.sdp && desc.sdp.indexOf('\\na=extmap-allow-mixed') !== -1) {\n const sdp = desc.sdp.split('\\n').filter((line) => {\n return line.trim() !== 'a=extmap-allow-mixed';\n }).join('\\n');\n // Safari enforces read-only-ness of RTCSessionDescription fields.\n if (window.RTCSessionDescription &&\n desc instanceof window.RTCSessionDescription) {\n arguments[0] = new window.RTCSessionDescription({\n type: desc.type,\n sdp,\n });\n } else {\n desc.sdp = sdp;\n }\n }\n return nativeSRD.apply(this, arguments);\n };\n}\n\nexport function shimAddIceCandidateNullOrEmpty(window, browserDetails) {\n // Support for addIceCandidate(null or undefined)\n // as well as addIceCandidate({candidate: \"\", ...})\n // https://bugs.chromium.org/p/chromium/issues/detail?id=978582\n // Note: must be called before other polyfills which change the signature.\n if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) {\n return;\n }\n const nativeAddIceCandidate =\n window.RTCPeerConnection.prototype.addIceCandidate;\n if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) {\n return;\n }\n window.RTCPeerConnection.prototype.addIceCandidate =\n function addIceCandidate() {\n if (!arguments[0]) {\n if (arguments[1]) {\n arguments[1].apply(null);\n }\n return Promise.resolve();\n }\n // Firefox 68+ emits and processes {candidate: \"\", ...}, ignore\n // in older versions.\n // Native support for ignoring exists for Chrome M77+.\n // Safari ignores as well, exact version unknown but works in the same\n // version that also ignores addIceCandidate(null).\n if (((browserDetails.browser === 'chrome' && browserDetails.version < 78)\n || (browserDetails.browser === 'firefox'\n && browserDetails.version < 68)\n || (browserDetails.browser === 'safari'))\n && arguments[0] && arguments[0].candidate === '') {\n return Promise.resolve();\n }\n return nativeAddIceCandidate.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\nimport * as utils from './utils';\n\n // Browser shims.\nimport * as chromeShim from './chrome/chrome_shim';\nimport * as edgeShim from './edge/edge_shim';\nimport * as firefoxShim from './firefox/firefox_shim';\nimport * as safariShim from './safari/safari_shim';\nimport * as commonShim from './common_shim';\n\n// Shimming starts here.\nexport function adapterFactory({window} = {}, options = {\n shimChrome: true,\n shimFirefox: true,\n shimEdge: true,\n shimSafari: true,\n}) {\n // Utils.\n const logging = utils.log;\n const browserDetails = utils.detectBrowser(window);\n\n const adapter = {\n browserDetails,\n commonShim,\n extractVersion: utils.extractVersion,\n disableLog: utils.disableLog,\n disableWarnings: utils.disableWarnings\n };\n\n // Shim browser if found.\n switch (browserDetails.browser) {\n case 'chrome':\n if (!chromeShim || !chromeShim.shimPeerConnection ||\n !options.shimChrome) {\n logging('Chrome shim is not included in this adapter release.');\n return adapter;\n }\n if (browserDetails.version === null) {\n logging('Chrome shim can not determine version, not shimming.');\n return adapter;\n }\n logging('adapter.js shimming chrome.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = chromeShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n chromeShim.shimGetUserMedia(window, browserDetails);\n chromeShim.shimMediaStream(window, browserDetails);\n chromeShim.shimPeerConnection(window, browserDetails);\n chromeShim.shimOnTrack(window, browserDetails);\n chromeShim.shimAddTrackRemoveTrack(window, browserDetails);\n chromeShim.shimGetSendersWithDtmf(window, browserDetails);\n chromeShim.shimGetStats(window, browserDetails);\n chromeShim.shimSenderReceiverGetStats(window, browserDetails);\n chromeShim.fixNegotiationNeeded(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n case 'firefox':\n if (!firefoxShim || !firefoxShim.shimPeerConnection ||\n !options.shimFirefox) {\n logging('Firefox shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming firefox.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = firefoxShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n firefoxShim.shimGetUserMedia(window, browserDetails);\n firefoxShim.shimPeerConnection(window, browserDetails);\n firefoxShim.shimOnTrack(window, browserDetails);\n firefoxShim.shimRemoveStream(window, browserDetails);\n firefoxShim.shimSenderGetStats(window, browserDetails);\n firefoxShim.shimReceiverGetStats(window, browserDetails);\n firefoxShim.shimRTCDataChannel(window, browserDetails);\n firefoxShim.shimAddTransceiver(window, browserDetails);\n firefoxShim.shimGetParameters(window, browserDetails);\n firefoxShim.shimCreateOffer(window, browserDetails);\n firefoxShim.shimCreateAnswer(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'edge':\n if (!edgeShim || !edgeShim.shimPeerConnection || !options.shimEdge) {\n logging('MS edge shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming edge.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = edgeShim;\n\n edgeShim.shimGetUserMedia(window, browserDetails);\n edgeShim.shimGetDisplayMedia(window, browserDetails);\n edgeShim.shimPeerConnection(window, browserDetails);\n edgeShim.shimReplaceTrack(window, browserDetails);\n\n // the edge shim implements the full RTCIceCandidate object.\n\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'safari':\n if (!safariShim || !options.shimSafari) {\n logging('Safari shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming safari.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = safariShim;\n\n // Must be called before shimCallbackAPI.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n safariShim.shimRTCIceServerUrls(window, browserDetails);\n safariShim.shimCreateOfferLegacy(window, browserDetails);\n safariShim.shimCallbacksAPI(window, browserDetails);\n safariShim.shimLocalStreamsAPI(window, browserDetails);\n safariShim.shimRemoteStreamsAPI(window, browserDetails);\n safariShim.shimTrackEventTransceiver(window, browserDetails);\n safariShim.shimGetUserMedia(window, browserDetails);\n safariShim.shimAudioContext(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n default:\n logging('Unsupported browser!');\n break;\n }\n\n return adapter;\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n\n'use strict';\n\nimport {adapterFactory} from './adapter_factory.js';\n\nconst adapter =\n adapterFactory({window: typeof window === 'undefined' ? undefined : window});\nexport default adapter;\n","\n'use strict';\nimport * as utils from './utils.js';\nimport * as MediaFormatModule from './mediaformat.js';\nimport adapter from 'webrtc-adapter';\n\n/**\n * @class AudioTrackConstraints\n * @classDesc Constraints for creating an audio MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track.\n */\nexport class AudioTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.AudioSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Values could be \"mic\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Do not provide deviceId if source is not \"mic\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n this.deviceId = undefined;\n }\n}\n\n/**\n * @class VideoTrackConstraints\n * @classDesc Constraints for creating a video MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.VideoSourceInfo} source Source info of this video track.\n */\nexport class VideoTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.VideoSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Values could be \"camera\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Do not provide deviceId if source is not \"camera\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n\n this.deviceId = undefined;\n\n /**\n * @member {Owt.Base.Resolution} resolution\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.resolution = undefined;\n\n /**\n * @member {number} frameRate\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.frameRate = undefined;\n }\n}\n/**\n * @class StreamConstraints\n * @classDesc Constraints for creating a MediaStream from screen mic and camera.\n * @memberof Owt.Base\n * @constructor\n * @param {?Owt.Base.AudioTrackConstraints} audioConstraints\n * @param {?Owt.Base.VideoTrackConstraints} videoConstraints\n */\nexport class StreamConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(audioConstraints = false, videoConstraints = false) {\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.audio = audioConstraints;\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.video = videoConstraints;\n }\n}\n\n// eslint-disable-next-line require-jsdoc\nfunction isVideoConstrainsForScreenCast(constraints) {\n return (typeof constraints.video === 'object' && constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST);\n}\n\n/**\n * @class MediaStreamFactory\n * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself.\n * @memberof Owt.Base\n */\nexport class MediaStreamFactory {\n /**\n * @function createMediaStream\n * @static\n * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are \"screen-cast\".\n * @memberof Owt.Base.MediaStreamFactory\n * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened:\n * - One or more parameters cannot be satisfied.\n * - Specified device is busy.\n * - Cannot obtain necessary permission or operation is canceled by user.\n * - Video source is screen cast, while audio source is not.\n * - Audio source is screen cast, while video source is disabled.\n * @param {Owt.Base.StreamConstraints} constraints\n */\n static createMediaStream(constraints) {\n if (typeof constraints !== 'object' ||\n (!constraints.audio && !constraints.video)) {\n return Promise.reject(new TypeError('Invalid constrains'));\n }\n if (!isVideoConstrainsForScreenCast(constraints) &&\n (typeof constraints.audio === 'object') &&\n constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(\n new TypeError('Cannot share screen without video.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) && !utils.isChrome() &&\n !utils.isFirefox()) {\n return Promise.reject(\n new TypeError('Screen sharing only supports Chrome and Firefox.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) &&\n typeof constraints.audio === 'object' &&\n constraints.audio.source !==\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(new TypeError(\n 'Cannot capture video from screen cast while capture audio from'\n + ' other source.'));\n }\n\n // Check and convert constraints.\n if (!constraints.audio && !constraints.video) {\n return Promise.reject(new TypeError(\n 'At least one of audio and video must be requested.'));\n }\n const mediaConstraints = Object.create({});\n if (typeof constraints.audio === 'object' &&\n constraints.audio.source === MediaFormatModule.AudioSourceInfo.MIC) {\n mediaConstraints.audio = Object.create({});\n if (utils.isEdge()) {\n mediaConstraints.audio.deviceId = constraints.audio.deviceId;\n } else {\n mediaConstraints.audio.deviceId = {\n exact: constraints.audio.deviceId,\n };\n }\n } else {\n if (constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n mediaConstraints.audio = true;\n } else {\n mediaConstraints.audio = constraints.audio;\n }\n }\n if (typeof constraints.video === 'object') {\n mediaConstraints.video = Object.create({});\n if (typeof constraints.video.frameRate === 'number') {\n mediaConstraints.video.frameRate = constraints.video.frameRate;\n }\n if (constraints.video.resolution &&\n constraints.video.resolution.width &&\n constraints.video.resolution.height) {\n if (constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.width = constraints.video.resolution.width;\n mediaConstraints.video.height = constraints.video.resolution.height;\n } else {\n mediaConstraints.video.width = Object.create({});\n mediaConstraints.video.width.exact =\n constraints.video.resolution.width;\n mediaConstraints.video.height = Object.create({});\n mediaConstraints.video.height.exact =\n constraints.video.resolution.height;\n }\n }\n if (typeof constraints.video.deviceId === 'string') {\n mediaConstraints.video.deviceId = {exact: constraints.video.deviceId};\n }\n if (utils.isFirefox() &&\n constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.mediaSource = 'screen';\n }\n } else {\n mediaConstraints.video = constraints.video;\n }\n\n if (isVideoConstrainsForScreenCast(constraints)) {\n return navigator.mediaDevices.getDisplayMedia(mediaConstraints);\n } else {\n return navigator.mediaDevices.getUserMedia(mediaConstraints);\n }\n }\n}\n","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n\nexport * from './mediastream-factory.js';\nexport * from './mediaformat.js';","let logger;\nlet errorLogger;\n\nexport function setLogger() {\n /*eslint-disable */\n logger = console.log;\n errorLogger = console.error;\n /*eslint-enable */\n}\n\nexport function isEnable() {\n return logger != null;\n}\n\nexport function log(message, ...optionalParams) {\n if (logger) {\n logger(message, ...optionalParams);\n }\n}\nexport function error(message, ...optionalParams) {\n if (errorLogger) {\n errorLogger(message, ...optionalParams);\n }\n}\n","export default class Event {\n constructor(type) {\n this.listener = {};\n this.type = type | '';\n }\n\n on(event, fn) {\n if (!this.listener[event]) {\n this.listener[event] = [];\n }\n this.listener[event].push(fn);\n return true;\n }\n\n off(event, fn) {\n if (this.listener[event]) {\n var index = this.listener[event].indexOf(fn);\n if (index > -1) {\n this.listener[event].splice(index, 1);\n }\n return true;\n }\n return false;\n }\n\n offAll() {\n this.listener = {};\n }\n\n dispatch(event, data) {\n if (this.listener[event]) {\n this.listener[event].map((each) => {\n each.apply(null, [data]);\n });\n return true;\n }\n return false;\n }\n}\n","'use strict';\n\nmodule.exports = function bind(fn, thisArg) {\n return function wrap() {\n var args = new Array(arguments.length);\n for (var i = 0; i < args.length; i++) {\n args[i] = arguments[i];\n }\n return fn.apply(thisArg, args);\n };\n};\n","'use strict';\n\nvar bind = require('./helpers/bind');\n\n/*global toString:true*/\n\n// utils is a library of generic helper functions non-specific to axios\n\nvar toString = Object.prototype.toString;\n\n/**\n * Determine if a value is an Array\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Array, otherwise false\n */\nfunction isArray(val) {\n return toString.call(val) === '[object Array]';\n}\n\n/**\n * Determine if a value is undefined\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if the value is undefined, otherwise false\n */\nfunction isUndefined(val) {\n return typeof val === 'undefined';\n}\n\n/**\n * Determine if a value is a Buffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Buffer, otherwise false\n */\nfunction isBuffer(val) {\n return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor)\n && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val);\n}\n\n/**\n * Determine if a value is an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an ArrayBuffer, otherwise false\n */\nfunction isArrayBuffer(val) {\n return toString.call(val) === '[object ArrayBuffer]';\n}\n\n/**\n * Determine if a value is a FormData\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an FormData, otherwise false\n */\nfunction isFormData(val) {\n return (typeof FormData !== 'undefined') && (val instanceof FormData);\n}\n\n/**\n * Determine if a value is a view on an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false\n */\nfunction isArrayBufferView(val) {\n var result;\n if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) {\n result = ArrayBuffer.isView(val);\n } else {\n result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer);\n }\n return result;\n}\n\n/**\n * Determine if a value is a String\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a String, otherwise false\n */\nfunction isString(val) {\n return typeof val === 'string';\n}\n\n/**\n * Determine if a value is a Number\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Number, otherwise false\n */\nfunction isNumber(val) {\n return typeof val === 'number';\n}\n\n/**\n * Determine if a value is an Object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Object, otherwise false\n */\nfunction isObject(val) {\n return val !== null && typeof val === 'object';\n}\n\n/**\n * Determine if a value is a plain Object\n *\n * @param {Object} val The value to test\n * @return {boolean} True if value is a plain Object, otherwise false\n */\nfunction isPlainObject(val) {\n if (toString.call(val) !== '[object Object]') {\n return false;\n }\n\n var prototype = Object.getPrototypeOf(val);\n return prototype === null || prototype === Object.prototype;\n}\n\n/**\n * Determine if a value is a Date\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Date, otherwise false\n */\nfunction isDate(val) {\n return toString.call(val) === '[object Date]';\n}\n\n/**\n * Determine if a value is a File\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a File, otherwise false\n */\nfunction isFile(val) {\n return toString.call(val) === '[object File]';\n}\n\n/**\n * Determine if a value is a Blob\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Blob, otherwise false\n */\nfunction isBlob(val) {\n return toString.call(val) === '[object Blob]';\n}\n\n/**\n * Determine if a value is a Function\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Function, otherwise false\n */\nfunction isFunction(val) {\n return toString.call(val) === '[object Function]';\n}\n\n/**\n * Determine if a value is a Stream\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Stream, otherwise false\n */\nfunction isStream(val) {\n return isObject(val) && isFunction(val.pipe);\n}\n\n/**\n * Determine if a value is a URLSearchParams object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a URLSearchParams object, otherwise false\n */\nfunction isURLSearchParams(val) {\n return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams;\n}\n\n/**\n * Trim excess whitespace off the beginning and end of a string\n *\n * @param {String} str The String to trim\n * @returns {String} The String freed of excess whitespace\n */\nfunction trim(str) {\n return str.replace(/^\\s*/, '').replace(/\\s*$/, '');\n}\n\n/**\n * Determine if we're running in a standard browser environment\n *\n * This allows axios to run in a web worker, and react-native.\n * Both environments support XMLHttpRequest, but not fully standard globals.\n *\n * web workers:\n * typeof window -> undefined\n * typeof document -> undefined\n *\n * react-native:\n * navigator.product -> 'ReactNative'\n * nativescript\n * navigator.product -> 'NativeScript' or 'NS'\n */\nfunction isStandardBrowserEnv() {\n if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' ||\n navigator.product === 'NativeScript' ||\n navigator.product === 'NS')) {\n return false;\n }\n return (\n typeof window !== 'undefined' &&\n typeof document !== 'undefined'\n );\n}\n\n/**\n * Iterate over an Array or an Object invoking a function for each item.\n *\n * If `obj` is an Array callback will be called passing\n * the value, index, and complete array for each item.\n *\n * If 'obj' is an Object callback will be called passing\n * the value, key, and complete object for each property.\n *\n * @param {Object|Array} obj The object to iterate\n * @param {Function} fn The callback to invoke for each item\n */\nfunction forEach(obj, fn) {\n // Don't bother if no value provided\n if (obj === null || typeof obj === 'undefined') {\n return;\n }\n\n // Force an array if not already something iterable\n if (typeof obj !== 'object') {\n /*eslint no-param-reassign:0*/\n obj = [obj];\n }\n\n if (isArray(obj)) {\n // Iterate over array values\n for (var i = 0, l = obj.length; i < l; i++) {\n fn.call(null, obj[i], i, obj);\n }\n } else {\n // Iterate over object keys\n for (var key in obj) {\n if (Object.prototype.hasOwnProperty.call(obj, key)) {\n fn.call(null, obj[key], key, obj);\n }\n }\n }\n}\n\n/**\n * Accepts varargs expecting each argument to be an object, then\n * immutably merges the properties of each object and returns result.\n *\n * When multiple objects contain the same key the later object in\n * the arguments list will take precedence.\n *\n * Example:\n *\n * ```js\n * var result = merge({foo: 123}, {foo: 456});\n * console.log(result.foo); // outputs 456\n * ```\n *\n * @param {Object} obj1 Object to merge\n * @returns {Object} Result of all merge properties\n */\nfunction merge(/* obj1, obj2, obj3, ... */) {\n var result = {};\n function assignValue(val, key) {\n if (isPlainObject(result[key]) && isPlainObject(val)) {\n result[key] = merge(result[key], val);\n } else if (isPlainObject(val)) {\n result[key] = merge({}, val);\n } else if (isArray(val)) {\n result[key] = val.slice();\n } else {\n result[key] = val;\n }\n }\n\n for (var i = 0, l = arguments.length; i < l; i++) {\n forEach(arguments[i], assignValue);\n }\n return result;\n}\n\n/**\n * Extends object a by mutably adding to it the properties of object b.\n *\n * @param {Object} a The object to be extended\n * @param {Object} b The object to copy properties from\n * @param {Object} thisArg The object to bind function to\n * @return {Object} The resulting value of object a\n */\nfunction extend(a, b, thisArg) {\n forEach(b, function assignValue(val, key) {\n if (thisArg && typeof val === 'function') {\n a[key] = bind(val, thisArg);\n } else {\n a[key] = val;\n }\n });\n return a;\n}\n\n/**\n * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM)\n *\n * @param {string} content with BOM\n * @return {string} content value without BOM\n */\nfunction stripBOM(content) {\n if (content.charCodeAt(0) === 0xFEFF) {\n content = content.slice(1);\n }\n return content;\n}\n\nmodule.exports = {\n isArray: isArray,\n isArrayBuffer: isArrayBuffer,\n isBuffer: isBuffer,\n isFormData: isFormData,\n isArrayBufferView: isArrayBufferView,\n isString: isString,\n isNumber: isNumber,\n isObject: isObject,\n isPlainObject: isPlainObject,\n isUndefined: isUndefined,\n isDate: isDate,\n isFile: isFile,\n isBlob: isBlob,\n isFunction: isFunction,\n isStream: isStream,\n isURLSearchParams: isURLSearchParams,\n isStandardBrowserEnv: isStandardBrowserEnv,\n forEach: forEach,\n merge: merge,\n extend: extend,\n trim: trim,\n stripBOM: stripBOM\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction encode(val) {\n return encodeURIComponent(val).\n replace(/%3A/gi, ':').\n replace(/%24/g, '$').\n replace(/%2C/gi, ',').\n replace(/%20/g, '+').\n replace(/%5B/gi, '[').\n replace(/%5D/gi, ']');\n}\n\n/**\n * Build a URL by appending params to the end\n *\n * @param {string} url The base of the url (e.g., http://www.google.com)\n * @param {object} [params] The params to be appended\n * @returns {string} The formatted url\n */\nmodule.exports = function buildURL(url, params, paramsSerializer) {\n /*eslint no-param-reassign:0*/\n if (!params) {\n return url;\n }\n\n var serializedParams;\n if (paramsSerializer) {\n serializedParams = paramsSerializer(params);\n } else if (utils.isURLSearchParams(params)) {\n serializedParams = params.toString();\n } else {\n var parts = [];\n\n utils.forEach(params, function serialize(val, key) {\n if (val === null || typeof val === 'undefined') {\n return;\n }\n\n if (utils.isArray(val)) {\n key = key + '[]';\n } else {\n val = [val];\n }\n\n utils.forEach(val, function parseValue(v) {\n if (utils.isDate(v)) {\n v = v.toISOString();\n } else if (utils.isObject(v)) {\n v = JSON.stringify(v);\n }\n parts.push(encode(key) + '=' + encode(v));\n });\n });\n\n serializedParams = parts.join('&');\n }\n\n if (serializedParams) {\n var hashmarkIndex = url.indexOf('#');\n if (hashmarkIndex !== -1) {\n url = url.slice(0, hashmarkIndex);\n }\n\n url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams;\n }\n\n return url;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction InterceptorManager() {\n this.handlers = [];\n}\n\n/**\n * Add a new interceptor to the stack\n *\n * @param {Function} fulfilled The function to handle `then` for a `Promise`\n * @param {Function} rejected The function to handle `reject` for a `Promise`\n *\n * @return {Number} An ID used to remove interceptor later\n */\nInterceptorManager.prototype.use = function use(fulfilled, rejected) {\n this.handlers.push({\n fulfilled: fulfilled,\n rejected: rejected\n });\n return this.handlers.length - 1;\n};\n\n/**\n * Remove an interceptor from the stack\n *\n * @param {Number} id The ID that was returned by `use`\n */\nInterceptorManager.prototype.eject = function eject(id) {\n if (this.handlers[id]) {\n this.handlers[id] = null;\n }\n};\n\n/**\n * Iterate over all the registered interceptors\n *\n * This method is particularly useful for skipping over any\n * interceptors that may have become `null` calling `eject`.\n *\n * @param {Function} fn The function to call for each interceptor\n */\nInterceptorManager.prototype.forEach = function forEach(fn) {\n utils.forEach(this.handlers, function forEachHandler(h) {\n if (h !== null) {\n fn(h);\n }\n });\n};\n\nmodule.exports = InterceptorManager;\n","'use strict';\n\nvar utils = require('./../utils');\n\n/**\n * Transform the data for a request or a response\n *\n * @param {Object|String} data The data to be transformed\n * @param {Array} headers The headers for the request or response\n * @param {Array|Function} fns A single function or Array of functions\n * @returns {*} The resulting transformed data\n */\nmodule.exports = function transformData(data, headers, fns) {\n /*eslint no-param-reassign:0*/\n utils.forEach(fns, function transform(fn) {\n data = fn(data, headers);\n });\n\n return data;\n};\n","'use strict';\n\nmodule.exports = function isCancel(value) {\n return !!(value && value.__CANCEL__);\n};\n","'use strict';\n\nvar utils = require('../utils');\n\nmodule.exports = function normalizeHeaderName(headers, normalizedName) {\n utils.forEach(headers, function processHeader(value, name) {\n if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) {\n headers[normalizedName] = value;\n delete headers[name];\n }\n });\n};\n","'use strict';\n\n/**\n * Update an Error with the specified config, error code, and response.\n *\n * @param {Error} error The error to update.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The error.\n */\nmodule.exports = function enhanceError(error, config, code, request, response) {\n error.config = config;\n if (code) {\n error.code = code;\n }\n\n error.request = request;\n error.response = response;\n error.isAxiosError = true;\n\n error.toJSON = function toJSON() {\n return {\n // Standard\n message: this.message,\n name: this.name,\n // Microsoft\n description: this.description,\n number: this.number,\n // Mozilla\n fileName: this.fileName,\n lineNumber: this.lineNumber,\n columnNumber: this.columnNumber,\n stack: this.stack,\n // Axios\n config: this.config,\n code: this.code\n };\n };\n return error;\n};\n","'use strict';\n\nvar enhanceError = require('./enhanceError');\n\n/**\n * Create an Error with the specified message, config, error code, request and response.\n *\n * @param {string} message The error message.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The created error.\n */\nmodule.exports = function createError(message, config, code, request, response) {\n var error = new Error(message);\n return enhanceError(error, config, code, request, response);\n};\n","'use strict';\n\nvar createError = require('./createError');\n\n/**\n * Resolve or reject a Promise based on response status.\n *\n * @param {Function} resolve A function that resolves the promise.\n * @param {Function} reject A function that rejects the promise.\n * @param {object} response The response.\n */\nmodule.exports = function settle(resolve, reject, response) {\n var validateStatus = response.config.validateStatus;\n if (!response.status || !validateStatus || validateStatus(response.status)) {\n resolve(response);\n } else {\n reject(createError(\n 'Request failed with status code ' + response.status,\n response.config,\n null,\n response.request,\n response\n ));\n }\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs support document.cookie\n (function standardBrowserEnv() {\n return {\n write: function write(name, value, expires, path, domain, secure) {\n var cookie = [];\n cookie.push(name + '=' + encodeURIComponent(value));\n\n if (utils.isNumber(expires)) {\n cookie.push('expires=' + new Date(expires).toGMTString());\n }\n\n if (utils.isString(path)) {\n cookie.push('path=' + path);\n }\n\n if (utils.isString(domain)) {\n cookie.push('domain=' + domain);\n }\n\n if (secure === true) {\n cookie.push('secure');\n }\n\n document.cookie = cookie.join('; ');\n },\n\n read: function read(name) {\n var match = document.cookie.match(new RegExp('(^|;\\\\s*)(' + name + ')=([^;]*)'));\n return (match ? decodeURIComponent(match[3]) : null);\n },\n\n remove: function remove(name) {\n this.write(name, '', Date.now() - 86400000);\n }\n };\n })() :\n\n // Non standard browser env (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return {\n write: function write() {},\n read: function read() { return null; },\n remove: function remove() {}\n };\n })()\n);\n","'use strict';\n\n/**\n * Determines whether the specified URL is absolute\n *\n * @param {string} url The URL to test\n * @returns {boolean} True if the specified URL is absolute, otherwise false\n */\nmodule.exports = function isAbsoluteURL(url) {\n // A URL is considered absolute if it begins with \"://\" or \"//\" (protocol-relative URL).\n // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed\n // by any combination of letters, digits, plus, period, or hyphen.\n return /^([a-z][a-z\\d\\+\\-\\.]*:)?\\/\\//i.test(url);\n};\n","'use strict';\n\n/**\n * Creates a new URL by combining the specified URLs\n *\n * @param {string} baseURL The base URL\n * @param {string} relativeURL The relative URL\n * @returns {string} The combined URL\n */\nmodule.exports = function combineURLs(baseURL, relativeURL) {\n return relativeURL\n ? baseURL.replace(/\\/+$/, '') + '/' + relativeURL.replace(/^\\/+/, '')\n : baseURL;\n};\n","'use strict';\n\nvar isAbsoluteURL = require('../helpers/isAbsoluteURL');\nvar combineURLs = require('../helpers/combineURLs');\n\n/**\n * Creates a new URL by combining the baseURL with the requestedURL,\n * only when the requestedURL is not already an absolute URL.\n * If the requestURL is absolute, this function returns the requestedURL untouched.\n *\n * @param {string} baseURL The base URL\n * @param {string} requestedURL Absolute or relative URL to combine\n * @returns {string} The combined full path\n */\nmodule.exports = function buildFullPath(baseURL, requestedURL) {\n if (baseURL && !isAbsoluteURL(requestedURL)) {\n return combineURLs(baseURL, requestedURL);\n }\n return requestedURL;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\n// Headers whose duplicates are ignored by node\n// c.f. https://nodejs.org/api/http.html#http_message_headers\nvar ignoreDuplicateOf = [\n 'age', 'authorization', 'content-length', 'content-type', 'etag',\n 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since',\n 'last-modified', 'location', 'max-forwards', 'proxy-authorization',\n 'referer', 'retry-after', 'user-agent'\n];\n\n/**\n * Parse headers into an object\n *\n * ```\n * Date: Wed, 27 Aug 2014 08:58:49 GMT\n * Content-Type: application/json\n * Connection: keep-alive\n * Transfer-Encoding: chunked\n * ```\n *\n * @param {String} headers Headers needing to be parsed\n * @returns {Object} Headers parsed into an object\n */\nmodule.exports = function parseHeaders(headers) {\n var parsed = {};\n var key;\n var val;\n var i;\n\n if (!headers) { return parsed; }\n\n utils.forEach(headers.split('\\n'), function parser(line) {\n i = line.indexOf(':');\n key = utils.trim(line.substr(0, i)).toLowerCase();\n val = utils.trim(line.substr(i + 1));\n\n if (key) {\n if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) {\n return;\n }\n if (key === 'set-cookie') {\n parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]);\n } else {\n parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val;\n }\n }\n });\n\n return parsed;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs have full support of the APIs needed to test\n // whether the request URL is of the same origin as current location.\n (function standardBrowserEnv() {\n var msie = /(msie|trident)/i.test(navigator.userAgent);\n var urlParsingNode = document.createElement('a');\n var originURL;\n\n /**\n * Parse a URL to discover it's components\n *\n * @param {String} url The URL to be parsed\n * @returns {Object}\n */\n function resolveURL(url) {\n var href = url;\n\n if (msie) {\n // IE needs attribute set twice to normalize properties\n urlParsingNode.setAttribute('href', href);\n href = urlParsingNode.href;\n }\n\n urlParsingNode.setAttribute('href', href);\n\n // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils\n return {\n href: urlParsingNode.href,\n protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '',\n host: urlParsingNode.host,\n search: urlParsingNode.search ? urlParsingNode.search.replace(/^\\?/, '') : '',\n hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '',\n hostname: urlParsingNode.hostname,\n port: urlParsingNode.port,\n pathname: (urlParsingNode.pathname.charAt(0) === '/') ?\n urlParsingNode.pathname :\n '/' + urlParsingNode.pathname\n };\n }\n\n originURL = resolveURL(window.location.href);\n\n /**\n * Determine if a URL shares the same origin as the current location\n *\n * @param {String} requestURL The URL to test\n * @returns {boolean} True if URL shares the same origin, otherwise false\n */\n return function isURLSameOrigin(requestURL) {\n var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL;\n return (parsed.protocol === originURL.protocol &&\n parsed.host === originURL.host);\n };\n })() :\n\n // Non standard browser envs (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return function isURLSameOrigin() {\n return true;\n };\n })()\n);\n","'use strict';\n\nvar utils = require('./../utils');\nvar settle = require('./../core/settle');\nvar cookies = require('./../helpers/cookies');\nvar buildURL = require('./../helpers/buildURL');\nvar buildFullPath = require('../core/buildFullPath');\nvar parseHeaders = require('./../helpers/parseHeaders');\nvar isURLSameOrigin = require('./../helpers/isURLSameOrigin');\nvar createError = require('../core/createError');\n\nmodule.exports = function xhrAdapter(config) {\n return new Promise(function dispatchXhrRequest(resolve, reject) {\n var requestData = config.data;\n var requestHeaders = config.headers;\n\n if (utils.isFormData(requestData)) {\n delete requestHeaders['Content-Type']; // Let the browser set it\n }\n\n var request = new XMLHttpRequest();\n\n // HTTP basic authentication\n if (config.auth) {\n var username = config.auth.username || '';\n var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : '';\n requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password);\n }\n\n var fullPath = buildFullPath(config.baseURL, config.url);\n request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true);\n\n // Set the request timeout in MS\n request.timeout = config.timeout;\n\n // Listen for ready state\n request.onreadystatechange = function handleLoad() {\n if (!request || request.readyState !== 4) {\n return;\n }\n\n // The request errored out and we didn't get a response, this will be\n // handled by onerror instead\n // With one exception: request that using file: protocol, most browsers\n // will return status as 0 even though it's a successful request\n if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) {\n return;\n }\n\n // Prepare the response\n var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null;\n var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response;\n var response = {\n data: responseData,\n status: request.status,\n statusText: request.statusText,\n headers: responseHeaders,\n config: config,\n request: request\n };\n\n settle(resolve, reject, response);\n\n // Clean up request\n request = null;\n };\n\n // Handle browser request cancellation (as opposed to a manual cancellation)\n request.onabort = function handleAbort() {\n if (!request) {\n return;\n }\n\n reject(createError('Request aborted', config, 'ECONNABORTED', request));\n\n // Clean up request\n request = null;\n };\n\n // Handle low level network errors\n request.onerror = function handleError() {\n // Real errors are hidden from us by the browser\n // onerror should only fire if it's a network error\n reject(createError('Network Error', config, null, request));\n\n // Clean up request\n request = null;\n };\n\n // Handle timeout\n request.ontimeout = function handleTimeout() {\n var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded';\n if (config.timeoutErrorMessage) {\n timeoutErrorMessage = config.timeoutErrorMessage;\n }\n reject(createError(timeoutErrorMessage, config, 'ECONNABORTED',\n request));\n\n // Clean up request\n request = null;\n };\n\n // Add xsrf header\n // This is only done if running in a standard browser environment.\n // Specifically not if we're in a web worker, or react-native.\n if (utils.isStandardBrowserEnv()) {\n // Add xsrf header\n var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ?\n cookies.read(config.xsrfCookieName) :\n undefined;\n\n if (xsrfValue) {\n requestHeaders[config.xsrfHeaderName] = xsrfValue;\n }\n }\n\n // Add headers to the request\n if ('setRequestHeader' in request) {\n utils.forEach(requestHeaders, function setRequestHeader(val, key) {\n if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') {\n // Remove Content-Type if data is undefined\n delete requestHeaders[key];\n } else {\n // Otherwise add header to the request\n request.setRequestHeader(key, val);\n }\n });\n }\n\n // Add withCredentials to request if needed\n if (!utils.isUndefined(config.withCredentials)) {\n request.withCredentials = !!config.withCredentials;\n }\n\n // Add responseType to request if needed\n if (config.responseType) {\n try {\n request.responseType = config.responseType;\n } catch (e) {\n // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2.\n // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function.\n if (config.responseType !== 'json') {\n throw e;\n }\n }\n }\n\n // Handle progress if needed\n if (typeof config.onDownloadProgress === 'function') {\n request.addEventListener('progress', config.onDownloadProgress);\n }\n\n // Not all browsers support upload events\n if (typeof config.onUploadProgress === 'function' && request.upload) {\n request.upload.addEventListener('progress', config.onUploadProgress);\n }\n\n if (config.cancelToken) {\n // Handle cancellation\n config.cancelToken.promise.then(function onCanceled(cancel) {\n if (!request) {\n return;\n }\n\n request.abort();\n reject(cancel);\n // Clean up request\n request = null;\n });\n }\n\n if (!requestData) {\n requestData = null;\n }\n\n // Send the request\n request.send(requestData);\n });\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar normalizeHeaderName = require('./helpers/normalizeHeaderName');\n\nvar DEFAULT_CONTENT_TYPE = {\n 'Content-Type': 'application/x-www-form-urlencoded'\n};\n\nfunction setContentTypeIfUnset(headers, value) {\n if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) {\n headers['Content-Type'] = value;\n }\n}\n\nfunction getDefaultAdapter() {\n var adapter;\n if (typeof XMLHttpRequest !== 'undefined') {\n // For browsers use XHR adapter\n adapter = require('./adapters/xhr');\n } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') {\n // For node use HTTP adapter\n adapter = require('./adapters/http');\n }\n return adapter;\n}\n\nvar defaults = {\n adapter: getDefaultAdapter(),\n\n transformRequest: [function transformRequest(data, headers) {\n normalizeHeaderName(headers, 'Accept');\n normalizeHeaderName(headers, 'Content-Type');\n if (utils.isFormData(data) ||\n utils.isArrayBuffer(data) ||\n utils.isBuffer(data) ||\n utils.isStream(data) ||\n utils.isFile(data) ||\n utils.isBlob(data)\n ) {\n return data;\n }\n if (utils.isArrayBufferView(data)) {\n return data.buffer;\n }\n if (utils.isURLSearchParams(data)) {\n setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8');\n return data.toString();\n }\n if (utils.isObject(data)) {\n setContentTypeIfUnset(headers, 'application/json;charset=utf-8');\n return JSON.stringify(data);\n }\n return data;\n }],\n\n transformResponse: [function transformResponse(data) {\n /*eslint no-param-reassign:0*/\n if (typeof data === 'string') {\n try {\n data = JSON.parse(data);\n } catch (e) { /* Ignore */ }\n }\n return data;\n }],\n\n /**\n * A timeout in milliseconds to abort a request. If set to 0 (default) a\n * timeout is not created.\n */\n timeout: 0,\n\n xsrfCookieName: 'XSRF-TOKEN',\n xsrfHeaderName: 'X-XSRF-TOKEN',\n\n maxContentLength: -1,\n maxBodyLength: -1,\n\n validateStatus: function validateStatus(status) {\n return status >= 200 && status < 300;\n }\n};\n\ndefaults.headers = {\n common: {\n 'Accept': 'application/json, text/plain, */*'\n }\n};\n\nutils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) {\n defaults.headers[method] = {};\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE);\n});\n\nmodule.exports = defaults;\n","'use strict';\n\nvar utils = require('./../utils');\nvar transformData = require('./transformData');\nvar isCancel = require('../cancel/isCancel');\nvar defaults = require('../defaults');\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nfunction throwIfCancellationRequested(config) {\n if (config.cancelToken) {\n config.cancelToken.throwIfRequested();\n }\n}\n\n/**\n * Dispatch a request to the server using the configured adapter.\n *\n * @param {object} config The config that is to be used for the request\n * @returns {Promise} The Promise to be fulfilled\n */\nmodule.exports = function dispatchRequest(config) {\n throwIfCancellationRequested(config);\n\n // Ensure headers exist\n config.headers = config.headers || {};\n\n // Transform request data\n config.data = transformData(\n config.data,\n config.headers,\n config.transformRequest\n );\n\n // Flatten headers\n config.headers = utils.merge(\n config.headers.common || {},\n config.headers[config.method] || {},\n config.headers\n );\n\n utils.forEach(\n ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'],\n function cleanHeaderConfig(method) {\n delete config.headers[method];\n }\n );\n\n var adapter = config.adapter || defaults.adapter;\n\n return adapter(config).then(function onAdapterResolution(response) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n response.data = transformData(\n response.data,\n response.headers,\n config.transformResponse\n );\n\n return response;\n }, function onAdapterRejection(reason) {\n if (!isCancel(reason)) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n if (reason && reason.response) {\n reason.response.data = transformData(\n reason.response.data,\n reason.response.headers,\n config.transformResponse\n );\n }\n }\n\n return Promise.reject(reason);\n });\n};\n","'use strict';\n\nvar utils = require('../utils');\n\n/**\n * Config-specific merge-function which creates a new config-object\n * by merging two configuration objects together.\n *\n * @param {Object} config1\n * @param {Object} config2\n * @returns {Object} New object resulting from merging config2 to config1\n */\nmodule.exports = function mergeConfig(config1, config2) {\n // eslint-disable-next-line no-param-reassign\n config2 = config2 || {};\n var config = {};\n\n var valueFromConfig2Keys = ['url', 'method', 'data'];\n var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params'];\n var defaultToConfig2Keys = [\n 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer',\n 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName',\n 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress',\n 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent',\n 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding'\n ];\n var directMergeKeys = ['validateStatus'];\n\n function getMergedValue(target, source) {\n if (utils.isPlainObject(target) && utils.isPlainObject(source)) {\n return utils.merge(target, source);\n } else if (utils.isPlainObject(source)) {\n return utils.merge({}, source);\n } else if (utils.isArray(source)) {\n return source.slice();\n }\n return source;\n }\n\n function mergeDeepProperties(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n }\n\n utils.forEach(valueFromConfig2Keys, function valueFromConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n }\n });\n\n utils.forEach(mergeDeepPropertiesKeys, mergeDeepProperties);\n\n utils.forEach(defaultToConfig2Keys, function defaultToConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n utils.forEach(directMergeKeys, function merge(prop) {\n if (prop in config2) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (prop in config1) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n var axiosKeys = valueFromConfig2Keys\n .concat(mergeDeepPropertiesKeys)\n .concat(defaultToConfig2Keys)\n .concat(directMergeKeys);\n\n var otherKeys = Object\n .keys(config1)\n .concat(Object.keys(config2))\n .filter(function filterAxiosKeys(key) {\n return axiosKeys.indexOf(key) === -1;\n });\n\n utils.forEach(otherKeys, mergeDeepProperties);\n\n return config;\n};\n","'use strict';\n\nvar utils = require('./../utils');\nvar buildURL = require('../helpers/buildURL');\nvar InterceptorManager = require('./InterceptorManager');\nvar dispatchRequest = require('./dispatchRequest');\nvar mergeConfig = require('./mergeConfig');\n\n/**\n * Create a new instance of Axios\n *\n * @param {Object} instanceConfig The default config for the instance\n */\nfunction Axios(instanceConfig) {\n this.defaults = instanceConfig;\n this.interceptors = {\n request: new InterceptorManager(),\n response: new InterceptorManager()\n };\n}\n\n/**\n * Dispatch a request\n *\n * @param {Object} config The config specific for this request (merged with this.defaults)\n */\nAxios.prototype.request = function request(config) {\n /*eslint no-param-reassign:0*/\n // Allow for axios('example/url'[, config]) a la fetch API\n if (typeof config === 'string') {\n config = arguments[1] || {};\n config.url = arguments[0];\n } else {\n config = config || {};\n }\n\n config = mergeConfig(this.defaults, config);\n\n // Set config.method\n if (config.method) {\n config.method = config.method.toLowerCase();\n } else if (this.defaults.method) {\n config.method = this.defaults.method.toLowerCase();\n } else {\n config.method = 'get';\n }\n\n // Hook up interceptors middleware\n var chain = [dispatchRequest, undefined];\n var promise = Promise.resolve(config);\n\n this.interceptors.request.forEach(function unshiftRequestInterceptors(interceptor) {\n chain.unshift(interceptor.fulfilled, interceptor.rejected);\n });\n\n this.interceptors.response.forEach(function pushResponseInterceptors(interceptor) {\n chain.push(interceptor.fulfilled, interceptor.rejected);\n });\n\n while (chain.length) {\n promise = promise.then(chain.shift(), chain.shift());\n }\n\n return promise;\n};\n\nAxios.prototype.getUri = function getUri(config) {\n config = mergeConfig(this.defaults, config);\n return buildURL(config.url, config.params, config.paramsSerializer).replace(/^\\?/, '');\n};\n\n// Provide aliases for supported request methods\nutils.forEach(['delete', 'get', 'head', 'options'], function forEachMethodNoData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: (config || {}).data\n }));\n };\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, data, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: data\n }));\n };\n});\n\nmodule.exports = Axios;\n","'use strict';\n\n/**\n * A `Cancel` is an object that is thrown when an operation is canceled.\n *\n * @class\n * @param {string=} message The message.\n */\nfunction Cancel(message) {\n this.message = message;\n}\n\nCancel.prototype.toString = function toString() {\n return 'Cancel' + (this.message ? ': ' + this.message : '');\n};\n\nCancel.prototype.__CANCEL__ = true;\n\nmodule.exports = Cancel;\n","'use strict';\n\nvar Cancel = require('./Cancel');\n\n/**\n * A `CancelToken` is an object that can be used to request cancellation of an operation.\n *\n * @class\n * @param {Function} executor The executor function.\n */\nfunction CancelToken(executor) {\n if (typeof executor !== 'function') {\n throw new TypeError('executor must be a function.');\n }\n\n var resolvePromise;\n this.promise = new Promise(function promiseExecutor(resolve) {\n resolvePromise = resolve;\n });\n\n var token = this;\n executor(function cancel(message) {\n if (token.reason) {\n // Cancellation has already been requested\n return;\n }\n\n token.reason = new Cancel(message);\n resolvePromise(token.reason);\n });\n}\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nCancelToken.prototype.throwIfRequested = function throwIfRequested() {\n if (this.reason) {\n throw this.reason;\n }\n};\n\n/**\n * Returns an object that contains a new `CancelToken` and a function that, when called,\n * cancels the `CancelToken`.\n */\nCancelToken.source = function source() {\n var cancel;\n var token = new CancelToken(function executor(c) {\n cancel = c;\n });\n return {\n token: token,\n cancel: cancel\n };\n};\n\nmodule.exports = CancelToken;\n","'use strict';\n\n/**\n * Syntactic sugar for invoking a function and expanding an array for arguments.\n *\n * Common use case would be to use `Function.prototype.apply`.\n *\n * ```js\n * function f(x, y, z) {}\n * var args = [1, 2, 3];\n * f.apply(null, args);\n * ```\n *\n * With `spread` this example can be re-written.\n *\n * ```js\n * spread(function(x, y, z) {})([1, 2, 3]);\n * ```\n *\n * @param {Function} callback\n * @returns {Function}\n */\nmodule.exports = function spread(callback) {\n return function wrap(arr) {\n return callback.apply(null, arr);\n };\n};\n","'use strict';\n\n/**\n * Determines whether the payload is an error thrown by Axios\n *\n * @param {*} payload The value to test\n * @returns {boolean} True if the payload is an error thrown by Axios, otherwise false\n */\nmodule.exports = function isAxiosError(payload) {\n return (typeof payload === 'object') && (payload.isAxiosError === true);\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar bind = require('./helpers/bind');\nvar Axios = require('./core/Axios');\nvar mergeConfig = require('./core/mergeConfig');\nvar defaults = require('./defaults');\n\n/**\n * Create an instance of Axios\n *\n * @param {Object} defaultConfig The default config for the instance\n * @return {Axios} A new instance of Axios\n */\nfunction createInstance(defaultConfig) {\n var context = new Axios(defaultConfig);\n var instance = bind(Axios.prototype.request, context);\n\n // Copy axios.prototype to instance\n utils.extend(instance, Axios.prototype, context);\n\n // Copy context to instance\n utils.extend(instance, context);\n\n return instance;\n}\n\n// Create the default instance to be exported\nvar axios = createInstance(defaults);\n\n// Expose Axios class to allow class inheritance\naxios.Axios = Axios;\n\n// Factory for creating new instances\naxios.create = function create(instanceConfig) {\n return createInstance(mergeConfig(axios.defaults, instanceConfig));\n};\n\n// Expose Cancel & CancelToken\naxios.Cancel = require('./cancel/Cancel');\naxios.CancelToken = require('./cancel/CancelToken');\naxios.isCancel = require('./cancel/isCancel');\n\n// Expose all/spread\naxios.all = function all(promises) {\n return Promise.all(promises);\n};\naxios.spread = require('./helpers/spread');\n\n// Expose isAxiosError\naxios.isAxiosError = require('./helpers/isAxiosError');\n\nmodule.exports = axios;\n\n// Allow use of default import syntax in TypeScript\nmodule.exports.default = axios;\n","module.exports = require('./lib/axios');","\nimport { setLogger } from '../ulity/debug';\nimport * as debug from '../ulity/debug';\nimport Event from '../ulity/event';\nimport Events from '../base/event';\nimport axios from 'axios';\nimport * as Base from '../base/export';\n\nexport default class RTCEndpoint extends Event\n{\n constructor(options)\n {\n super('RTCPusherPlayer');\n this.TAG = '[RTCPusherPlayer]';\n\n let defaults = {\n element: '',// html video element\n debug: false,// if output debug log\n zlmsdpUrl:'',\n simulecast:false,\n useCamera:true,\n audioEnable:true,\n videoEnable:true,\n recvOnly:false,\n resolution:{w:0,h:0}\n };\n \n this.options = Object.assign({}, defaults, options);\n\n if(this.options.debug)\n {\n setLogger();\n }\n\n this.e = {\n onicecandidate:this._onIceCandidate.bind(this),\n ontrack:this._onTrack.bind(this),\n onicecandidateerror:this._onIceCandidateError.bind(this)\n };\n\n this._remoteStream = null;\n this._localStream = null;\n\n this.pc = new RTCPeerConnection(null);\n\n this.pc.onicecandidate = this.e.onicecandidate;\n this.pc.onicecandidateerror = this.e.onicecandidateerror;\n this.pc.ontrack = this.e.ontrack;\n\n if(!this.options.recvOnly && (this.options.audioEnable || this.options.videoEnable))\n this.start();\n else\n this.receive();\n \n }\n\n receive()\n {\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n //debug.error(this.TAG,'this not implement');\n const AudioTransceiverInit = {\n direction: 'recvonly',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'recvonly',\n sendEncodings:[],\n };\n\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n videoTransceiver = this.pc.addTransceiver('video',VideoTransceiverInit);\n \n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/anwser exchange \n this.dispatch(Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED,ret);\n return;\n }\n let anwser = {};\n anwser.sdp = ret.sdp;\n anwser.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n\n this.pc.setRemoteDescription(anwser).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n }\n\n start()\n {\n let videoConstraints = false;\n let audioConstraints = false;\n\n if(this.options.useCamera)\n {\n if(this.options.videoEnable)\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.CAMERA);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n }\n else\n {\n if(this.options.videoEnable)\n {\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.SCREENCAST);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.SCREENCAST);\n }\n else\n {\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n else\n {// error shared display media not only audio\n debug.error(this.TAG,'error paramter');\n }\n }\n \n }\n\n if(this.options.resolution.w !=0 && this.options.resolution.h!=0 && typeof videoConstraints == 'object'){\n videoConstraints.resolution = new Base.Resolution(this.options.resolution.w ,this.options.resolution.h);\n }\n\n Base.MediaStreamFactory.createMediaStream(new Base.StreamConstraints(\n audioConstraints, videoConstraints)).then(stream => {\n\n this._localStream = stream;\n\n this.dispatch(Events.WEBRTC_ON_LOCAL_STREAM,stream);\n\n const AudioTransceiverInit = {\n direction: 'sendrecv',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'sendrecv',\n sendEncodings:[],\n };\n \n if(this.options.simulecast && stream.getVideoTracks().length>0)\n {\n VideoTransceiverInit.sendEncodings = [\n {rid: 'q', active: true, scaleResolutionDownBy: 4.0},\n {rid: 'h', active: true, scaleResolutionDownBy: 2.0},\n {rid: 'f', active: true}\n ];\n }\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n if(stream.getAudioTracks().length>0)\n {\n audioTransceiver = this.pc.addTransceiver(stream.getAudioTracks()[0],\n AudioTransceiverInit);\n }\n else\n {\n AudioTransceiverInit.direction ='recvonly';\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n }\n \n if(stream.getVideoTracks().length>0)\n {\n videoTransceiver = this.pc.addTransceiver(stream.getVideoTracks()[0],\n VideoTransceiverInit);\n }\n else\n {\n VideoTransceiverInit.direction = 'recvonly';\n videoTransceiver = this.pc.addTransceiver('video',\n VideoTransceiverInit);\n }\n\n /*\n stream.getTracks().forEach((track,idx)=>{\n debug.log(this.TAG,track);\n this.pc.addTrack(track);\n });\n */\n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/anwser exchange \n this.dispatch(Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED,ret);\n return;\n }\n let anwser = {};\n anwser.sdp = ret.sdp;\n anwser.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n \n this.pc.setRemoteDescription(anwser).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n\n }).catch(e=>{\n this.dispatch(Events.CAPTURE_STREAM_FAILED);\n //debug.error(this.TAG,e);\n });\n \n //const offerOptions = {};\n /*\n if (typeof this.pc.addTransceiver === 'function') {\n // |direction| seems not working on Safari.\n this.pc.addTransceiver('audio', { direction: 'recvonly' });\n this.pc.addTransceiver('video', { direction: 'recvonly' });\n } else {\n offerOptions.offerToReceiveAudio = true;\n offerOptions.offerToReceiveVideo = true;\n }\n */\n\n\n\n }\n _onIceCandidate(event) {\n if (event.candidate) { \n debug.log('Remote ICE candidate: \\n ' + event.candidate.candidate);\n // Send the candidate to the remote peer\n }\n else {\n // All ICE candidates have been sent\n }\n }\n\n _onTrack(event){\n if(this.options.element && event.streams && event.streams.length>0)\n {\n this.options.element.srcObject = event.streams[0];\n this._remoteStream = event.streams[0];\n\n this.dispatch(Events.WEBRTC_ON_REMOTE_STREAMS,event);\n }\n else\n {\n debug.error('element pararm is failed');\n }\n }\n\n _onIceCandidateError(event){\n this.dispatch(Events.WEBRTC_ICE_CANDIDATE_ERROR,event);\n }\n\n close()\n {\n if(this.pc)\n {\n this.pc.close();\n this.pc=null;\n }\n\n if(this.options)\n {\n this.options=null;\n }\n\n if(this._localStream)\n {\n this._localStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n\n if(this._remoteStream)\n {\n this._remoteStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n }\n\n get remoteStream()\n {\n return this._remoteStream;\n }\n \n get localStream()\n {\n return this._localStream;\n }\n}\n","import * as mediaformat from './mediaformat';\nimport * as MediaFactory from './mediastream-factory';\n\n\nconst quickScan=[\n {\n 'label': '4K(UHD)',\n 'width': 3840,\n 'height': 2160\n },\n {\n 'label': '1080p(FHD)',\n 'width': 1920,\n 'height': 1080\n },\n {\n 'label': 'UXGA',\n 'width': 1600,\n 'height': 1200,\n 'ratio': '4:3'\n },\n {\n 'label': '720p(HD)',\n 'width': 1280,\n 'height': 720\n },\n {\n 'label': 'SVGA',\n 'width': 800,\n 'height': 600\n },\n {\n 'label': 'VGA',\n 'width': 640,\n 'height': 480\n },\n {\n 'label': '360p(nHD)',\n 'width': 640,\n 'height': 360\n },\n {\n 'label': 'CIF',\n 'width': 352,\n 'height': 288\n },\n {\n 'label': 'QVGA',\n 'width': 320,\n 'height': 240\n },\n {\n 'label': 'QCIF',\n 'width': 176,\n 'height': 144\n },\n {\n 'label': 'QQVGA',\n 'width': 160,\n 'height': 120\n }\n];\n\n\n\n\nexport default function GetSupportCameraResolutions(){\n return new Promise(function (resolve, reject) {\n let resolutions = [];\n let ok = 0;\n let err = 0;\n for (let i = 0; i < quickScan.length; ++i) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(quickScan[i].width, quickScan[i].height);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolutions.push(quickScan[i]);\n ok++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n }).catch(e => {\n err++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n });\n }\n });\n}\n\nexport function GetAllScanResolution()\n{\n return quickScan;\n}\nexport function isSupportResolution(w,h)\n{\n return new Promise(function (resolve, reject) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(w,h);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolve();\n }).catch(e => {\n reject(e);\n });\n });\n}","import * as events from './base/event';\nimport * as compile from './ulity/version';\nimport * as media from './base/export';\nimport * as endpoint from './endpoint/endpoint';\nimport * as resolution from './base/resolutionfind';\n\n\n\nconsole.log('build date:',compile.BUILD_DATE);\nconsole.log('version:',compile.VERSION);\n\nexport const Events = events.default;\nexport const Media = media;\nexport const Endpoint = endpoint.default;\nexport const GetSupportCameraResolutions = resolution.default;\nexport const GetAllScanResolution = resolution.GetAllScanResolution;\nexport const isSupportResolution = 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+{"version":3,"file":"ZLMRTCClient.js","sources":["../src/base/event.js","../src/ulity/version.js","../src/base/utils.js","../src/base/mediaformat.js","../node_modules/webrtc-adapter/src/js/utils.js","../node_modules/webrtc-adapter/src/js/chrome/getusermedia.js","../node_modules/webrtc-adapter/src/js/chrome/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/chrome/chrome_shim.js","../node_modules/webrtc-adapter/src/js/edge/filtericeservers.js","../node_modules/sdp/sdp.js","../node_modules/rtcpeerconnection-shim/rtcpeerconnection.js","../node_modules/webrtc-adapter/src/js/edge/getusermedia.js","../node_modules/webrtc-adapter/src/js/edge/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/edge/edge_shim.js","../node_modules/webrtc-adapter/src/js/firefox/getusermedia.js","../node_modules/webrtc-adapter/src/js/firefox/getdisplaymedia.js","../node_modules/webrtc-adapter/src/js/firefox/firefox_shim.js","../node_modules/webrtc-adapter/src/js/safari/safari_shim.js","../node_modules/webrtc-adapter/src/js/common_shim.js","../node_modules/webrtc-adapter/src/js/adapter_factory.js","../node_modules/webrtc-adapter/src/js/adapter_core.js","../src/base/mediastream-factory.js","../src/base/export.js","../src/ulity/debug.js","../src/ulity/event.js","../node_modules/axios/lib/helpers/bind.js","../node_modules/axios/lib/utils.js","../node_modules/axios/lib/helpers/buildURL.js","../node_modules/axios/lib/core/InterceptorManager.js","../node_modules/axios/lib/core/transformData.js","../node_modules/axios/lib/cancel/isCancel.js","../node_modules/axios/lib/helpers/normalizeHeaderName.js","../node_modules/axios/lib/core/enhanceError.js","../node_modules/axios/lib/core/createError.js","../node_modules/axios/lib/core/settle.js","../node_modules/axios/lib/helpers/cookies.js","../node_modules/axios/lib/helpers/isAbsoluteURL.js","../node_modules/axios/lib/helpers/combineURLs.js","../node_modules/axios/lib/core/buildFullPath.js","../node_modules/axios/lib/helpers/parseHeaders.js","../node_modules/axios/lib/helpers/isURLSameOrigin.js","../node_modules/axios/lib/adapters/xhr.js","../node_modules/axios/lib/defaults.js","../node_modules/axios/lib/core/dispatchRequest.js","../node_modules/axios/lib/core/mergeConfig.js","../node_modules/axios/lib/core/Axios.js","../node_modules/axios/lib/cancel/Cancel.js","../node_modules/axios/lib/cancel/CancelToken.js","../node_modules/axios/lib/helpers/spread.js","../node_modules/axios/lib/helpers/isAxiosError.js","../node_modules/axios/lib/axios.js","../node_modules/axios/index.js","../src/endpoint/endpoint.js","../src/base/resolutionfind.js","../src/export.js"],"sourcesContent":["const Events = {\n\tWEBRTC_NOT_SUPPORT : 'WEBRTC_NOT_SUPPORT',\n\tWEBRTC_ICE_CANDIDATE_ERROR : 'WEBRTC_ICE_CANDIDATE_ERROR',\n\tWEBRTC_OFFER_ANSWER_EXCHANGE_FAILED:'WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED',\n\tWEBRTC_ON_REMOTE_STREAMS:'WEBRTC_ON_REMOTE_STREAMS',\n\tWEBRTC_ON_LOCAL_STREAM:'WEBRTC_ON_LOCAL_STREAM',\n\tCAPTURE_STREAM_FAILED:'CAPTURE_STREAM_FAILED'\n};\n\nexport default Events;","export const VERSION = '__VERSION__';\nexport const BUILD_DATE = '__BUILD_DATE__';","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n\n// eslint-disable-next-line require-jsdoc\nexport function isFirefox() {\n return window.navigator.userAgent.match('Firefox') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isChrome() {\n return window.navigator.userAgent.match('Chrome') !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function isSafari() {\n return /^((?!chrome|android).)*safari/i.test(window.navigator.userAgent);\n}\n// eslint-disable-next-line require-jsdoc\nexport function isEdge() {\n return window.navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/) !== null;\n}\n// eslint-disable-next-line require-jsdoc\nexport function createUuid() {\n return 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx'.replace(/[xy]/g, function(c) {\n const r = Math.random() * 16 | 0;\n const v = c === 'x' ? r : (r & 0x3 | 0x8);\n return v.toString(16);\n });\n}","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n/**\n * @class AudioSourceInfo\n * @classDesc Source info about an audio track. Values: 'mic', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const AudioSourceInfo = {\n MIC: 'mic',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class VideoSourceInfo\n * @classDesc Source info about a video track. Values: 'camera', 'screen-cast', 'file', 'mixed'.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const VideoSourceInfo = {\n CAMERA: 'camera',\n SCREENCAST: 'screen-cast',\n FILE: 'file',\n MIXED: 'mixed',\n};\n\n/**\n * @class TrackKind\n * @classDesc Kind of a track. Values: 'audio' for audio track, 'video' for video track, 'av' for both audio and video tracks.\n * @memberOf Owt.Base\n * @readonly\n * @enum {string}\n */\nexport const TrackKind = {\n /**\n * Audio tracks.\n * @type string\n */\n AUDIO: 'audio',\n /**\n * Video tracks.\n * @type string\n */\n VIDEO: 'video',\n /**\n * Both audio and video tracks.\n * @type string\n */\n AUDIO_AND_VIDEO: 'av',\n};\n/**\n * @class Resolution\n * @memberOf Owt.Base\n * @classDesc The Resolution defines the size of a rectangle.\n * @constructor\n * @param {number} width\n * @param {number} height\n */\nexport class Resolution {\n // eslint-disable-next-line require-jsdoc\n constructor(width, height) {\n /**\n * @member {number} width\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.width = width;\n /**\n * @member {number} height\n * @instance\n * @memberof Owt.Base.Resolution\n */\n this.height = height;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nlet logDisabled_ = true;\nlet deprecationWarnings_ = true;\n\n/**\n * Extract browser version out of the provided user agent string.\n *\n * @param {!string} uastring userAgent string.\n * @param {!string} expr Regular expression used as match criteria.\n * @param {!number} pos position in the version string to be returned.\n * @return {!number} browser version.\n */\nexport function extractVersion(uastring, expr, pos) {\n const match = uastring.match(expr);\n return match && match.length >= pos && parseInt(match[pos], 10);\n}\n\n// Wraps the peerconnection event eventNameToWrap in a function\n// which returns the modified event object (or false to prevent\n// the event).\nexport function wrapPeerConnectionEvent(window, eventNameToWrap, wrapper) {\n if (!window.RTCPeerConnection) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n const nativeAddEventListener = proto.addEventListener;\n proto.addEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap) {\n return nativeAddEventListener.apply(this, arguments);\n }\n const wrappedCallback = (e) => {\n const modifiedEvent = wrapper(e);\n if (modifiedEvent) {\n if (cb.handleEvent) {\n cb.handleEvent(modifiedEvent);\n } else {\n cb(modifiedEvent);\n }\n }\n };\n this._eventMap = this._eventMap || {};\n if (!this._eventMap[eventNameToWrap]) {\n this._eventMap[eventNameToWrap] = new Map();\n }\n this._eventMap[eventNameToWrap].set(cb, wrappedCallback);\n return nativeAddEventListener.apply(this, [nativeEventName,\n wrappedCallback]);\n };\n\n const nativeRemoveEventListener = proto.removeEventListener;\n proto.removeEventListener = function(nativeEventName, cb) {\n if (nativeEventName !== eventNameToWrap || !this._eventMap\n || !this._eventMap[eventNameToWrap]) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n if (!this._eventMap[eventNameToWrap].has(cb)) {\n return nativeRemoveEventListener.apply(this, arguments);\n }\n const unwrappedCb = this._eventMap[eventNameToWrap].get(cb);\n this._eventMap[eventNameToWrap].delete(cb);\n if (this._eventMap[eventNameToWrap].size === 0) {\n delete this._eventMap[eventNameToWrap];\n }\n if (Object.keys(this._eventMap).length === 0) {\n delete this._eventMap;\n }\n return nativeRemoveEventListener.apply(this, [nativeEventName,\n unwrappedCb]);\n };\n\n Object.defineProperty(proto, 'on' + eventNameToWrap, {\n get() {\n return this['_on' + eventNameToWrap];\n },\n set(cb) {\n if (this['_on' + eventNameToWrap]) {\n this.removeEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap]);\n delete this['_on' + eventNameToWrap];\n }\n if (cb) {\n this.addEventListener(eventNameToWrap,\n this['_on' + eventNameToWrap] = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n}\n\nexport function disableLog(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n logDisabled_ = bool;\n return (bool) ? 'adapter.js logging disabled' :\n 'adapter.js logging enabled';\n}\n\n/**\n * Disable or enable deprecation warnings\n * @param {!boolean} bool set to true to disable warnings.\n */\nexport function disableWarnings(bool) {\n if (typeof bool !== 'boolean') {\n return new Error('Argument type: ' + typeof bool +\n '. Please use a boolean.');\n }\n deprecationWarnings_ = !bool;\n return 'adapter.js deprecation warnings ' + (bool ? 'disabled' : 'enabled');\n}\n\nexport function log() {\n if (typeof window === 'object') {\n if (logDisabled_) {\n return;\n }\n if (typeof console !== 'undefined' && typeof console.log === 'function') {\n console.log.apply(console, arguments);\n }\n }\n}\n\n/**\n * Shows a deprecation warning suggesting the modern and spec-compatible API.\n */\nexport function deprecated(oldMethod, newMethod) {\n if (!deprecationWarnings_) {\n return;\n }\n console.warn(oldMethod + ' is deprecated, please use ' + newMethod +\n ' instead.');\n}\n\n/**\n * Browser detector.\n *\n * @return {object} result containing browser and version\n * properties.\n */\nexport function detectBrowser(window) {\n // Returned result object.\n const result = {browser: null, version: null};\n\n // Fail early if it's not a browser\n if (typeof window === 'undefined' || !window.navigator) {\n result.browser = 'Not a browser.';\n return result;\n }\n\n const {navigator} = window;\n\n if (navigator.mozGetUserMedia) { // Firefox.\n result.browser = 'firefox';\n result.version = extractVersion(navigator.userAgent,\n /Firefox\\/(\\d+)\\./, 1);\n } else if (navigator.webkitGetUserMedia ||\n (window.isSecureContext === false && window.webkitRTCPeerConnection &&\n !window.RTCIceGatherer)) {\n // Chrome, Chromium, Webview, Opera.\n // Version matches Chrome/WebRTC version.\n // Chrome 74 removed webkitGetUserMedia on http as well so we need the\n // more complicated fallback to webkitRTCPeerConnection.\n result.browser = 'chrome';\n result.version = extractVersion(navigator.userAgent,\n /Chrom(e|ium)\\/(\\d+)\\./, 2);\n } else if (navigator.mediaDevices &&\n navigator.userAgent.match(/Edge\\/(\\d+).(\\d+)$/)) { // Edge.\n result.browser = 'edge';\n result.version = extractVersion(navigator.userAgent,\n /Edge\\/(\\d+).(\\d+)$/, 2);\n } else if (window.RTCPeerConnection &&\n navigator.userAgent.match(/AppleWebKit\\/(\\d+)\\./)) { // Safari.\n result.browser = 'safari';\n result.version = extractVersion(navigator.userAgent,\n /AppleWebKit\\/(\\d+)\\./, 1);\n result.supportsUnifiedPlan = window.RTCRtpTransceiver &&\n 'currentDirection' in window.RTCRtpTransceiver.prototype;\n } else { // Default fallthrough: not supported.\n result.browser = 'Not a supported browser.';\n return result;\n }\n\n return result;\n}\n\n/**\n * Checks if something is an object.\n *\n * @param {*} val The something you want to check.\n * @return true if val is an object, false otherwise.\n */\nfunction isObject(val) {\n return Object.prototype.toString.call(val) === '[object Object]';\n}\n\n/**\n * Remove all empty objects and undefined values\n * from a nested object -- an enhanced and vanilla version\n * of Lodash's `compact`.\n */\nexport function compactObject(data) {\n if (!isObject(data)) {\n return data;\n }\n\n return Object.keys(data).reduce(function(accumulator, key) {\n const isObj = isObject(data[key]);\n const value = isObj ? compactObject(data[key]) : data[key];\n const isEmptyObject = isObj && !Object.keys(value).length;\n if (value === undefined || isEmptyObject) {\n return accumulator;\n }\n return Object.assign(accumulator, {[key]: value});\n }, {});\n}\n\n/* iterates the stats graph recursively. */\nexport function walkStats(stats, base, resultSet) {\n if (!base || resultSet.has(base.id)) {\n return;\n }\n resultSet.set(base.id, base);\n Object.keys(base).forEach(name => {\n if (name.endsWith('Id')) {\n walkStats(stats, stats.get(base[name]), resultSet);\n } else if (name.endsWith('Ids')) {\n base[name].forEach(id => {\n walkStats(stats, stats.get(id), resultSet);\n });\n }\n });\n}\n\n/* filter getStats for a sender/receiver track. */\nexport function filterStats(result, track, outbound) {\n const streamStatsType = outbound ? 'outbound-rtp' : 'inbound-rtp';\n const filteredResult = new Map();\n if (track === null) {\n return filteredResult;\n }\n const trackStats = [];\n result.forEach(value => {\n if (value.type === 'track' &&\n value.trackIdentifier === track.id) {\n trackStats.push(value);\n }\n });\n trackStats.forEach(trackStat => {\n result.forEach(stats => {\n if (stats.type === streamStatsType && stats.trackId === trackStat.id) {\n walkStats(result, stats, filteredResult);\n }\n });\n });\n return filteredResult;\n}\n\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\nconst logging = utils.log;\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n\n if (!navigator.mediaDevices) {\n return;\n }\n\n const constraintsToChrome_ = function(c) {\n if (typeof c !== 'object' || c.mandatory || c.optional) {\n return c;\n }\n const cc = {};\n Object.keys(c).forEach(key => {\n if (key === 'require' || key === 'advanced' || key === 'mediaSource') {\n return;\n }\n const r = (typeof c[key] === 'object') ? c[key] : {ideal: c[key]};\n if (r.exact !== undefined && typeof r.exact === 'number') {\n r.min = r.max = r.exact;\n }\n const oldname_ = function(prefix, name) {\n if (prefix) {\n return prefix + name.charAt(0).toUpperCase() + name.slice(1);\n }\n return (name === 'deviceId') ? 'sourceId' : name;\n };\n if (r.ideal !== undefined) {\n cc.optional = cc.optional || [];\n let oc = {};\n if (typeof r.ideal === 'number') {\n oc[oldname_('min', key)] = r.ideal;\n cc.optional.push(oc);\n oc = {};\n oc[oldname_('max', key)] = r.ideal;\n cc.optional.push(oc);\n } else {\n oc[oldname_('', key)] = r.ideal;\n cc.optional.push(oc);\n }\n }\n if (r.exact !== undefined && typeof r.exact !== 'number') {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_('', key)] = r.exact;\n } else {\n ['min', 'max'].forEach(mix => {\n if (r[mix] !== undefined) {\n cc.mandatory = cc.mandatory || {};\n cc.mandatory[oldname_(mix, key)] = r[mix];\n }\n });\n }\n });\n if (c.advanced) {\n cc.optional = (cc.optional || []).concat(c.advanced);\n }\n return cc;\n };\n\n const shimConstraints_ = function(constraints, func) {\n if (browserDetails.version >= 61) {\n return func(constraints);\n }\n constraints = JSON.parse(JSON.stringify(constraints));\n if (constraints && typeof constraints.audio === 'object') {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n constraints = JSON.parse(JSON.stringify(constraints));\n remap(constraints.audio, 'autoGainControl', 'googAutoGainControl');\n remap(constraints.audio, 'noiseSuppression', 'googNoiseSuppression');\n constraints.audio = constraintsToChrome_(constraints.audio);\n }\n if (constraints && typeof constraints.video === 'object') {\n // Shim facingMode for mobile & surface pro.\n let face = constraints.video.facingMode;\n face = face && ((typeof face === 'object') ? face : {ideal: face});\n const getSupportedFacingModeLies = browserDetails.version < 66;\n\n if ((face && (face.exact === 'user' || face.exact === 'environment' ||\n face.ideal === 'user' || face.ideal === 'environment')) &&\n !(navigator.mediaDevices.getSupportedConstraints &&\n navigator.mediaDevices.getSupportedConstraints().facingMode &&\n !getSupportedFacingModeLies)) {\n delete constraints.video.facingMode;\n let matches;\n if (face.exact === 'environment' || face.ideal === 'environment') {\n matches = ['back', 'rear'];\n } else if (face.exact === 'user' || face.ideal === 'user') {\n matches = ['front'];\n }\n if (matches) {\n // Look for matches in label, or use last cam for back (typical).\n return navigator.mediaDevices.enumerateDevices()\n .then(devices => {\n devices = devices.filter(d => d.kind === 'videoinput');\n let dev = devices.find(d => matches.some(match =>\n d.label.toLowerCase().includes(match)));\n if (!dev && devices.length && matches.includes('back')) {\n dev = devices[devices.length - 1]; // more likely the back cam\n }\n if (dev) {\n constraints.video.deviceId = face.exact ? {exact: dev.deviceId} :\n {ideal: dev.deviceId};\n }\n constraints.video = constraintsToChrome_(constraints.video);\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n });\n }\n }\n constraints.video = constraintsToChrome_(constraints.video);\n }\n logging('chrome: ' + JSON.stringify(constraints));\n return func(constraints);\n };\n\n const shimError_ = function(e) {\n if (browserDetails.version >= 64) {\n return e;\n }\n return {\n name: {\n PermissionDeniedError: 'NotAllowedError',\n PermissionDismissedError: 'NotAllowedError',\n InvalidStateError: 'NotAllowedError',\n DevicesNotFoundError: 'NotFoundError',\n ConstraintNotSatisfiedError: 'OverconstrainedError',\n TrackStartError: 'NotReadableError',\n MediaDeviceFailedDueToShutdown: 'NotAllowedError',\n MediaDeviceKillSwitchOn: 'NotAllowedError',\n TabCaptureError: 'AbortError',\n ScreenCaptureError: 'AbortError',\n DeviceCaptureError: 'AbortError'\n }[e.name] || e.name,\n message: e.message,\n constraint: e.constraint || e.constraintName,\n toString() {\n return this.name + (this.message && ': ') + this.message;\n }\n };\n };\n\n const getUserMedia_ = function(constraints, onSuccess, onError) {\n shimConstraints_(constraints, c => {\n navigator.webkitGetUserMedia(c, onSuccess, e => {\n if (onError) {\n onError(shimError_(e));\n }\n });\n });\n };\n navigator.getUserMedia = getUserMedia_.bind(navigator);\n\n // Even though Chrome 45 has navigator.mediaDevices and a getUserMedia\n // function which returns a Promise, it does not accept spec-style\n // constraints.\n if (navigator.mediaDevices.getUserMedia) {\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(cs) {\n return shimConstraints_(cs, c => origGetUserMedia(c).then(stream => {\n if (c.audio && !stream.getAudioTracks().length ||\n c.video && !stream.getVideoTracks().length) {\n stream.getTracks().forEach(track => {\n track.stop();\n });\n throw new DOMException('', 'NotFoundError');\n }\n return stream;\n }, e => Promise.reject(shimError_(e))));\n };\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\nexport function shimGetDisplayMedia(window, getSourceId) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n // getSourceId is a function that returns a promise resolving with\n // the sourceId of the screen/window/tab to be shared.\n if (typeof getSourceId !== 'function') {\n console.error('shimGetDisplayMedia: getSourceId argument is not ' +\n 'a function');\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n return getSourceId(constraints)\n .then(sourceId => {\n const widthSpecified = constraints.video && constraints.video.width;\n const heightSpecified = constraints.video &&\n constraints.video.height;\n const frameRateSpecified = constraints.video &&\n constraints.video.frameRate;\n constraints.video = {\n mandatory: {\n chromeMediaSource: 'desktop',\n chromeMediaSourceId: sourceId,\n maxFrameRate: frameRateSpecified || 3\n }\n };\n if (widthSpecified) {\n constraints.video.mandatory.maxWidth = widthSpecified;\n }\n if (heightSpecified) {\n constraints.video.mandatory.maxHeight = heightSpecified;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n });\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\nimport * as utils from '../utils.js';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimMediaStream(window) {\n window.MediaStream = window.MediaStream || window.webkitMediaStream;\n}\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCPeerConnection && !('ontrack' in\n window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'ontrack', {\n get() {\n return this._ontrack;\n },\n set(f) {\n if (this._ontrack) {\n this.removeEventListener('track', this._ontrack);\n }\n this.addEventListener('track', this._ontrack = f);\n },\n enumerable: true,\n configurable: true\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n if (!this._ontrackpoly) {\n this._ontrackpoly = (e) => {\n // onaddstream does not fire when a track is added to an existing\n // stream. But stream.onaddtrack is implemented so we use that.\n e.stream.addEventListener('addtrack', te => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === te.track.id);\n } else {\n receiver = {track: te.track};\n }\n\n const event = new Event('track');\n event.track = te.track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n e.stream.getTracks().forEach(track => {\n let receiver;\n if (window.RTCPeerConnection.prototype.getReceivers) {\n receiver = this.getReceivers()\n .find(r => r.track && r.track.id === track.id);\n } else {\n receiver = {track};\n }\n const event = new Event('track');\n event.track = track;\n event.receiver = receiver;\n event.transceiver = {receiver};\n event.streams = [e.stream];\n this.dispatchEvent(event);\n });\n };\n this.addEventListener('addstream', this._ontrackpoly);\n }\n return origSetRemoteDescription.apply(this, arguments);\n };\n } else {\n // even if RTCRtpTransceiver is in window, it is only used and\n // emitted in unified-plan. Unfortunately this means we need\n // to unconditionally wrap the event.\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n if (!e.transceiver) {\n Object.defineProperty(e, 'transceiver',\n {value: {receiver: e.receiver}});\n }\n return e;\n });\n }\n}\n\nexport function shimGetSendersWithDtmf(window) {\n // Overrides addTrack/removeTrack, depends on shimAddTrackRemoveTrack.\n if (typeof window === 'object' && window.RTCPeerConnection &&\n !('getSenders' in window.RTCPeerConnection.prototype) &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype) {\n const shimSenderWithDtmf = function(pc, track) {\n return {\n track,\n get dtmf() {\n if (this._dtmf === undefined) {\n if (track.kind === 'audio') {\n this._dtmf = pc.createDTMFSender(track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n },\n _pc: pc\n };\n };\n\n // augment addTrack when getSenders is not available.\n if (!window.RTCPeerConnection.prototype.getSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n this._senders = this._senders || [];\n return this._senders.slice(); // return a copy of the internal state.\n };\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n let sender = origAddTrack.apply(this, arguments);\n if (!sender) {\n sender = shimSenderWithDtmf(this, track);\n this._senders.push(sender);\n }\n return sender;\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n origRemoveTrack.apply(this, arguments);\n const idx = this._senders.indexOf(sender);\n if (idx !== -1) {\n this._senders.splice(idx, 1);\n }\n };\n }\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._senders = this._senders || [];\n origAddStream.apply(this, [stream]);\n stream.getTracks().forEach(track => {\n this._senders.push(shimSenderWithDtmf(this, track));\n });\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._senders = this._senders || [];\n origRemoveStream.apply(this, [stream]);\n\n stream.getTracks().forEach(track => {\n const sender = this._senders.find(s => s.track === track);\n if (sender) { // remove sender\n this._senders.splice(this._senders.indexOf(sender), 1);\n }\n });\n };\n } else if (typeof window === 'object' && window.RTCPeerConnection &&\n 'getSenders' in window.RTCPeerConnection.prototype &&\n 'createDTMFSender' in window.RTCPeerConnection.prototype &&\n window.RTCRtpSender &&\n !('dtmf' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = this._pc.createDTMFSender(this.track);\n } else {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n}\n\nexport function shimGetStats(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n\n // If selector is a function then we are in the old style stats so just\n // pass back the original getStats format to avoid breaking old users.\n if (arguments.length > 0 && typeof selector === 'function') {\n return origGetStats.apply(this, arguments);\n }\n\n // When spec-style getStats is supported, return those when called with\n // either no arguments or the selector argument is null.\n if (origGetStats.length === 0 && (arguments.length === 0 ||\n typeof selector !== 'function')) {\n return origGetStats.apply(this, []);\n }\n\n const fixChromeStats_ = function(response) {\n const standardReport = {};\n const reports = response.result();\n reports.forEach(report => {\n const standardStats = {\n id: report.id,\n timestamp: report.timestamp,\n type: {\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[report.type] || report.type\n };\n report.names().forEach(name => {\n standardStats[name] = report.stat(name);\n });\n standardReport[standardStats.id] = standardStats;\n });\n\n return standardReport;\n };\n\n // shim getStats with maplike support\n const makeMapStats = function(stats) {\n return new Map(Object.keys(stats).map(key => [key, stats[key]]));\n };\n\n if (arguments.length >= 2) {\n const successCallbackWrapper_ = function(response) {\n onSucc(makeMapStats(fixChromeStats_(response)));\n };\n\n return origGetStats.apply(this, [successCallbackWrapper_,\n selector]);\n }\n\n // promise-support\n return new Promise((resolve, reject) => {\n origGetStats.apply(this, [\n function(response) {\n resolve(makeMapStats(fixChromeStats_(response)));\n }, reject]);\n }).then(onSucc, onErr);\n };\n}\n\nexport function shimSenderReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender && window.RTCRtpReceiver)) {\n return;\n }\n\n // shim sender stats.\n if (!('getStats' in window.RTCRtpSender.prototype)) {\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n const sender = this;\n return this._pc.getStats().then(result =>\n /* Note: this will include stats of all senders that\n * send a track with the same id as sender.track as\n * it is not possible to identify the RTCRtpSender.\n */\n utils.filterStats(result, sender.track, true));\n };\n }\n\n // shim receiver stats.\n if (!('getStats' in window.RTCRtpReceiver.prototype)) {\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers =\n function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n const receiver = this;\n return this._pc.getStats().then(result =>\n utils.filterStats(result, receiver.track, false));\n };\n }\n\n if (!('getStats' in window.RTCRtpSender.prototype &&\n 'getStats' in window.RTCRtpReceiver.prototype)) {\n return;\n }\n\n // shim RTCPeerConnection.getStats(track).\n const origGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n if (arguments.length > 0 &&\n arguments[0] instanceof window.MediaStreamTrack) {\n const track = arguments[0];\n let sender;\n let receiver;\n let err;\n this.getSenders().forEach(s => {\n if (s.track === track) {\n if (sender) {\n err = true;\n } else {\n sender = s;\n }\n }\n });\n this.getReceivers().forEach(r => {\n if (r.track === track) {\n if (receiver) {\n err = true;\n } else {\n receiver = r;\n }\n }\n return r.track === track;\n });\n if (err || (sender && receiver)) {\n return Promise.reject(new DOMException(\n 'There are more than one sender or receiver for the track.',\n 'InvalidAccessError'));\n } else if (sender) {\n return sender.getStats();\n } else if (receiver) {\n return receiver.getStats();\n }\n return Promise.reject(new DOMException(\n 'There is no sender or receiver for the track.',\n 'InvalidAccessError'));\n }\n return origGetStats.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrackWithNative(window) {\n // shim addTrack/removeTrack with native variants in order to make\n // the interactions with legacy getLocalStreams behave as in other browsers.\n // Keeps a mapping stream.id => [stream, rtpsenders...]\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n return Object.keys(this._shimmedLocalStreams)\n .map(streamId => this._shimmedLocalStreams[streamId][0]);\n };\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (!stream) {\n return origAddTrack.apply(this, arguments);\n }\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n const sender = origAddTrack.apply(this, arguments);\n if (!this._shimmedLocalStreams[stream.id]) {\n this._shimmedLocalStreams[stream.id] = [stream, sender];\n } else if (this._shimmedLocalStreams[stream.id].indexOf(sender) === -1) {\n this._shimmedLocalStreams[stream.id].push(sender);\n }\n return sender;\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n const existingSenders = this.getSenders();\n origAddStream.apply(this, arguments);\n const newSenders = this.getSenders()\n .filter(newSender => existingSenders.indexOf(newSender) === -1);\n this._shimmedLocalStreams[stream.id] = [stream].concat(newSenders);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n delete this._shimmedLocalStreams[stream.id];\n return origRemoveStream.apply(this, arguments);\n };\n\n const origRemoveTrack = window.RTCPeerConnection.prototype.removeTrack;\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n this._shimmedLocalStreams = this._shimmedLocalStreams || {};\n if (sender) {\n Object.keys(this._shimmedLocalStreams).forEach(streamId => {\n const idx = this._shimmedLocalStreams[streamId].indexOf(sender);\n if (idx !== -1) {\n this._shimmedLocalStreams[streamId].splice(idx, 1);\n }\n if (this._shimmedLocalStreams[streamId].length === 1) {\n delete this._shimmedLocalStreams[streamId];\n }\n });\n }\n return origRemoveTrack.apply(this, arguments);\n };\n}\n\nexport function shimAddTrackRemoveTrack(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // shim addTrack and removeTrack.\n if (window.RTCPeerConnection.prototype.addTrack &&\n browserDetails.version >= 65) {\n return shimAddTrackRemoveTrackWithNative(window);\n }\n\n // also shim pc.getLocalStreams when addTrack is shimmed\n // to return the original streams.\n const origGetLocalStreams = window.RTCPeerConnection.prototype\n .getLocalStreams;\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n const nativeStreams = origGetLocalStreams.apply(this);\n this._reverseStreams = this._reverseStreams || {};\n return nativeStreams.map(stream => this._reverseStreams[stream.id]);\n };\n\n const origAddStream = window.RTCPeerConnection.prototype.addStream;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n stream.getTracks().forEach(track => {\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n });\n // Add identity mapping for consistency with addTrack.\n // Unless this is being used with a stream from addTrack.\n if (!this._reverseStreams[stream.id]) {\n const newStream = new window.MediaStream(stream.getTracks());\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n stream = newStream;\n }\n origAddStream.apply(this, [stream]);\n };\n\n const origRemoveStream = window.RTCPeerConnection.prototype.removeStream;\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n\n origRemoveStream.apply(this, [(this._streams[stream.id] || stream)]);\n delete this._reverseStreams[(this._streams[stream.id] ?\n this._streams[stream.id].id : stream.id)];\n delete this._streams[stream.id];\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, stream) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n const streams = [].slice.call(arguments, 1);\n if (streams.length !== 1 ||\n !streams[0].getTracks().find(t => t === track)) {\n // this is not fully correct but all we can manage without\n // [[associated MediaStreams]] internal slot.\n throw new DOMException(\n 'The adapter.js addTrack polyfill only supports a single ' +\n ' stream which is associated with the specified track.',\n 'NotSupportedError');\n }\n\n const alreadyExists = this.getSenders().find(s => s.track === track);\n if (alreadyExists) {\n throw new DOMException('Track already exists.',\n 'InvalidAccessError');\n }\n\n this._streams = this._streams || {};\n this._reverseStreams = this._reverseStreams || {};\n const oldStream = this._streams[stream.id];\n if (oldStream) {\n // this is using odd Chrome behaviour, use with caution:\n // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815\n // Note: we rely on the high-level addTrack/dtmf shim to\n // create the sender with a dtmf sender.\n oldStream.addTrack(track);\n\n // Trigger ONN async.\n Promise.resolve().then(() => {\n this.dispatchEvent(new Event('negotiationneeded'));\n });\n } else {\n const newStream = new window.MediaStream([track]);\n this._streams[stream.id] = newStream;\n this._reverseStreams[newStream.id] = stream;\n this.addStream(newStream);\n }\n return this.getSenders().find(s => s.track === track);\n };\n\n // replace the internal stream id with the external one and\n // vice versa.\n function replaceInternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(internalStream.id, 'g'),\n externalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n function replaceExternalStreamId(pc, description) {\n let sdp = description.sdp;\n Object.keys(pc._reverseStreams || []).forEach(internalId => {\n const externalStream = pc._reverseStreams[internalId];\n const internalStream = pc._streams[externalStream.id];\n sdp = sdp.replace(new RegExp(externalStream.id, 'g'),\n internalStream.id);\n });\n return new RTCSessionDescription({\n type: description.type,\n sdp\n });\n }\n ['createOffer', 'createAnswer'].forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n const args = arguments;\n const isLegacyCall = arguments.length &&\n typeof arguments[0] === 'function';\n if (isLegacyCall) {\n return nativeMethod.apply(this, [\n (description) => {\n const desc = replaceInternalStreamId(this, description);\n args[0].apply(null, [desc]);\n },\n (err) => {\n if (args[1]) {\n args[1].apply(null, err);\n }\n }, arguments[2]\n ]);\n }\n return nativeMethod.apply(this, arguments)\n .then(description => replaceInternalStreamId(this, description));\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n\n const origSetLocalDescription =\n window.RTCPeerConnection.prototype.setLocalDescription;\n window.RTCPeerConnection.prototype.setLocalDescription =\n function setLocalDescription() {\n if (!arguments.length || !arguments[0].type) {\n return origSetLocalDescription.apply(this, arguments);\n }\n arguments[0] = replaceExternalStreamId(this, arguments[0]);\n return origSetLocalDescription.apply(this, arguments);\n };\n\n // TODO: mangle getStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats-streamidentifier\n\n const origLocalDescription = Object.getOwnPropertyDescriptor(\n window.RTCPeerConnection.prototype, 'localDescription');\n Object.defineProperty(window.RTCPeerConnection.prototype,\n 'localDescription', {\n get() {\n const description = origLocalDescription.get.apply(this);\n if (description.type === '') {\n return description;\n }\n return replaceInternalStreamId(this, description);\n }\n });\n\n window.RTCPeerConnection.prototype.removeTrack =\n function removeTrack(sender) {\n if (this.signalingState === 'closed') {\n throw new DOMException(\n 'The RTCPeerConnection\\'s signalingState is \\'closed\\'.',\n 'InvalidStateError');\n }\n // We can not yet check for sender instanceof RTCRtpSender\n // since we shim RTPSender. So we check if sender._pc is set.\n if (!sender._pc) {\n throw new DOMException('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.', 'TypeError');\n }\n const isLocal = sender._pc === this;\n if (!isLocal) {\n throw new DOMException('Sender was not created by this connection.',\n 'InvalidAccessError');\n }\n\n // Search for the native stream the senders track belongs to.\n this._streams = this._streams || {};\n let stream;\n Object.keys(this._streams).forEach(streamid => {\n const hasTrack = this._streams[streamid].getTracks()\n .find(track => sender.track === track);\n if (hasTrack) {\n stream = this._streams[streamid];\n }\n });\n\n if (stream) {\n if (stream.getTracks().length === 1) {\n // if this is the last track of the stream, remove the stream. This\n // takes care of any shimmed _senders.\n this.removeStream(this._reverseStreams[stream.id]);\n } else {\n // relying on the same odd chrome behaviour as above.\n stream.removeTrack(sender.track);\n }\n this.dispatchEvent(new Event('negotiationneeded'));\n }\n };\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (!window.RTCPeerConnection && window.webkitRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.webkitRTCPeerConnection;\n }\n if (!window.RTCPeerConnection) {\n return;\n }\n\n // shim implicit creation of RTCSessionDescription/RTCIceCandidate\n if (browserDetails.version < 53) {\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n}\n\n// Attempt to fix ONN in plan-b mode.\nexport function fixNegotiationNeeded(window, browserDetails) {\n utils.wrapPeerConnectionEvent(window, 'negotiationneeded', e => {\n const pc = e.target;\n if (browserDetails.version < 72 || (pc.getConfiguration &&\n pc.getConfiguration().sdpSemantics === 'plan-b')) {\n if (pc.signalingState !== 'stable') {\n return;\n }\n }\n return e;\n });\n}\n","/*\n * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nexport function filterIceServers(iceServers, edgeVersion) {\n let hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(server => {\n if (server && (server.urls || server.url)) {\n let urls = server.urls || server.url;\n if (server.url && !server.urls) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n }\n const isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(url => {\n // filter STUN unconditionally.\n if (url.indexOf('stun:') === 0) {\n return false;\n }\n\n const validTurn = url.startsWith('turn') &&\n !url.startsWith('turn:[') &&\n url.includes('transport=udp');\n if (validTurn && !hasTurn) {\n hasTurn = true;\n return true;\n }\n return validTurn && !hasTurn;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n","/* eslint-env node */\n'use strict';\n\n// SDP helpers.\nvar SDPUtils = {};\n\n// Generate an alphanumeric identifier for cname or mids.\n// TODO: use UUIDs instead? https://gist.github.com/jed/982883\nSDPUtils.generateIdentifier = function() {\n return Math.random().toString(36).substr(2, 10);\n};\n\n// The RTCP CNAME used by all peerconnections from the same JS.\nSDPUtils.localCName = SDPUtils.generateIdentifier();\n\n// Splits SDP into lines, dealing with both CRLF and LF.\nSDPUtils.splitLines = function(blob) {\n return blob.trim().split('\\n').map(function(line) {\n return line.trim();\n });\n};\n// Splits SDP into sessionpart and mediasections. Ensures CRLF.\nSDPUtils.splitSections = function(blob) {\n var parts = blob.split('\\nm=');\n return parts.map(function(part, index) {\n return (index > 0 ? 'm=' + part : part).trim() + '\\r\\n';\n });\n};\n\n// returns the session description.\nSDPUtils.getDescription = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n return sections && sections[0];\n};\n\n// returns the individual media sections.\nSDPUtils.getMediaSections = function(blob) {\n var sections = SDPUtils.splitSections(blob);\n sections.shift();\n return sections;\n};\n\n// Returns lines that start with a certain prefix.\nSDPUtils.matchPrefix = function(blob, prefix) {\n return SDPUtils.splitLines(blob).filter(function(line) {\n return line.indexOf(prefix) === 0;\n });\n};\n\n// Parses an ICE candidate line. Sample input:\n// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8\n// rport 55996\"\nSDPUtils.parseCandidate = function(line) {\n var parts;\n // Parse both variants.\n if (line.indexOf('a=candidate:') === 0) {\n parts = line.substring(12).split(' ');\n } else {\n parts = line.substring(10).split(' ');\n }\n\n var candidate = {\n foundation: parts[0],\n component: parseInt(parts[1], 10),\n protocol: parts[2].toLowerCase(),\n priority: parseInt(parts[3], 10),\n ip: parts[4],\n address: parts[4], // address is an alias for ip.\n port: parseInt(parts[5], 10),\n // skip parts[6] == 'typ'\n type: parts[7]\n };\n\n for (var i = 8; i < parts.length; i += 2) {\n switch (parts[i]) {\n case 'raddr':\n candidate.relatedAddress = parts[i + 1];\n break;\n case 'rport':\n candidate.relatedPort = parseInt(parts[i + 1], 10);\n break;\n case 'tcptype':\n candidate.tcpType = parts[i + 1];\n break;\n case 'ufrag':\n candidate.ufrag = parts[i + 1]; // for backward compability.\n candidate.usernameFragment = parts[i + 1];\n break;\n default: // extension handling, in particular ufrag\n candidate[parts[i]] = parts[i + 1];\n break;\n }\n }\n return candidate;\n};\n\n// Translates a candidate object into SDP candidate attribute.\nSDPUtils.writeCandidate = function(candidate) {\n var sdp = [];\n sdp.push(candidate.foundation);\n sdp.push(candidate.component);\n sdp.push(candidate.protocol.toUpperCase());\n sdp.push(candidate.priority);\n sdp.push(candidate.address || candidate.ip);\n sdp.push(candidate.port);\n\n var type = candidate.type;\n sdp.push('typ');\n sdp.push(type);\n if (type !== 'host' && candidate.relatedAddress &&\n candidate.relatedPort) {\n sdp.push('raddr');\n sdp.push(candidate.relatedAddress);\n sdp.push('rport');\n sdp.push(candidate.relatedPort);\n }\n if (candidate.tcpType && candidate.protocol.toLowerCase() === 'tcp') {\n sdp.push('tcptype');\n sdp.push(candidate.tcpType);\n }\n if (candidate.usernameFragment || candidate.ufrag) {\n sdp.push('ufrag');\n sdp.push(candidate.usernameFragment || candidate.ufrag);\n }\n return 'candidate:' + sdp.join(' ');\n};\n\n// Parses an ice-options line, returns an array of option tags.\n// a=ice-options:foo bar\nSDPUtils.parseIceOptions = function(line) {\n return line.substr(14).split(' ');\n};\n\n// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:\n// a=rtpmap:111 opus/48000/2\nSDPUtils.parseRtpMap = function(line) {\n var parts = line.substr(9).split(' ');\n var parsed = {\n payloadType: parseInt(parts.shift(), 10) // was: id\n };\n\n parts = parts[0].split('/');\n\n parsed.name = parts[0];\n parsed.clockRate = parseInt(parts[1], 10); // was: clockrate\n parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;\n // legacy alias, got renamed back to channels in ORTC.\n parsed.numChannels = parsed.channels;\n return parsed;\n};\n\n// Generate an a=rtpmap line from RTCRtpCodecCapability or\n// RTCRtpCodecParameters.\nSDPUtils.writeRtpMap = function(codec) {\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n var channels = codec.channels || codec.numChannels || 1;\n return 'a=rtpmap:' + pt + ' ' + codec.name + '/' + codec.clockRate +\n (channels !== 1 ? '/' + channels : '') + '\\r\\n';\n};\n\n// Parses an a=extmap line (headerextension from RFC 5285). Sample input:\n// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset\nSDPUtils.parseExtmap = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n id: parseInt(parts[0], 10),\n direction: parts[0].indexOf('/') > 0 ? parts[0].split('/')[1] : 'sendrecv',\n uri: parts[1]\n };\n};\n\n// Generates a=extmap line from RTCRtpHeaderExtensionParameters or\n// RTCRtpHeaderExtension.\nSDPUtils.writeExtmap = function(headerExtension) {\n return 'a=extmap:' + (headerExtension.id || headerExtension.preferredId) +\n (headerExtension.direction && headerExtension.direction !== 'sendrecv'\n ? '/' + headerExtension.direction\n : '') +\n ' ' + headerExtension.uri + '\\r\\n';\n};\n\n// Parses an ftmp line, returns dictionary. Sample input:\n// a=fmtp:96 vbr=on;cng=on\n// Also deals with vbr=on; cng=on\nSDPUtils.parseFmtp = function(line) {\n var parsed = {};\n var kv;\n var parts = line.substr(line.indexOf(' ') + 1).split(';');\n for (var j = 0; j < parts.length; j++) {\n kv = parts[j].trim().split('=');\n parsed[kv[0].trim()] = kv[1];\n }\n return parsed;\n};\n\n// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeFmtp = function(codec) {\n var line = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.parameters && Object.keys(codec.parameters).length) {\n var params = [];\n Object.keys(codec.parameters).forEach(function(param) {\n if (codec.parameters[param]) {\n params.push(param + '=' + codec.parameters[param]);\n } else {\n params.push(param);\n }\n });\n line += 'a=fmtp:' + pt + ' ' + params.join(';') + '\\r\\n';\n }\n return line;\n};\n\n// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:\n// a=rtcp-fb:98 nack rpsi\nSDPUtils.parseRtcpFb = function(line) {\n var parts = line.substr(line.indexOf(' ') + 1).split(' ');\n return {\n type: parts.shift(),\n parameter: parts.join(' ')\n };\n};\n// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.\nSDPUtils.writeRtcpFb = function(codec) {\n var lines = '';\n var pt = codec.payloadType;\n if (codec.preferredPayloadType !== undefined) {\n pt = codec.preferredPayloadType;\n }\n if (codec.rtcpFeedback && codec.rtcpFeedback.length) {\n // FIXME: special handling for trr-int?\n codec.rtcpFeedback.forEach(function(fb) {\n lines += 'a=rtcp-fb:' + pt + ' ' + fb.type +\n (fb.parameter && fb.parameter.length ? ' ' + fb.parameter : '') +\n '\\r\\n';\n });\n }\n return lines;\n};\n\n// Parses an RFC 5576 ssrc media attribute. Sample input:\n// a=ssrc:3735928559 cname:something\nSDPUtils.parseSsrcMedia = function(line) {\n var sp = line.indexOf(' ');\n var parts = {\n ssrc: parseInt(line.substr(7, sp - 7), 10)\n };\n var colon = line.indexOf(':', sp);\n if (colon > -1) {\n parts.attribute = line.substr(sp + 1, colon - sp - 1);\n parts.value = line.substr(colon + 1);\n } else {\n parts.attribute = line.substr(sp + 1);\n }\n return parts;\n};\n\nSDPUtils.parseSsrcGroup = function(line) {\n var parts = line.substr(13).split(' ');\n return {\n semantics: parts.shift(),\n ssrcs: parts.map(function(ssrc) {\n return parseInt(ssrc, 10);\n })\n };\n};\n\n// Extracts the MID (RFC 5888) from a media section.\n// returns the MID or undefined if no mid line was found.\nSDPUtils.getMid = function(mediaSection) {\n var mid = SDPUtils.matchPrefix(mediaSection, 'a=mid:')[0];\n if (mid) {\n return mid.substr(6);\n }\n};\n\nSDPUtils.parseFingerprint = function(line) {\n var parts = line.substr(14).split(' ');\n return {\n algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.\n value: parts[1]\n };\n};\n\n// Extracts DTLS parameters from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the fingerprint line as input. See also getIceParameters.\nSDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=fingerprint:');\n // Note: a=setup line is ignored since we use the 'auto' role.\n // Note2: 'algorithm' is not case sensitive except in Edge.\n return {\n role: 'auto',\n fingerprints: lines.map(SDPUtils.parseFingerprint)\n };\n};\n\n// Serializes DTLS parameters to SDP.\nSDPUtils.writeDtlsParameters = function(params, setupType) {\n var sdp = 'a=setup:' + setupType + '\\r\\n';\n params.fingerprints.forEach(function(fp) {\n sdp += 'a=fingerprint:' + fp.algorithm + ' ' + fp.value + '\\r\\n';\n });\n return sdp;\n};\n\n// Parses a=crypto lines into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members\nSDPUtils.parseCryptoLine = function(line) {\n var parts = line.substr(9).split(' ');\n return {\n tag: parseInt(parts[0], 10),\n cryptoSuite: parts[1],\n keyParams: parts[2],\n sessionParams: parts.slice(3),\n };\n};\n\nSDPUtils.writeCryptoLine = function(parameters) {\n return 'a=crypto:' + parameters.tag + ' ' +\n parameters.cryptoSuite + ' ' +\n (typeof parameters.keyParams === 'object'\n ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)\n : parameters.keyParams) +\n (parameters.sessionParams ? ' ' + parameters.sessionParams.join(' ') : '') +\n '\\r\\n';\n};\n\n// Parses the crypto key parameters into\n// https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*\nSDPUtils.parseCryptoKeyParams = function(keyParams) {\n if (keyParams.indexOf('inline:') !== 0) {\n return null;\n }\n var parts = keyParams.substr(7).split('|');\n return {\n keyMethod: 'inline',\n keySalt: parts[0],\n lifeTime: parts[1],\n mkiValue: parts[2] ? parts[2].split(':')[0] : undefined,\n mkiLength: parts[2] ? parts[2].split(':')[1] : undefined,\n };\n};\n\nSDPUtils.writeCryptoKeyParams = function(keyParams) {\n return keyParams.keyMethod + ':'\n + keyParams.keySalt +\n (keyParams.lifeTime ? '|' + keyParams.lifeTime : '') +\n (keyParams.mkiValue && keyParams.mkiLength\n ? '|' + keyParams.mkiValue + ':' + keyParams.mkiLength\n : '');\n};\n\n// Extracts all SDES paramters.\nSDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {\n var lines = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=crypto:');\n return lines.map(SDPUtils.parseCryptoLine);\n};\n\n// Parses ICE information from SDP media section or sessionpart.\n// FIXME: for consistency with other functions this should only\n// get the ice-ufrag and ice-pwd lines as input.\nSDPUtils.getIceParameters = function(mediaSection, sessionpart) {\n var ufrag = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-ufrag:')[0];\n var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart,\n 'a=ice-pwd:')[0];\n if (!(ufrag && pwd)) {\n return null;\n }\n return {\n usernameFragment: ufrag.substr(12),\n password: pwd.substr(10),\n };\n};\n\n// Serializes ICE parameters to SDP.\nSDPUtils.writeIceParameters = function(params) {\n return 'a=ice-ufrag:' + params.usernameFragment + '\\r\\n' +\n 'a=ice-pwd:' + params.password + '\\r\\n';\n};\n\n// Parses the SDP media section and returns RTCRtpParameters.\nSDPUtils.parseRtpParameters = function(mediaSection) {\n var description = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: [],\n rtcp: []\n };\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n for (var i = 3; i < mline.length; i++) { // find all codecs from mline[3..]\n var pt = mline[i];\n var rtpmapline = SDPUtils.matchPrefix(\n mediaSection, 'a=rtpmap:' + pt + ' ')[0];\n if (rtpmapline) {\n var codec = SDPUtils.parseRtpMap(rtpmapline);\n var fmtps = SDPUtils.matchPrefix(\n mediaSection, 'a=fmtp:' + pt + ' ');\n // Only the first a=fmtp: is considered.\n codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};\n codec.rtcpFeedback = SDPUtils.matchPrefix(\n mediaSection, 'a=rtcp-fb:' + pt + ' ')\n .map(SDPUtils.parseRtcpFb);\n description.codecs.push(codec);\n // parse FEC mechanisms from rtpmap lines.\n switch (codec.name.toUpperCase()) {\n case 'RED':\n case 'ULPFEC':\n description.fecMechanisms.push(codec.name.toUpperCase());\n break;\n default: // only RED and ULPFEC are recognized as FEC mechanisms.\n break;\n }\n }\n }\n SDPUtils.matchPrefix(mediaSection, 'a=extmap:').forEach(function(line) {\n description.headerExtensions.push(SDPUtils.parseExtmap(line));\n });\n // FIXME: parse rtcp.\n return description;\n};\n\n// Generates parts of the SDP media section describing the capabilities /\n// parameters.\nSDPUtils.writeRtpDescription = function(kind, caps) {\n var sdp = '';\n\n // Build the mline.\n sdp += 'm=' + kind + ' ';\n sdp += caps.codecs.length > 0 ? '9' : '0'; // reject if no codecs.\n sdp += ' UDP/TLS/RTP/SAVPF ';\n sdp += caps.codecs.map(function(codec) {\n if (codec.preferredPayloadType !== undefined) {\n return codec.preferredPayloadType;\n }\n return codec.payloadType;\n }).join(' ') + '\\r\\n';\n\n sdp += 'c=IN IP4 0.0.0.0\\r\\n';\n sdp += 'a=rtcp:9 IN IP4 0.0.0.0\\r\\n';\n\n // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.\n caps.codecs.forEach(function(codec) {\n sdp += SDPUtils.writeRtpMap(codec);\n sdp += SDPUtils.writeFmtp(codec);\n sdp += SDPUtils.writeRtcpFb(codec);\n });\n var maxptime = 0;\n caps.codecs.forEach(function(codec) {\n if (codec.maxptime > maxptime) {\n maxptime = codec.maxptime;\n }\n });\n if (maxptime > 0) {\n sdp += 'a=maxptime:' + maxptime + '\\r\\n';\n }\n sdp += 'a=rtcp-mux\\r\\n';\n\n if (caps.headerExtensions) {\n caps.headerExtensions.forEach(function(extension) {\n sdp += SDPUtils.writeExtmap(extension);\n });\n }\n // FIXME: write fecMechanisms.\n return sdp;\n};\n\n// Parses the SDP media section and returns an array of\n// RTCRtpEncodingParameters.\nSDPUtils.parseRtpEncodingParameters = function(mediaSection) {\n var encodingParameters = [];\n var description = SDPUtils.parseRtpParameters(mediaSection);\n var hasRed = description.fecMechanisms.indexOf('RED') !== -1;\n var hasUlpfec = description.fecMechanisms.indexOf('ULPFEC') !== -1;\n\n // filter a=ssrc:... cname:, ignore PlanB-msid\n var ssrcs = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(parts) {\n return parts.attribute === 'cname';\n });\n var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;\n var secondarySsrc;\n\n var flows = SDPUtils.matchPrefix(mediaSection, 'a=ssrc-group:FID')\n .map(function(line) {\n var parts = line.substr(17).split(' ');\n return parts.map(function(part) {\n return parseInt(part, 10);\n });\n });\n if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {\n secondarySsrc = flows[0][1];\n }\n\n description.codecs.forEach(function(codec) {\n if (codec.name.toUpperCase() === 'RTX' && codec.parameters.apt) {\n var encParam = {\n ssrc: primarySsrc,\n codecPayloadType: parseInt(codec.parameters.apt, 10)\n };\n if (primarySsrc && secondarySsrc) {\n encParam.rtx = {ssrc: secondarySsrc};\n }\n encodingParameters.push(encParam);\n if (hasRed) {\n encParam = JSON.parse(JSON.stringify(encParam));\n encParam.fec = {\n ssrc: primarySsrc,\n mechanism: hasUlpfec ? 'red+ulpfec' : 'red'\n };\n encodingParameters.push(encParam);\n }\n }\n });\n if (encodingParameters.length === 0 && primarySsrc) {\n encodingParameters.push({\n ssrc: primarySsrc\n });\n }\n\n // we support both b=AS and b=TIAS but interpret AS as TIAS.\n var bandwidth = SDPUtils.matchPrefix(mediaSection, 'b=');\n if (bandwidth.length) {\n if (bandwidth[0].indexOf('b=TIAS:') === 0) {\n bandwidth = parseInt(bandwidth[0].substr(7), 10);\n } else if (bandwidth[0].indexOf('b=AS:') === 0) {\n // use formula from JSEP to convert b=AS to TIAS value.\n bandwidth = parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95\n - (50 * 40 * 8);\n } else {\n bandwidth = undefined;\n }\n encodingParameters.forEach(function(params) {\n params.maxBitrate = bandwidth;\n });\n }\n return encodingParameters;\n};\n\n// parses http://draft.ortc.org/#rtcrtcpparameters*\nSDPUtils.parseRtcpParameters = function(mediaSection) {\n var rtcpParameters = {};\n\n // Gets the first SSRC. Note tha with RTX there might be multiple\n // SSRCs.\n var remoteSsrc = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(obj) {\n return obj.attribute === 'cname';\n })[0];\n if (remoteSsrc) {\n rtcpParameters.cname = remoteSsrc.value;\n rtcpParameters.ssrc = remoteSsrc.ssrc;\n }\n\n // Edge uses the compound attribute instead of reducedSize\n // compound is !reducedSize\n var rsize = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-rsize');\n rtcpParameters.reducedSize = rsize.length > 0;\n rtcpParameters.compound = rsize.length === 0;\n\n // parses the rtcp-mux attrіbute.\n // Note that Edge does not support unmuxed RTCP.\n var mux = SDPUtils.matchPrefix(mediaSection, 'a=rtcp-mux');\n rtcpParameters.mux = mux.length > 0;\n\n return rtcpParameters;\n};\n\n// parses either a=msid: or a=ssrc:... msid lines and returns\n// the id of the MediaStream and MediaStreamTrack.\nSDPUtils.parseMsid = function(mediaSection) {\n var parts;\n var spec = SDPUtils.matchPrefix(mediaSection, 'a=msid:');\n if (spec.length === 1) {\n parts = spec[0].substr(7).split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n var planB = SDPUtils.matchPrefix(mediaSection, 'a=ssrc:')\n .map(function(line) {\n return SDPUtils.parseSsrcMedia(line);\n })\n .filter(function(msidParts) {\n return msidParts.attribute === 'msid';\n });\n if (planB.length > 0) {\n parts = planB[0].value.split(' ');\n return {stream: parts[0], track: parts[1]};\n }\n};\n\n// SCTP\n// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back\n// to draft-ietf-mmusic-sctp-sdp-05\nSDPUtils.parseSctpDescription = function(mediaSection) {\n var mline = SDPUtils.parseMLine(mediaSection);\n var maxSizeLine = SDPUtils.matchPrefix(mediaSection, 'a=max-message-size:');\n var maxMessageSize;\n if (maxSizeLine.length > 0) {\n maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);\n }\n if (isNaN(maxMessageSize)) {\n maxMessageSize = 65536;\n }\n var sctpPort = SDPUtils.matchPrefix(mediaSection, 'a=sctp-port:');\n if (sctpPort.length > 0) {\n return {\n port: parseInt(sctpPort[0].substr(12), 10),\n protocol: mline.fmt,\n maxMessageSize: maxMessageSize\n };\n }\n var sctpMapLines = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:');\n if (sctpMapLines.length > 0) {\n var parts = SDPUtils.matchPrefix(mediaSection, 'a=sctpmap:')[0]\n .substr(10)\n .split(' ');\n return {\n port: parseInt(parts[0], 10),\n protocol: parts[1],\n maxMessageSize: maxMessageSize\n };\n }\n};\n\n// SCTP\n// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers\n// support by now receiving in this format, unless we originally parsed\n// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line\n// protocol of DTLS/SCTP -- without UDP/ or TCP/)\nSDPUtils.writeSctpDescription = function(media, sctp) {\n var output = [];\n if (media.protocol !== 'DTLS/SCTP') {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.protocol + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctp-port:' + sctp.port + '\\r\\n'\n ];\n } else {\n output = [\n 'm=' + media.kind + ' 9 ' + media.protocol + ' ' + sctp.port + '\\r\\n',\n 'c=IN IP4 0.0.0.0\\r\\n',\n 'a=sctpmap:' + sctp.port + ' ' + sctp.protocol + ' 65535\\r\\n'\n ];\n }\n if (sctp.maxMessageSize !== undefined) {\n output.push('a=max-message-size:' + sctp.maxMessageSize + '\\r\\n');\n }\n return output.join('');\n};\n\n// Generate a session ID for SDP.\n// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1\n// recommends using a cryptographically random +ve 64-bit value\n// but right now this should be acceptable and within the right range\nSDPUtils.generateSessionId = function() {\n return Math.random().toString().substr(2, 21);\n};\n\n// Write boilder plate for start of SDP\n// sessId argument is optional - if not supplied it will\n// be generated randomly\n// sessVersion is optional and defaults to 2\n// sessUser is optional and defaults to 'thisisadapterortc'\nSDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {\n var sessionId;\n var version = sessVer !== undefined ? sessVer : 2;\n if (sessId) {\n sessionId = sessId;\n } else {\n sessionId = SDPUtils.generateSessionId();\n }\n var user = sessUser || 'thisisadapterortc';\n // FIXME: sess-id should be an NTP timestamp.\n return 'v=0\\r\\n' +\n 'o=' + user + ' ' + sessionId + ' ' + version +\n ' IN IP4 127.0.0.1\\r\\n' +\n 's=-\\r\\n' +\n 't=0 0\\r\\n';\n};\n\nSDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.direction) {\n sdp += 'a=' + transceiver.direction + '\\r\\n';\n } else if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n // spec.\n var msid = 'msid:' + stream.id + ' ' +\n transceiver.rtpSender.track.id + '\\r\\n';\n sdp += 'a=' + msid;\n\n // for Chrome.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n};\n\n// Gets the direction from the mediaSection or the sessionpart.\nSDPUtils.getDirection = function(mediaSection, sessionpart) {\n // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.\n var lines = SDPUtils.splitLines(mediaSection);\n for (var i = 0; i < lines.length; i++) {\n switch (lines[i]) {\n case 'a=sendrecv':\n case 'a=sendonly':\n case 'a=recvonly':\n case 'a=inactive':\n return lines[i].substr(2);\n default:\n // FIXME: What should happen here?\n }\n }\n if (sessionpart) {\n return SDPUtils.getDirection(sessionpart);\n }\n return 'sendrecv';\n};\n\nSDPUtils.getKind = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var mline = lines[0].split(' ');\n return mline[0].substr(2);\n};\n\nSDPUtils.isRejected = function(mediaSection) {\n return mediaSection.split(' ', 2)[1] === '0';\n};\n\nSDPUtils.parseMLine = function(mediaSection) {\n var lines = SDPUtils.splitLines(mediaSection);\n var parts = lines[0].substr(2).split(' ');\n return {\n kind: parts[0],\n port: parseInt(parts[1], 10),\n protocol: parts[2],\n fmt: parts.slice(3).join(' ')\n };\n};\n\nSDPUtils.parseOLine = function(mediaSection) {\n var line = SDPUtils.matchPrefix(mediaSection, 'o=')[0];\n var parts = line.substr(2).split(' ');\n return {\n username: parts[0],\n sessionId: parts[1],\n sessionVersion: parseInt(parts[2], 10),\n netType: parts[3],\n addressType: parts[4],\n address: parts[5]\n };\n};\n\n// a very naive interpretation of a valid SDP.\nSDPUtils.isValidSDP = function(blob) {\n if (typeof blob !== 'string' || blob.length === 0) {\n return false;\n }\n var lines = SDPUtils.splitLines(blob);\n for (var i = 0; i < lines.length; i++) {\n if (lines[i].length < 2 || lines[i].charAt(1) !== '=') {\n return false;\n }\n // TODO: check the modifier a bit more.\n }\n return true;\n};\n\n// Expose public methods.\nif (typeof module === 'object') {\n module.exports = SDPUtils;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nvar SDPUtils = require('sdp');\n\nfunction fixStatsType(stat) {\n return {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n }[stat.type] || stat.type;\n}\n\nfunction writeMediaSection(transceiver, caps, type, stream, dtlsRole) {\n var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);\n\n // Map ICE parameters (ufrag, pwd) to SDP.\n sdp += SDPUtils.writeIceParameters(\n transceiver.iceGatherer.getLocalParameters());\n\n // Map DTLS parameters to SDP.\n sdp += SDPUtils.writeDtlsParameters(\n transceiver.dtlsTransport.getLocalParameters(),\n type === 'offer' ? 'actpass' : dtlsRole || 'active');\n\n sdp += 'a=mid:' + transceiver.mid + '\\r\\n';\n\n if (transceiver.rtpSender && transceiver.rtpReceiver) {\n sdp += 'a=sendrecv\\r\\n';\n } else if (transceiver.rtpSender) {\n sdp += 'a=sendonly\\r\\n';\n } else if (transceiver.rtpReceiver) {\n sdp += 'a=recvonly\\r\\n';\n } else {\n sdp += 'a=inactive\\r\\n';\n }\n\n if (transceiver.rtpSender) {\n var trackId = transceiver.rtpSender._initialTrackId ||\n transceiver.rtpSender.track.id;\n transceiver.rtpSender._initialTrackId = trackId;\n // spec.\n var msid = 'msid:' + (stream ? stream.id : '-') + ' ' +\n trackId + '\\r\\n';\n sdp += 'a=' + msid;\n // for Chrome. Legacy should no longer be required.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' ' + msid;\n\n // RTX\n if (transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' ' + msid;\n sdp += 'a=ssrc-group:FID ' +\n transceiver.sendEncodingParameters[0].ssrc + ' ' +\n transceiver.sendEncodingParameters[0].rtx.ssrc +\n '\\r\\n';\n }\n }\n // FIXME: this should be written by writeRtpDescription.\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {\n sdp += 'a=ssrc:' + transceiver.sendEncodingParameters[0].rtx.ssrc +\n ' cname:' + SDPUtils.localCName + '\\r\\n';\n }\n return sdp;\n}\n\n// Edge does not like\n// 1) stun: filtered after 14393 unless ?transport=udp is present\n// 2) turn: that does not have all of turn:host:port?transport=udp\n// 3) turn: with ipv6 addresses\n// 4) turn: occurring muliple times\nfunction filterIceServers(iceServers, edgeVersion) {\n var hasTurn = false;\n iceServers = JSON.parse(JSON.stringify(iceServers));\n return iceServers.filter(function(server) {\n if (server && (server.urls || server.url)) {\n var urls = server.urls || server.url;\n if (server.url && !server.urls) {\n console.warn('RTCIceServer.url is deprecated! Use urls instead.');\n }\n var isString = typeof urls === 'string';\n if (isString) {\n urls = [urls];\n }\n urls = urls.filter(function(url) {\n var validTurn = url.indexOf('turn:') === 0 &&\n url.indexOf('transport=udp') !== -1 &&\n url.indexOf('turn:[') === -1 &&\n !hasTurn;\n\n if (validTurn) {\n hasTurn = true;\n return true;\n }\n return url.indexOf('stun:') === 0 && edgeVersion >= 14393 &&\n url.indexOf('?transport=udp') === -1;\n });\n\n delete server.url;\n server.urls = isString ? urls[0] : urls;\n return !!urls.length;\n }\n });\n}\n\n// Determines the intersection of local and remote capabilities.\nfunction getCommonCapabilities(localCapabilities, remoteCapabilities) {\n var commonCapabilities = {\n codecs: [],\n headerExtensions: [],\n fecMechanisms: []\n };\n\n var findCodecByPayloadType = function(pt, codecs) {\n pt = parseInt(pt, 10);\n for (var i = 0; i < codecs.length; i++) {\n if (codecs[i].payloadType === pt ||\n codecs[i].preferredPayloadType === pt) {\n return codecs[i];\n }\n }\n };\n\n var rtxCapabilityMatches = function(lRtx, rRtx, lCodecs, rCodecs) {\n var lCodec = findCodecByPayloadType(lRtx.parameters.apt, lCodecs);\n var rCodec = findCodecByPayloadType(rRtx.parameters.apt, rCodecs);\n return lCodec && rCodec &&\n lCodec.name.toLowerCase() === rCodec.name.toLowerCase();\n };\n\n localCapabilities.codecs.forEach(function(lCodec) {\n for (var i = 0; i < remoteCapabilities.codecs.length; i++) {\n var rCodec = remoteCapabilities.codecs[i];\n if (lCodec.name.toLowerCase() === rCodec.name.toLowerCase() &&\n lCodec.clockRate === rCodec.clockRate) {\n if (lCodec.name.toLowerCase() === 'rtx' &&\n lCodec.parameters && rCodec.parameters.apt) {\n // for RTX we need to find the local rtx that has a apt\n // which points to the same local codec as the remote one.\n if (!rtxCapabilityMatches(lCodec, rCodec,\n localCapabilities.codecs, remoteCapabilities.codecs)) {\n continue;\n }\n }\n rCodec = JSON.parse(JSON.stringify(rCodec)); // deepcopy\n // number of channels is the highest common number of channels\n rCodec.numChannels = Math.min(lCodec.numChannels,\n rCodec.numChannels);\n // push rCodec so we reply with offerer payload type\n commonCapabilities.codecs.push(rCodec);\n\n // determine common feedback mechanisms\n rCodec.rtcpFeedback = rCodec.rtcpFeedback.filter(function(fb) {\n for (var j = 0; j < lCodec.rtcpFeedback.length; j++) {\n if (lCodec.rtcpFeedback[j].type === fb.type &&\n lCodec.rtcpFeedback[j].parameter === fb.parameter) {\n return true;\n }\n }\n return false;\n });\n // FIXME: also need to determine .parameters\n // see https://github.com/openpeer/ortc/issues/569\n break;\n }\n }\n });\n\n localCapabilities.headerExtensions.forEach(function(lHeaderExtension) {\n for (var i = 0; i < remoteCapabilities.headerExtensions.length;\n i++) {\n var rHeaderExtension = remoteCapabilities.headerExtensions[i];\n if (lHeaderExtension.uri === rHeaderExtension.uri) {\n commonCapabilities.headerExtensions.push(rHeaderExtension);\n break;\n }\n }\n });\n\n // FIXME: fecMechanisms\n return commonCapabilities;\n}\n\n// is action=setLocalDescription with type allowed in signalingState\nfunction isActionAllowedInSignalingState(action, type, signalingState) {\n return {\n offer: {\n setLocalDescription: ['stable', 'have-local-offer'],\n setRemoteDescription: ['stable', 'have-remote-offer']\n },\n answer: {\n setLocalDescription: ['have-remote-offer', 'have-local-pranswer'],\n setRemoteDescription: ['have-local-offer', 'have-remote-pranswer']\n }\n }[type][action].indexOf(signalingState) !== -1;\n}\n\nfunction maybeAddCandidate(iceTransport, candidate) {\n // Edge's internal representation adds some fields therefore\n // not all fieldѕ are taken into account.\n var alreadyAdded = iceTransport.getRemoteCandidates()\n .find(function(remoteCandidate) {\n return candidate.foundation === remoteCandidate.foundation &&\n candidate.ip === remoteCandidate.ip &&\n candidate.port === remoteCandidate.port &&\n candidate.priority === remoteCandidate.priority &&\n candidate.protocol === remoteCandidate.protocol &&\n candidate.type === remoteCandidate.type;\n });\n if (!alreadyAdded) {\n iceTransport.addRemoteCandidate(candidate);\n }\n return !alreadyAdded;\n}\n\n\nfunction makeError(name, description) {\n var e = new Error(description);\n e.name = name;\n // legacy error codes from https://heycam.github.io/webidl/#idl-DOMException-error-names\n e.code = {\n NotSupportedError: 9,\n InvalidStateError: 11,\n InvalidAccessError: 15,\n TypeError: undefined,\n OperationError: undefined\n }[name];\n return e;\n}\n\nmodule.exports = function(window, edgeVersion) {\n // https://w3c.github.io/mediacapture-main/#mediastream\n // Helper function to add the track to the stream and\n // dispatch the event ourselves.\n function addTrackToStreamAndFireEvent(track, stream) {\n stream.addTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('addtrack',\n {track: track}));\n }\n\n function removeTrackFromStreamAndFireEvent(track, stream) {\n stream.removeTrack(track);\n stream.dispatchEvent(new window.MediaStreamTrackEvent('removetrack',\n {track: track}));\n }\n\n function fireAddTrack(pc, track, receiver, streams) {\n var trackEvent = new Event('track');\n trackEvent.track = track;\n trackEvent.receiver = receiver;\n trackEvent.transceiver = {receiver: receiver};\n trackEvent.streams = streams;\n window.setTimeout(function() {\n pc._dispatchEvent('track', trackEvent);\n });\n }\n\n var RTCPeerConnection = function(config) {\n var pc = this;\n\n var _eventTarget = document.createDocumentFragment();\n ['addEventListener', 'removeEventListener', 'dispatchEvent']\n .forEach(function(method) {\n pc[method] = _eventTarget[method].bind(_eventTarget);\n });\n\n this.canTrickleIceCandidates = null;\n\n this.needNegotiation = false;\n\n this.localStreams = [];\n this.remoteStreams = [];\n\n this._localDescription = null;\n this._remoteDescription = null;\n\n this.signalingState = 'stable';\n this.iceConnectionState = 'new';\n this.connectionState = 'new';\n this.iceGatheringState = 'new';\n\n config = JSON.parse(JSON.stringify(config || {}));\n\n this.usingBundle = config.bundlePolicy === 'max-bundle';\n if (config.rtcpMuxPolicy === 'negotiate') {\n throw(makeError('NotSupportedError',\n 'rtcpMuxPolicy \\'negotiate\\' is not supported'));\n } else if (!config.rtcpMuxPolicy) {\n config.rtcpMuxPolicy = 'require';\n }\n\n switch (config.iceTransportPolicy) {\n case 'all':\n case 'relay':\n break;\n default:\n config.iceTransportPolicy = 'all';\n break;\n }\n\n switch (config.bundlePolicy) {\n case 'balanced':\n case 'max-compat':\n case 'max-bundle':\n break;\n default:\n config.bundlePolicy = 'balanced';\n break;\n }\n\n config.iceServers = filterIceServers(config.iceServers || [], edgeVersion);\n\n this._iceGatherers = [];\n if (config.iceCandidatePoolSize) {\n for (var i = config.iceCandidatePoolSize; i > 0; i--) {\n this._iceGatherers.push(new window.RTCIceGatherer({\n iceServers: config.iceServers,\n gatherPolicy: config.iceTransportPolicy\n }));\n }\n } else {\n config.iceCandidatePoolSize = 0;\n }\n\n this._config = config;\n\n // per-track iceGathers, iceTransports, dtlsTransports, rtpSenders, ...\n // everything that is needed to describe a SDP m-line.\n this.transceivers = [];\n\n this._sdpSessionId = SDPUtils.generateSessionId();\n this._sdpSessionVersion = 0;\n\n this._dtlsRole = undefined; // role for a=setup to use in answers.\n\n this._isClosed = false;\n };\n\n Object.defineProperty(RTCPeerConnection.prototype, 'localDescription', {\n configurable: true,\n get: function() {\n return this._localDescription;\n }\n });\n Object.defineProperty(RTCPeerConnection.prototype, 'remoteDescription', {\n configurable: true,\n get: function() {\n return this._remoteDescription;\n }\n });\n\n // set up event handlers on prototype\n RTCPeerConnection.prototype.onicecandidate = null;\n RTCPeerConnection.prototype.onaddstream = null;\n RTCPeerConnection.prototype.ontrack = null;\n RTCPeerConnection.prototype.onremovestream = null;\n RTCPeerConnection.prototype.onsignalingstatechange = null;\n RTCPeerConnection.prototype.oniceconnectionstatechange = null;\n RTCPeerConnection.prototype.onconnectionstatechange = null;\n RTCPeerConnection.prototype.onicegatheringstatechange = null;\n RTCPeerConnection.prototype.onnegotiationneeded = null;\n RTCPeerConnection.prototype.ondatachannel = null;\n\n RTCPeerConnection.prototype._dispatchEvent = function(name, event) {\n if (this._isClosed) {\n return;\n }\n this.dispatchEvent(event);\n if (typeof this['on' + name] === 'function') {\n this['on' + name](event);\n }\n };\n\n RTCPeerConnection.prototype._emitGatheringStateChange = function() {\n var event = new Event('icegatheringstatechange');\n this._dispatchEvent('icegatheringstatechange', event);\n };\n\n RTCPeerConnection.prototype.getConfiguration = function() {\n return this._config;\n };\n\n RTCPeerConnection.prototype.getLocalStreams = function() {\n return this.localStreams;\n };\n\n RTCPeerConnection.prototype.getRemoteStreams = function() {\n return this.remoteStreams;\n };\n\n // internal helper to create a transceiver object.\n // (which is not yet the same as the WebRTC 1.0 transceiver)\n RTCPeerConnection.prototype._createTransceiver = function(kind, doNotAdd) {\n var hasBundleTransport = this.transceivers.length > 0;\n var transceiver = {\n track: null,\n iceGatherer: null,\n iceTransport: null,\n dtlsTransport: null,\n localCapabilities: null,\n remoteCapabilities: null,\n rtpSender: null,\n rtpReceiver: null,\n kind: kind,\n mid: null,\n sendEncodingParameters: null,\n recvEncodingParameters: null,\n stream: null,\n associatedRemoteMediaStreams: [],\n wantReceive: true\n };\n if (this.usingBundle && hasBundleTransport) {\n transceiver.iceTransport = this.transceivers[0].iceTransport;\n transceiver.dtlsTransport = this.transceivers[0].dtlsTransport;\n } else {\n var transports = this._createIceAndDtlsTransports();\n transceiver.iceTransport = transports.iceTransport;\n transceiver.dtlsTransport = transports.dtlsTransport;\n }\n if (!doNotAdd) {\n this.transceivers.push(transceiver);\n }\n return transceiver;\n };\n\n RTCPeerConnection.prototype.addTrack = function(track, stream) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call addTrack on a closed peerconnection.');\n }\n\n var alreadyExists = this.transceivers.find(function(s) {\n return s.track === track;\n });\n\n if (alreadyExists) {\n throw makeError('InvalidAccessError', 'Track already exists.');\n }\n\n var transceiver;\n for (var i = 0; i < this.transceivers.length; i++) {\n if (!this.transceivers[i].track &&\n this.transceivers[i].kind === track.kind) {\n transceiver = this.transceivers[i];\n }\n }\n if (!transceiver) {\n transceiver = this._createTransceiver(track.kind);\n }\n\n this._maybeFireNegotiationNeeded();\n\n if (this.localStreams.indexOf(stream) === -1) {\n this.localStreams.push(stream);\n }\n\n transceiver.track = track;\n transceiver.stream = stream;\n transceiver.rtpSender = new window.RTCRtpSender(track,\n transceiver.dtlsTransport);\n return transceiver.rtpSender;\n };\n\n RTCPeerConnection.prototype.addStream = function(stream) {\n var pc = this;\n if (edgeVersion >= 15025) {\n stream.getTracks().forEach(function(track) {\n pc.addTrack(track, stream);\n });\n } else {\n // Clone is necessary for local demos mostly, attaching directly\n // to two different senders does not work (build 10547).\n // Fixed in 15025 (or earlier)\n var clonedStream = stream.clone();\n stream.getTracks().forEach(function(track, idx) {\n var clonedTrack = clonedStream.getTracks()[idx];\n track.addEventListener('enabled', function(event) {\n clonedTrack.enabled = event.enabled;\n });\n });\n clonedStream.getTracks().forEach(function(track) {\n pc.addTrack(track, clonedStream);\n });\n }\n };\n\n RTCPeerConnection.prototype.removeTrack = function(sender) {\n if (this._isClosed) {\n throw makeError('InvalidStateError',\n 'Attempted to call removeTrack on a closed peerconnection.');\n }\n\n if (!(sender instanceof window.RTCRtpSender)) {\n throw new TypeError('Argument 1 of RTCPeerConnection.removeTrack ' +\n 'does not implement interface RTCRtpSender.');\n }\n\n var transceiver = this.transceivers.find(function(t) {\n return t.rtpSender === sender;\n });\n\n if (!transceiver) {\n throw makeError('InvalidAccessError',\n 'Sender was not created by this connection.');\n }\n var stream = transceiver.stream;\n\n transceiver.rtpSender.stop();\n transceiver.rtpSender = null;\n transceiver.track = null;\n transceiver.stream = null;\n\n // remove the stream from the set of local streams\n var localStreams = this.transceivers.map(function(t) {\n return t.stream;\n });\n if (localStreams.indexOf(stream) === -1 &&\n this.localStreams.indexOf(stream) > -1) {\n this.localStreams.splice(this.localStreams.indexOf(stream), 1);\n }\n\n this._maybeFireNegotiationNeeded();\n };\n\n RTCPeerConnection.prototype.removeStream = function(stream) {\n var pc = this;\n stream.getTracks().forEach(function(track) {\n var sender = pc.getSenders().find(function(s) {\n return s.track === track;\n });\n if (sender) {\n pc.removeTrack(sender);\n }\n });\n };\n\n RTCPeerConnection.prototype.getSenders = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpSender;\n })\n .map(function(transceiver) {\n return transceiver.rtpSender;\n });\n };\n\n RTCPeerConnection.prototype.getReceivers = function() {\n return this.transceivers.filter(function(transceiver) {\n return !!transceiver.rtpReceiver;\n })\n .map(function(transceiver) {\n return transceiver.rtpReceiver;\n });\n };\n\n\n RTCPeerConnection.prototype._createIceGatherer = function(sdpMLineIndex,\n usingBundle) {\n var pc = this;\n if (usingBundle && sdpMLineIndex > 0) {\n return this.transceivers[0].iceGatherer;\n } else if (this._iceGatherers.length) {\n return this._iceGatherers.shift();\n }\n var iceGatherer = new window.RTCIceGatherer({\n iceServers: this._config.iceServers,\n gatherPolicy: this._config.iceTransportPolicy\n });\n Object.defineProperty(iceGatherer, 'state',\n {value: 'new', writable: true}\n );\n\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = [];\n this.transceivers[sdpMLineIndex].bufferCandidates = function(event) {\n var end = !event.candidate || Object.keys(event.candidate).length === 0;\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n iceGatherer.state = end ? 'completed' : 'gathering';\n if (pc.transceivers[sdpMLineIndex].bufferedCandidateEvents !== null) {\n pc.transceivers[sdpMLineIndex].bufferedCandidateEvents.push(event);\n }\n };\n iceGatherer.addEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n return iceGatherer;\n };\n\n // start gathering from an RTCIceGatherer.\n RTCPeerConnection.prototype._gather = function(mid, sdpMLineIndex) {\n var pc = this;\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer.onlocalcandidate) {\n return;\n }\n var bufferedCandidateEvents =\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents;\n this.transceivers[sdpMLineIndex].bufferedCandidateEvents = null;\n iceGatherer.removeEventListener('localcandidate',\n this.transceivers[sdpMLineIndex].bufferCandidates);\n iceGatherer.onlocalcandidate = function(evt) {\n if (pc.usingBundle && sdpMLineIndex > 0) {\n // if we know that we use bundle we can drop candidates with\n // ѕdpMLineIndex > 0. If we don't do this then our state gets\n // confused since we dispose the extra ice gatherer.\n return;\n }\n var event = new Event('icecandidate');\n event.candidate = {sdpMid: mid, sdpMLineIndex: sdpMLineIndex};\n\n var cand = evt.candidate;\n // Edge emits an empty object for RTCIceCandidateComplete‥\n var end = !cand || Object.keys(cand).length === 0;\n if (end) {\n // polyfill since RTCIceGatherer.state is not implemented in\n // Edge 10547 yet.\n if (iceGatherer.state === 'new' || iceGatherer.state === 'gathering') {\n iceGatherer.state = 'completed';\n }\n } else {\n if (iceGatherer.state === 'new') {\n iceGatherer.state = 'gathering';\n }\n // RTCIceCandidate doesn't have a component, needs to be added\n cand.component = 1;\n // also the usernameFragment. TODO: update SDP to take both variants.\n cand.ufrag = iceGatherer.getLocalParameters().usernameFragment;\n\n var serializedCandidate = SDPUtils.writeCandidate(cand);\n event.candidate = Object.assign(event.candidate,\n SDPUtils.parseCandidate(serializedCandidate));\n\n event.candidate.candidate = serializedCandidate;\n event.candidate.toJSON = function() {\n return {\n candidate: event.candidate.candidate,\n sdpMid: event.candidate.sdpMid,\n sdpMLineIndex: event.candidate.sdpMLineIndex,\n usernameFragment: event.candidate.usernameFragment\n };\n };\n }\n\n // update local description.\n var sections = SDPUtils.getMediaSections(pc._localDescription.sdp);\n if (!end) {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=' + event.candidate.candidate + '\\r\\n';\n } else {\n sections[event.candidate.sdpMLineIndex] +=\n 'a=end-of-candidates\\r\\n';\n }\n pc._localDescription.sdp =\n SDPUtils.getDescription(pc._localDescription.sdp) +\n sections.join('');\n var complete = pc.transceivers.every(function(transceiver) {\n return transceiver.iceGatherer &&\n transceiver.iceGatherer.state === 'completed';\n });\n\n if (pc.iceGatheringState !== 'gathering') {\n pc.iceGatheringState = 'gathering';\n pc._emitGatheringStateChange();\n }\n\n // Emit candidate. Also emit null candidate when all gatherers are\n // complete.\n if (!end) {\n pc._dispatchEvent('icecandidate', event);\n }\n if (complete) {\n pc._dispatchEvent('icecandidate', new Event('icecandidate'));\n pc.iceGatheringState = 'complete';\n pc._emitGatheringStateChange();\n }\n };\n\n // emit already gathered candidates.\n window.setTimeout(function() {\n bufferedCandidateEvents.forEach(function(e) {\n iceGatherer.onlocalcandidate(e);\n });\n }, 0);\n };\n\n // Create ICE transport and DTLS transport.\n RTCPeerConnection.prototype._createIceAndDtlsTransports = function() {\n var pc = this;\n var iceTransport = new window.RTCIceTransport(null);\n iceTransport.onicestatechange = function() {\n pc._updateIceConnectionState();\n pc._updateConnectionState();\n };\n\n var dtlsTransport = new window.RTCDtlsTransport(iceTransport);\n dtlsTransport.ondtlsstatechange = function() {\n pc._updateConnectionState();\n };\n dtlsTransport.onerror = function() {\n // onerror does not set state to failed by itself.\n Object.defineProperty(dtlsTransport, 'state',\n {value: 'failed', writable: true});\n pc._updateConnectionState();\n };\n\n return {\n iceTransport: iceTransport,\n dtlsTransport: dtlsTransport\n };\n };\n\n // Destroy ICE gatherer, ICE transport and DTLS transport.\n // Without triggering the callbacks.\n RTCPeerConnection.prototype._disposeIceAndDtlsTransports = function(\n sdpMLineIndex) {\n var iceGatherer = this.transceivers[sdpMLineIndex].iceGatherer;\n if (iceGatherer) {\n delete iceGatherer.onlocalcandidate;\n delete this.transceivers[sdpMLineIndex].iceGatherer;\n }\n var iceTransport = this.transceivers[sdpMLineIndex].iceTransport;\n if (iceTransport) {\n delete iceTransport.onicestatechange;\n delete this.transceivers[sdpMLineIndex].iceTransport;\n }\n var dtlsTransport = this.transceivers[sdpMLineIndex].dtlsTransport;\n if (dtlsTransport) {\n delete dtlsTransport.ondtlsstatechange;\n delete dtlsTransport.onerror;\n delete this.transceivers[sdpMLineIndex].dtlsTransport;\n }\n };\n\n // Start the RTP Sender and Receiver for a transceiver.\n RTCPeerConnection.prototype._transceive = function(transceiver,\n send, recv) {\n var params = getCommonCapabilities(transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n if (send && transceiver.rtpSender) {\n params.encodings = transceiver.sendEncodingParameters;\n params.rtcp = {\n cname: SDPUtils.localCName,\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.recvEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.recvEncodingParameters[0].ssrc;\n }\n transceiver.rtpSender.send(params);\n }\n if (recv && transceiver.rtpReceiver && params.codecs.length > 0) {\n // remove RTX field in Edge 14942\n if (transceiver.kind === 'video'\n && transceiver.recvEncodingParameters\n && edgeVersion < 15019) {\n transceiver.recvEncodingParameters.forEach(function(p) {\n delete p.rtx;\n });\n }\n if (transceiver.recvEncodingParameters.length) {\n params.encodings = transceiver.recvEncodingParameters;\n } else {\n params.encodings = [{}];\n }\n params.rtcp = {\n compound: transceiver.rtcpParameters.compound\n };\n if (transceiver.rtcpParameters.cname) {\n params.rtcp.cname = transceiver.rtcpParameters.cname;\n }\n if (transceiver.sendEncodingParameters.length) {\n params.rtcp.ssrc = transceiver.sendEncodingParameters[0].ssrc;\n }\n transceiver.rtpReceiver.receive(params);\n }\n };\n\n RTCPeerConnection.prototype.setLocalDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setLocalDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set local ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var sections;\n var sessionpart;\n if (description.type === 'offer') {\n // VERY limited support for SDP munging. Limited to:\n // * changing the order of codecs\n sections = SDPUtils.splitSections(description.sdp);\n sessionpart = sections.shift();\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var caps = SDPUtils.parseRtpParameters(mediaSection);\n pc.transceivers[sdpMLineIndex].localCapabilities = caps;\n });\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n });\n } else if (description.type === 'answer') {\n sections = SDPUtils.splitSections(pc._remoteDescription.sdp);\n sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var transceiver = pc.transceivers[sdpMLineIndex];\n var iceGatherer = transceiver.iceGatherer;\n var iceTransport = transceiver.iceTransport;\n var dtlsTransport = transceiver.dtlsTransport;\n var localCapabilities = transceiver.localCapabilities;\n var remoteCapabilities = transceiver.remoteCapabilities;\n\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n\n if (!rejected && !transceiver.rejected) {\n var remoteIceParameters = SDPUtils.getIceParameters(\n mediaSection, sessionpart);\n var remoteDtlsParameters = SDPUtils.getDtlsParameters(\n mediaSection, sessionpart);\n if (isIceLite) {\n remoteDtlsParameters.role = 'server';\n }\n\n if (!pc.usingBundle || sdpMLineIndex === 0) {\n pc._gather(transceiver.mid, sdpMLineIndex);\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n isIceLite ? 'controlling' : 'controlled');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // Calculate intersection of capabilities.\n var params = getCommonCapabilities(localCapabilities,\n remoteCapabilities);\n\n // Start the RTCRtpSender. The RTCRtpReceiver for this\n // transceiver has already been started in setRemoteDescription.\n pc._transceive(transceiver,\n params.codecs.length > 0,\n false);\n }\n });\n }\n\n pc._localDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-local-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.setRemoteDescription = function(description) {\n var pc = this;\n\n // Note: pranswer is not supported.\n if (['offer', 'answer'].indexOf(description.type) === -1) {\n return Promise.reject(makeError('TypeError',\n 'Unsupported type \"' + description.type + '\"'));\n }\n\n if (!isActionAllowedInSignalingState('setRemoteDescription',\n description.type, pc.signalingState) || pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not set remote ' + description.type +\n ' in state ' + pc.signalingState));\n }\n\n var streams = {};\n pc.remoteStreams.forEach(function(stream) {\n streams[stream.id] = stream;\n });\n var receiverList = [];\n var sections = SDPUtils.splitSections(description.sdp);\n var sessionpart = sections.shift();\n var isIceLite = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-lite').length > 0;\n var usingBundle = SDPUtils.matchPrefix(sessionpart,\n 'a=group:BUNDLE ').length > 0;\n pc.usingBundle = usingBundle;\n var iceOptions = SDPUtils.matchPrefix(sessionpart,\n 'a=ice-options:')[0];\n if (iceOptions) {\n pc.canTrickleIceCandidates = iceOptions.substr(14).split(' ')\n .indexOf('trickle') >= 0;\n } else {\n pc.canTrickleIceCandidates = false;\n }\n\n sections.forEach(function(mediaSection, sdpMLineIndex) {\n var lines = SDPUtils.splitLines(mediaSection);\n var kind = SDPUtils.getKind(mediaSection);\n // treat bundle-only as not-rejected.\n var rejected = SDPUtils.isRejected(mediaSection) &&\n SDPUtils.matchPrefix(mediaSection, 'a=bundle-only').length === 0;\n var protocol = lines[0].substr(2).split(' ')[2];\n\n var direction = SDPUtils.getDirection(mediaSection, sessionpart);\n var remoteMsid = SDPUtils.parseMsid(mediaSection);\n\n var mid = SDPUtils.getMid(mediaSection) || SDPUtils.generateIdentifier();\n\n // Reject datachannels which are not implemented yet.\n if (rejected || (kind === 'application' && (protocol === 'DTLS/SCTP' ||\n protocol === 'UDP/DTLS/SCTP'))) {\n // TODO: this is dangerous in the case where a non-rejected m-line\n // becomes rejected.\n pc.transceivers[sdpMLineIndex] = {\n mid: mid,\n kind: kind,\n protocol: protocol,\n rejected: true\n };\n return;\n }\n\n if (!rejected && pc.transceivers[sdpMLineIndex] &&\n pc.transceivers[sdpMLineIndex].rejected) {\n // recycle a rejected transceiver.\n pc.transceivers[sdpMLineIndex] = pc._createTransceiver(kind, true);\n }\n\n var transceiver;\n var iceGatherer;\n var iceTransport;\n var dtlsTransport;\n var rtpReceiver;\n var sendEncodingParameters;\n var recvEncodingParameters;\n var localCapabilities;\n\n var track;\n // FIXME: ensure the mediaSection has rtcp-mux set.\n var remoteCapabilities = SDPUtils.parseRtpParameters(mediaSection);\n var remoteIceParameters;\n var remoteDtlsParameters;\n if (!rejected) {\n remoteIceParameters = SDPUtils.getIceParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters = SDPUtils.getDtlsParameters(mediaSection,\n sessionpart);\n remoteDtlsParameters.role = 'client';\n }\n recvEncodingParameters =\n SDPUtils.parseRtpEncodingParameters(mediaSection);\n\n var rtcpParameters = SDPUtils.parseRtcpParameters(mediaSection);\n\n var isComplete = SDPUtils.matchPrefix(mediaSection,\n 'a=end-of-candidates', sessionpart).length > 0;\n var cands = SDPUtils.matchPrefix(mediaSection, 'a=candidate:')\n .map(function(cand) {\n return SDPUtils.parseCandidate(cand);\n })\n .filter(function(cand) {\n return cand.component === 1;\n });\n\n // Check if we can use BUNDLE and dispose transports.\n if ((description.type === 'offer' || description.type === 'answer') &&\n !rejected && usingBundle && sdpMLineIndex > 0 &&\n pc.transceivers[sdpMLineIndex]) {\n pc._disposeIceAndDtlsTransports(sdpMLineIndex);\n pc.transceivers[sdpMLineIndex].iceGatherer =\n pc.transceivers[0].iceGatherer;\n pc.transceivers[sdpMLineIndex].iceTransport =\n pc.transceivers[0].iceTransport;\n pc.transceivers[sdpMLineIndex].dtlsTransport =\n pc.transceivers[0].dtlsTransport;\n if (pc.transceivers[sdpMLineIndex].rtpSender) {\n pc.transceivers[sdpMLineIndex].rtpSender.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n if (pc.transceivers[sdpMLineIndex].rtpReceiver) {\n pc.transceivers[sdpMLineIndex].rtpReceiver.setTransport(\n pc.transceivers[0].dtlsTransport);\n }\n }\n if (description.type === 'offer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex] ||\n pc._createTransceiver(kind);\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n usingBundle);\n }\n\n if (cands.length && transceiver.iceTransport.state === 'new') {\n if (isComplete && (!usingBundle || sdpMLineIndex === 0)) {\n transceiver.iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n localCapabilities = window.RTCRtpReceiver.getCapabilities(kind);\n\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n\n sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 2) * 1001\n }];\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n var isNewTrack = false;\n if (direction === 'sendrecv' || direction === 'sendonly') {\n isNewTrack = !transceiver.rtpReceiver;\n rtpReceiver = transceiver.rtpReceiver ||\n new window.RTCRtpReceiver(transceiver.dtlsTransport, kind);\n\n if (isNewTrack) {\n var stream;\n track = rtpReceiver.track;\n // FIXME: does not work with Plan B.\n if (remoteMsid && remoteMsid.stream === '-') {\n // no-op. a stream id of '-' means: no associated stream.\n } else if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n Object.defineProperty(streams[remoteMsid.stream], 'id', {\n get: function() {\n return remoteMsid.stream;\n }\n });\n }\n Object.defineProperty(track, 'id', {\n get: function() {\n return remoteMsid.track;\n }\n });\n stream = streams[remoteMsid.stream];\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n stream = streams.default;\n }\n if (stream) {\n addTrackToStreamAndFireEvent(track, stream);\n transceiver.associatedRemoteMediaStreams.push(stream);\n }\n receiverList.push([track, rtpReceiver, stream]);\n }\n } else if (transceiver.rtpReceiver && transceiver.rtpReceiver.track) {\n transceiver.associatedRemoteMediaStreams.forEach(function(s) {\n var nativeTrack = s.getTracks().find(function(t) {\n return t.id === transceiver.rtpReceiver.track.id;\n });\n if (nativeTrack) {\n removeTrackFromStreamAndFireEvent(nativeTrack, s);\n }\n });\n transceiver.associatedRemoteMediaStreams = [];\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.remoteCapabilities = remoteCapabilities;\n transceiver.rtpReceiver = rtpReceiver;\n transceiver.rtcpParameters = rtcpParameters;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n transceiver.recvEncodingParameters = recvEncodingParameters;\n\n // Start the RTCRtpReceiver now. The RTPSender is started in\n // setLocalDescription.\n pc._transceive(pc.transceivers[sdpMLineIndex],\n false,\n isNewTrack);\n } else if (description.type === 'answer' && !rejected) {\n transceiver = pc.transceivers[sdpMLineIndex];\n iceGatherer = transceiver.iceGatherer;\n iceTransport = transceiver.iceTransport;\n dtlsTransport = transceiver.dtlsTransport;\n rtpReceiver = transceiver.rtpReceiver;\n sendEncodingParameters = transceiver.sendEncodingParameters;\n localCapabilities = transceiver.localCapabilities;\n\n pc.transceivers[sdpMLineIndex].recvEncodingParameters =\n recvEncodingParameters;\n pc.transceivers[sdpMLineIndex].remoteCapabilities =\n remoteCapabilities;\n pc.transceivers[sdpMLineIndex].rtcpParameters = rtcpParameters;\n\n if (cands.length && iceTransport.state === 'new') {\n if ((isIceLite || isComplete) &&\n (!usingBundle || sdpMLineIndex === 0)) {\n iceTransport.setRemoteCandidates(cands);\n } else {\n cands.forEach(function(candidate) {\n maybeAddCandidate(transceiver.iceTransport, candidate);\n });\n }\n }\n\n if (!usingBundle || sdpMLineIndex === 0) {\n if (iceTransport.state === 'new') {\n iceTransport.start(iceGatherer, remoteIceParameters,\n 'controlling');\n }\n if (dtlsTransport.state === 'new') {\n dtlsTransport.start(remoteDtlsParameters);\n }\n }\n\n // If the offer contained RTX but the answer did not,\n // remove RTX from sendEncodingParameters.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n pc._transceive(transceiver,\n direction === 'sendrecv' || direction === 'recvonly',\n direction === 'sendrecv' || direction === 'sendonly');\n\n // TODO: rewrite to use http://w3c.github.io/webrtc-pc/#set-associated-remote-streams\n if (rtpReceiver &&\n (direction === 'sendrecv' || direction === 'sendonly')) {\n track = rtpReceiver.track;\n if (remoteMsid) {\n if (!streams[remoteMsid.stream]) {\n streams[remoteMsid.stream] = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams[remoteMsid.stream]);\n receiverList.push([track, rtpReceiver, streams[remoteMsid.stream]]);\n } else {\n if (!streams.default) {\n streams.default = new window.MediaStream();\n }\n addTrackToStreamAndFireEvent(track, streams.default);\n receiverList.push([track, rtpReceiver, streams.default]);\n }\n } else {\n // FIXME: actually the receiver should be created later.\n delete transceiver.rtpReceiver;\n }\n }\n });\n\n if (pc._dtlsRole === undefined) {\n pc._dtlsRole = description.type === 'offer' ? 'active' : 'passive';\n }\n\n pc._remoteDescription = {\n type: description.type,\n sdp: description.sdp\n };\n if (description.type === 'offer') {\n pc._updateSignalingState('have-remote-offer');\n } else {\n pc._updateSignalingState('stable');\n }\n Object.keys(streams).forEach(function(sid) {\n var stream = streams[sid];\n if (stream.getTracks().length) {\n if (pc.remoteStreams.indexOf(stream) === -1) {\n pc.remoteStreams.push(stream);\n var event = new Event('addstream');\n event.stream = stream;\n window.setTimeout(function() {\n pc._dispatchEvent('addstream', event);\n });\n }\n\n receiverList.forEach(function(item) {\n var track = item[0];\n var receiver = item[1];\n if (stream.id !== item[2].id) {\n return;\n }\n fireAddTrack(pc, track, receiver, [stream]);\n });\n }\n });\n receiverList.forEach(function(item) {\n if (item[2]) {\n return;\n }\n fireAddTrack(pc, item[0], item[1], []);\n });\n\n // check whether addIceCandidate({}) was called within four seconds after\n // setRemoteDescription.\n window.setTimeout(function() {\n if (!(pc && pc.transceivers)) {\n return;\n }\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport &&\n transceiver.iceTransport.state === 'new' &&\n transceiver.iceTransport.getRemoteCandidates().length > 0) {\n console.warn('Timeout for addRemoteCandidate. Consider sending ' +\n 'an end-of-candidates notification');\n transceiver.iceTransport.addRemoteCandidate({});\n }\n });\n }, 4000);\n\n return Promise.resolve();\n };\n\n RTCPeerConnection.prototype.close = function() {\n this.transceivers.forEach(function(transceiver) {\n /* not yet\n if (transceiver.iceGatherer) {\n transceiver.iceGatherer.close();\n }\n */\n if (transceiver.iceTransport) {\n transceiver.iceTransport.stop();\n }\n if (transceiver.dtlsTransport) {\n transceiver.dtlsTransport.stop();\n }\n if (transceiver.rtpSender) {\n transceiver.rtpSender.stop();\n }\n if (transceiver.rtpReceiver) {\n transceiver.rtpReceiver.stop();\n }\n });\n // FIXME: clean up tracks, local streams, remote streams, etc\n this._isClosed = true;\n this._updateSignalingState('closed');\n };\n\n // Update the signaling state.\n RTCPeerConnection.prototype._updateSignalingState = function(newState) {\n this.signalingState = newState;\n var event = new Event('signalingstatechange');\n this._dispatchEvent('signalingstatechange', event);\n };\n\n // Determine whether to fire the negotiationneeded event.\n RTCPeerConnection.prototype._maybeFireNegotiationNeeded = function() {\n var pc = this;\n if (this.signalingState !== 'stable' || this.needNegotiation === true) {\n return;\n }\n this.needNegotiation = true;\n window.setTimeout(function() {\n if (pc.needNegotiation) {\n pc.needNegotiation = false;\n var event = new Event('negotiationneeded');\n pc._dispatchEvent('negotiationneeded', event);\n }\n }, 0);\n };\n\n // Update the ice connection state.\n RTCPeerConnection.prototype._updateIceConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n checking: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n }\n });\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.checking > 0) {\n newState = 'checking';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n } else if (states.completed > 0) {\n newState = 'completed';\n }\n\n if (newState !== this.iceConnectionState) {\n this.iceConnectionState = newState;\n var event = new Event('iceconnectionstatechange');\n this._dispatchEvent('iceconnectionstatechange', event);\n }\n };\n\n // Update the connection state.\n RTCPeerConnection.prototype._updateConnectionState = function() {\n var newState;\n var states = {\n 'new': 0,\n closed: 0,\n connecting: 0,\n connected: 0,\n completed: 0,\n disconnected: 0,\n failed: 0\n };\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.iceTransport && transceiver.dtlsTransport &&\n !transceiver.rejected) {\n states[transceiver.iceTransport.state]++;\n states[transceiver.dtlsTransport.state]++;\n }\n });\n // ICETransport.completed and connected are the same for this purpose.\n states.connected += states.completed;\n\n newState = 'new';\n if (states.failed > 0) {\n newState = 'failed';\n } else if (states.connecting > 0) {\n newState = 'connecting';\n } else if (states.disconnected > 0) {\n newState = 'disconnected';\n } else if (states.new > 0) {\n newState = 'new';\n } else if (states.connected > 0) {\n newState = 'connected';\n }\n\n if (newState !== this.connectionState) {\n this.connectionState = newState;\n var event = new Event('connectionstatechange');\n this._dispatchEvent('connectionstatechange', event);\n }\n };\n\n RTCPeerConnection.prototype.createOffer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createOffer after close'));\n }\n\n var numAudioTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'audio';\n }).length;\n var numVideoTracks = pc.transceivers.filter(function(t) {\n return t.kind === 'video';\n }).length;\n\n // Determine number of audio and video tracks we need to send/recv.\n var offerOptions = arguments[0];\n if (offerOptions) {\n // Reject Chrome legacy constraints.\n if (offerOptions.mandatory || offerOptions.optional) {\n throw new TypeError(\n 'Legacy mandatory/optional constraints not supported.');\n }\n if (offerOptions.offerToReceiveAudio !== undefined) {\n if (offerOptions.offerToReceiveAudio === true) {\n numAudioTracks = 1;\n } else if (offerOptions.offerToReceiveAudio === false) {\n numAudioTracks = 0;\n } else {\n numAudioTracks = offerOptions.offerToReceiveAudio;\n }\n }\n if (offerOptions.offerToReceiveVideo !== undefined) {\n if (offerOptions.offerToReceiveVideo === true) {\n numVideoTracks = 1;\n } else if (offerOptions.offerToReceiveVideo === false) {\n numVideoTracks = 0;\n } else {\n numVideoTracks = offerOptions.offerToReceiveVideo;\n }\n }\n }\n\n pc.transceivers.forEach(function(transceiver) {\n if (transceiver.kind === 'audio') {\n numAudioTracks--;\n if (numAudioTracks < 0) {\n transceiver.wantReceive = false;\n }\n } else if (transceiver.kind === 'video') {\n numVideoTracks--;\n if (numVideoTracks < 0) {\n transceiver.wantReceive = false;\n }\n }\n });\n\n // Create M-lines for recvonly streams.\n while (numAudioTracks > 0 || numVideoTracks > 0) {\n if (numAudioTracks > 0) {\n pc._createTransceiver('audio');\n numAudioTracks--;\n }\n if (numVideoTracks > 0) {\n pc._createTransceiver('video');\n numVideoTracks--;\n }\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n // For each track, create an ice gatherer, ice transport,\n // dtls transport, potentially rtpsender and rtpreceiver.\n var track = transceiver.track;\n var kind = transceiver.kind;\n var mid = transceiver.mid || SDPUtils.generateIdentifier();\n transceiver.mid = mid;\n\n if (!transceiver.iceGatherer) {\n transceiver.iceGatherer = pc._createIceGatherer(sdpMLineIndex,\n pc.usingBundle);\n }\n\n var localCapabilities = window.RTCRtpSender.getCapabilities(kind);\n // filter RTX until additional stuff needed for RTX is implemented\n // in adapter.js\n if (edgeVersion < 15019) {\n localCapabilities.codecs = localCapabilities.codecs.filter(\n function(codec) {\n return codec.name !== 'rtx';\n });\n }\n localCapabilities.codecs.forEach(function(codec) {\n // work around https://bugs.chromium.org/p/webrtc/issues/detail?id=6552\n // by adding level-asymmetry-allowed=1\n if (codec.name === 'H264' &&\n codec.parameters['level-asymmetry-allowed'] === undefined) {\n codec.parameters['level-asymmetry-allowed'] = '1';\n }\n\n // for subsequent offers, we might have to re-use the payload\n // type of the last offer.\n if (transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.codecs) {\n transceiver.remoteCapabilities.codecs.forEach(function(remoteCodec) {\n if (codec.name.toLowerCase() === remoteCodec.name.toLowerCase() &&\n codec.clockRate === remoteCodec.clockRate) {\n codec.preferredPayloadType = remoteCodec.payloadType;\n }\n });\n }\n });\n localCapabilities.headerExtensions.forEach(function(hdrExt) {\n var remoteExtensions = transceiver.remoteCapabilities &&\n transceiver.remoteCapabilities.headerExtensions || [];\n remoteExtensions.forEach(function(rHdrExt) {\n if (hdrExt.uri === rHdrExt.uri) {\n hdrExt.id = rHdrExt.id;\n }\n });\n });\n\n // generate an ssrc now, to be used later in rtpSender.send\n var sendEncodingParameters = transceiver.sendEncodingParameters || [{\n ssrc: (2 * sdpMLineIndex + 1) * 1001\n }];\n if (track) {\n // add RTX\n if (edgeVersion >= 15019 && kind === 'video' &&\n !sendEncodingParameters[0].rtx) {\n sendEncodingParameters[0].rtx = {\n ssrc: sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n\n if (transceiver.wantReceive) {\n transceiver.rtpReceiver = new window.RTCRtpReceiver(\n transceiver.dtlsTransport, kind);\n }\n\n transceiver.localCapabilities = localCapabilities;\n transceiver.sendEncodingParameters = sendEncodingParameters;\n });\n\n // always offer BUNDLE and dispose on return if not supported.\n if (pc._config.bundlePolicy !== 'max-compat') {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n sdp += writeMediaSection(transceiver, transceiver.localCapabilities,\n 'offer', transceiver.stream, pc._dtlsRole);\n sdp += 'a=rtcp-rsize\\r\\n';\n\n if (transceiver.iceGatherer && pc.iceGatheringState !== 'new' &&\n (sdpMLineIndex === 0 || !pc.usingBundle)) {\n transceiver.iceGatherer.getLocalCandidates().forEach(function(cand) {\n cand.component = 1;\n sdp += 'a=' + SDPUtils.writeCandidate(cand) + '\\r\\n';\n });\n\n if (transceiver.iceGatherer.state === 'completed') {\n sdp += 'a=end-of-candidates\\r\\n';\n }\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'offer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.createAnswer = function() {\n var pc = this;\n\n if (pc._isClosed) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer after close'));\n }\n\n if (!(pc.signalingState === 'have-remote-offer' ||\n pc.signalingState === 'have-local-pranswer')) {\n return Promise.reject(makeError('InvalidStateError',\n 'Can not call createAnswer in signalingState ' + pc.signalingState));\n }\n\n var sdp = SDPUtils.writeSessionBoilerplate(pc._sdpSessionId,\n pc._sdpSessionVersion++);\n if (pc.usingBundle) {\n sdp += 'a=group:BUNDLE ' + pc.transceivers.map(function(t) {\n return t.mid;\n }).join(' ') + '\\r\\n';\n }\n sdp += 'a=ice-options:trickle\\r\\n';\n\n var mediaSectionsInOffer = SDPUtils.getMediaSections(\n pc._remoteDescription.sdp).length;\n pc.transceivers.forEach(function(transceiver, sdpMLineIndex) {\n if (sdpMLineIndex + 1 > mediaSectionsInOffer) {\n return;\n }\n if (transceiver.rejected) {\n if (transceiver.kind === 'application') {\n if (transceiver.protocol === 'DTLS/SCTP') { // legacy fmt\n sdp += 'm=application 0 DTLS/SCTP 5000\\r\\n';\n } else {\n sdp += 'm=application 0 ' + transceiver.protocol +\n ' webrtc-datachannel\\r\\n';\n }\n } else if (transceiver.kind === 'audio') {\n sdp += 'm=audio 0 UDP/TLS/RTP/SAVPF 0\\r\\n' +\n 'a=rtpmap:0 PCMU/8000\\r\\n';\n } else if (transceiver.kind === 'video') {\n sdp += 'm=video 0 UDP/TLS/RTP/SAVPF 120\\r\\n' +\n 'a=rtpmap:120 VP8/90000\\r\\n';\n }\n sdp += 'c=IN IP4 0.0.0.0\\r\\n' +\n 'a=inactive\\r\\n' +\n 'a=mid:' + transceiver.mid + '\\r\\n';\n return;\n }\n\n // FIXME: look at direction.\n if (transceiver.stream) {\n var localTrack;\n if (transceiver.kind === 'audio') {\n localTrack = transceiver.stream.getAudioTracks()[0];\n } else if (transceiver.kind === 'video') {\n localTrack = transceiver.stream.getVideoTracks()[0];\n }\n if (localTrack) {\n // add RTX\n if (edgeVersion >= 15019 && transceiver.kind === 'video' &&\n !transceiver.sendEncodingParameters[0].rtx) {\n transceiver.sendEncodingParameters[0].rtx = {\n ssrc: transceiver.sendEncodingParameters[0].ssrc + 1\n };\n }\n }\n }\n\n // Calculate intersection of capabilities.\n var commonCapabilities = getCommonCapabilities(\n transceiver.localCapabilities,\n transceiver.remoteCapabilities);\n\n var hasRtx = commonCapabilities.codecs.filter(function(c) {\n return c.name.toLowerCase() === 'rtx';\n }).length;\n if (!hasRtx && transceiver.sendEncodingParameters[0].rtx) {\n delete transceiver.sendEncodingParameters[0].rtx;\n }\n\n sdp += writeMediaSection(transceiver, commonCapabilities,\n 'answer', transceiver.stream, pc._dtlsRole);\n if (transceiver.rtcpParameters &&\n transceiver.rtcpParameters.reducedSize) {\n sdp += 'a=rtcp-rsize\\r\\n';\n }\n });\n\n var desc = new window.RTCSessionDescription({\n type: 'answer',\n sdp: sdp\n });\n return Promise.resolve(desc);\n };\n\n RTCPeerConnection.prototype.addIceCandidate = function(candidate) {\n var pc = this;\n var sections;\n if (candidate && !(candidate.sdpMLineIndex !== undefined ||\n candidate.sdpMid)) {\n return Promise.reject(new TypeError('sdpMLineIndex or sdpMid required'));\n }\n\n // TODO: needs to go into ops queue.\n return new Promise(function(resolve, reject) {\n if (!pc._remoteDescription) {\n return reject(makeError('InvalidStateError',\n 'Can not add ICE candidate without a remote description'));\n } else if (!candidate || candidate.candidate === '') {\n for (var j = 0; j < pc.transceivers.length; j++) {\n if (pc.transceivers[j].rejected) {\n continue;\n }\n pc.transceivers[j].iceTransport.addRemoteCandidate({});\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[j] += 'a=end-of-candidates\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n if (pc.usingBundle) {\n break;\n }\n }\n } else {\n var sdpMLineIndex = candidate.sdpMLineIndex;\n if (candidate.sdpMid) {\n for (var i = 0; i < pc.transceivers.length; i++) {\n if (pc.transceivers[i].mid === candidate.sdpMid) {\n sdpMLineIndex = i;\n break;\n }\n }\n }\n var transceiver = pc.transceivers[sdpMLineIndex];\n if (transceiver) {\n if (transceiver.rejected) {\n return resolve();\n }\n var cand = Object.keys(candidate.candidate).length > 0 ?\n SDPUtils.parseCandidate(candidate.candidate) : {};\n // Ignore Chrome's invalid candidates since Edge does not like them.\n if (cand.protocol === 'tcp' && (cand.port === 0 || cand.port === 9)) {\n return resolve();\n }\n // Ignore RTCP candidates, we assume RTCP-MUX.\n if (cand.component && cand.component !== 1) {\n return resolve();\n }\n // when using bundle, avoid adding candidates to the wrong\n // ice transport. And avoid adding candidates added in the SDP.\n if (sdpMLineIndex === 0 || (sdpMLineIndex > 0 &&\n transceiver.iceTransport !== pc.transceivers[0].iceTransport)) {\n if (!maybeAddCandidate(transceiver.iceTransport, cand)) {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n\n // update the remoteDescription.\n var candidateString = candidate.candidate.trim();\n if (candidateString.indexOf('a=') === 0) {\n candidateString = candidateString.substr(2);\n }\n sections = SDPUtils.getMediaSections(pc._remoteDescription.sdp);\n sections[sdpMLineIndex] += 'a=' +\n (cand.type ? candidateString : 'end-of-candidates')\n + '\\r\\n';\n pc._remoteDescription.sdp =\n SDPUtils.getDescription(pc._remoteDescription.sdp) +\n sections.join('');\n } else {\n return reject(makeError('OperationError',\n 'Can not add ICE candidate'));\n }\n }\n resolve();\n });\n };\n\n RTCPeerConnection.prototype.getStats = function(selector) {\n if (selector && selector instanceof window.MediaStreamTrack) {\n var senderOrReceiver = null;\n this.transceivers.forEach(function(transceiver) {\n if (transceiver.rtpSender &&\n transceiver.rtpSender.track === selector) {\n senderOrReceiver = transceiver.rtpSender;\n } else if (transceiver.rtpReceiver &&\n transceiver.rtpReceiver.track === selector) {\n senderOrReceiver = transceiver.rtpReceiver;\n }\n });\n if (!senderOrReceiver) {\n throw makeError('InvalidAccessError', 'Invalid selector.');\n }\n return senderOrReceiver.getStats();\n }\n\n var promises = [];\n this.transceivers.forEach(function(transceiver) {\n ['rtpSender', 'rtpReceiver', 'iceGatherer', 'iceTransport',\n 'dtlsTransport'].forEach(function(method) {\n if (transceiver[method]) {\n promises.push(transceiver[method].getStats());\n }\n });\n });\n return Promise.all(promises).then(function(allStats) {\n var results = new Map();\n allStats.forEach(function(stats) {\n stats.forEach(function(stat) {\n results.set(stat.id, stat);\n });\n });\n return results;\n });\n };\n\n // fix low-level stat names and return Map instead of object.\n var ortcObjects = ['RTCRtpSender', 'RTCRtpReceiver', 'RTCIceGatherer',\n 'RTCIceTransport', 'RTCDtlsTransport'];\n ortcObjects.forEach(function(ortcObjectName) {\n var obj = window[ortcObjectName];\n if (obj && obj.prototype && obj.prototype.getStats) {\n var nativeGetstats = obj.prototype.getStats;\n obj.prototype.getStats = function() {\n return nativeGetstats.apply(this)\n .then(function(nativeStats) {\n var mapStats = new Map();\n Object.keys(nativeStats).forEach(function(id) {\n nativeStats[id].type = fixStatsType(nativeStats[id]);\n mapStats.set(id, nativeStats[id]);\n });\n return mapStats;\n });\n };\n }\n });\n\n // legacy callback shims. Should be moved to adapter.js some days.\n var methods = ['createOffer', 'createAnswer'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[0] === 'function' ||\n typeof args[1] === 'function') { // legacy\n return nativeMethod.apply(this, [arguments[2]])\n .then(function(description) {\n if (typeof args[0] === 'function') {\n args[0].apply(null, [description]);\n }\n }, function(error) {\n if (typeof args[1] === 'function') {\n args[1].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n methods = ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate'];\n methods.forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function' ||\n typeof args[2] === 'function') { // legacy\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n }, function(error) {\n if (typeof args[2] === 'function') {\n args[2].apply(null, [error]);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n // getStats is special. It doesn't have a spec legacy method yet we support\n // getStats(something, cb) without error callbacks.\n ['getStats'].forEach(function(method) {\n var nativeMethod = RTCPeerConnection.prototype[method];\n RTCPeerConnection.prototype[method] = function() {\n var args = arguments;\n if (typeof args[1] === 'function') {\n return nativeMethod.apply(this, arguments)\n .then(function() {\n if (typeof args[1] === 'function') {\n args[1].apply(null);\n }\n });\n }\n return nativeMethod.apply(this, arguments);\n };\n });\n\n return RTCPeerConnection;\n};\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n const shimError_ = function(e) {\n return {\n name: {PermissionDeniedError: 'NotAllowedError'}[e.name] || e.name,\n message: e.message,\n constraint: e.constraint,\n toString() {\n return this.name;\n }\n };\n };\n\n // getUserMedia error shim.\n const origGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n return origGetUserMedia(c).catch(e => Promise.reject(shimError_(e)));\n };\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n /* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window) {\n if (!('getDisplayMedia' in window.navigator)) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n window.navigator.getDisplayMedia.bind(window.navigator);\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nimport {filterIceServers} from './filtericeservers';\nimport shimRTCPeerConnection from 'rtcpeerconnection-shim';\n\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimPeerConnection(window, browserDetails) {\n if (window.RTCIceGatherer) {\n if (!window.RTCIceCandidate) {\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n return args;\n };\n }\n if (!window.RTCSessionDescription) {\n window.RTCSessionDescription = function RTCSessionDescription(args) {\n return args;\n };\n }\n // this adds an additional event listener to MediaStrackTrack that signals\n // when a tracks enabled property was changed. Workaround for a bug in\n // addStream, see below. No longer required in 15025+\n if (browserDetails.version < 15025) {\n const origMSTEnabled = Object.getOwnPropertyDescriptor(\n window.MediaStreamTrack.prototype, 'enabled');\n Object.defineProperty(window.MediaStreamTrack.prototype, 'enabled', {\n set(value) {\n origMSTEnabled.set.call(this, value);\n const ev = new Event('enabled');\n ev.enabled = value;\n this.dispatchEvent(ev);\n }\n });\n }\n }\n\n // ORTC defines the DTMF sender a bit different.\n // https://github.com/w3c/ortc/issues/714\n if (window.RTCRtpSender && !('dtmf' in window.RTCRtpSender.prototype)) {\n Object.defineProperty(window.RTCRtpSender.prototype, 'dtmf', {\n get() {\n if (this._dtmf === undefined) {\n if (this.track.kind === 'audio') {\n this._dtmf = new window.RTCDtmfSender(this);\n } else if (this.track.kind === 'video') {\n this._dtmf = null;\n }\n }\n return this._dtmf;\n }\n });\n }\n // Edge currently only implements the RTCDtmfSender, not the\n // RTCDTMFSender alias. See http://draft.ortc.org/#rtcdtmfsender2*\n if (window.RTCDtmfSender && !window.RTCDTMFSender) {\n window.RTCDTMFSender = window.RTCDtmfSender;\n }\n\n const RTCPeerConnectionShim = shimRTCPeerConnection(window,\n browserDetails.version);\n window.RTCPeerConnection = function RTCPeerConnection(config) {\n if (config && config.iceServers) {\n config.iceServers = filterIceServers(config.iceServers,\n browserDetails.version);\n utils.log('ICE servers after filtering:', config.iceServers);\n }\n return new RTCPeerConnectionShim(config);\n };\n window.RTCPeerConnection.prototype = RTCPeerConnectionShim.prototype;\n}\n\nexport function shimReplaceTrack(window) {\n // ORTC has replaceTrack -- https://github.com/w3c/ortc/issues/614\n if (window.RTCRtpSender &&\n !('replaceTrack' in window.RTCRtpSender.prototype)) {\n window.RTCRtpSender.prototype.replaceTrack =\n window.RTCRtpSender.prototype.setTrack;\n }\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\n\nexport function shimGetUserMedia(window, browserDetails) {\n const navigator = window && window.navigator;\n const MediaStreamTrack = window && window.MediaStreamTrack;\n\n navigator.getUserMedia = function(constraints, onSuccess, onError) {\n // Replace Firefox 44+'s deprecation warning with unprefixed version.\n utils.deprecated('navigator.getUserMedia',\n 'navigator.mediaDevices.getUserMedia');\n navigator.mediaDevices.getUserMedia(constraints).then(onSuccess, onError);\n };\n\n if (!(browserDetails.version > 55 &&\n 'autoGainControl' in navigator.mediaDevices.getSupportedConstraints())) {\n const remap = function(obj, a, b) {\n if (a in obj && !(b in obj)) {\n obj[b] = obj[a];\n delete obj[a];\n }\n };\n\n const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.\n bind(navigator.mediaDevices);\n navigator.mediaDevices.getUserMedia = function(c) {\n if (typeof c === 'object' && typeof c.audio === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c.audio, 'autoGainControl', 'mozAutoGainControl');\n remap(c.audio, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeGetUserMedia(c);\n };\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.getSettings) {\n const nativeGetSettings = MediaStreamTrack.prototype.getSettings;\n MediaStreamTrack.prototype.getSettings = function() {\n const obj = nativeGetSettings.apply(this, arguments);\n remap(obj, 'mozAutoGainControl', 'autoGainControl');\n remap(obj, 'mozNoiseSuppression', 'noiseSuppression');\n return obj;\n };\n }\n\n if (MediaStreamTrack && MediaStreamTrack.prototype.applyConstraints) {\n const nativeApplyConstraints =\n MediaStreamTrack.prototype.applyConstraints;\n MediaStreamTrack.prototype.applyConstraints = function(c) {\n if (this.kind === 'audio' && typeof c === 'object') {\n c = JSON.parse(JSON.stringify(c));\n remap(c, 'autoGainControl', 'mozAutoGainControl');\n remap(c, 'noiseSuppression', 'mozNoiseSuppression');\n }\n return nativeApplyConstraints.apply(this, [c]);\n };\n }\n }\n}\n","/*\n * Copyright (c) 2018 The adapter.js project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nexport function shimGetDisplayMedia(window, preferredMediaSource) {\n if (window.navigator.mediaDevices &&\n 'getDisplayMedia' in window.navigator.mediaDevices) {\n return;\n }\n if (!(window.navigator.mediaDevices)) {\n return;\n }\n window.navigator.mediaDevices.getDisplayMedia =\n function getDisplayMedia(constraints) {\n if (!(constraints && constraints.video)) {\n const err = new DOMException('getDisplayMedia without video ' +\n 'constraints is undefined');\n err.name = 'NotFoundError';\n // from https://heycam.github.io/webidl/#idl-DOMException-error-names\n err.code = 8;\n return Promise.reject(err);\n }\n if (constraints.video === true) {\n constraints.video = {mediaSource: preferredMediaSource};\n } else {\n constraints.video.mediaSource = preferredMediaSource;\n }\n return window.navigator.mediaDevices.getUserMedia(constraints);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport * as utils from '../utils';\nexport {shimGetUserMedia} from './getusermedia';\nexport {shimGetDisplayMedia} from './getdisplaymedia';\n\nexport function shimOnTrack(window) {\n if (typeof window === 'object' && window.RTCTrackEvent &&\n ('receiver' in window.RTCTrackEvent.prototype) &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimPeerConnection(window, browserDetails) {\n if (typeof window !== 'object' ||\n !(window.RTCPeerConnection || window.mozRTCPeerConnection)) {\n return; // probably media.peerconnection.enabled=false in about:config\n }\n if (!window.RTCPeerConnection && window.mozRTCPeerConnection) {\n // very basic support for old versions.\n window.RTCPeerConnection = window.mozRTCPeerConnection;\n }\n\n if (browserDetails.version < 53) {\n // shim away need for obsolete RTCIceCandidate/RTCSessionDescription.\n ['setLocalDescription', 'setRemoteDescription', 'addIceCandidate']\n .forEach(function(method) {\n const nativeMethod = window.RTCPeerConnection.prototype[method];\n const methodObj = {[method]() {\n arguments[0] = new ((method === 'addIceCandidate') ?\n window.RTCIceCandidate :\n window.RTCSessionDescription)(arguments[0]);\n return nativeMethod.apply(this, arguments);\n }};\n window.RTCPeerConnection.prototype[method] = methodObj[method];\n });\n }\n\n const modernStatsTypes = {\n inboundrtp: 'inbound-rtp',\n outboundrtp: 'outbound-rtp',\n candidatepair: 'candidate-pair',\n localcandidate: 'local-candidate',\n remotecandidate: 'remote-candidate'\n };\n\n const nativeGetStats = window.RTCPeerConnection.prototype.getStats;\n window.RTCPeerConnection.prototype.getStats = function getStats() {\n const [selector, onSucc, onErr] = arguments;\n return nativeGetStats.apply(this, [selector || null])\n .then(stats => {\n if (browserDetails.version < 53 && !onSucc) {\n // Shim only promise getStats with spec-hyphens in type names\n // Leave callback version alone; misc old uses of forEach before Map\n try {\n stats.forEach(stat => {\n stat.type = modernStatsTypes[stat.type] || stat.type;\n });\n } catch (e) {\n if (e.name !== 'TypeError') {\n throw e;\n }\n // Avoid TypeError: \"type\" is read-only, in old versions. 34-43ish\n stats.forEach((stat, i) => {\n stats.set(i, Object.assign({}, stat, {\n type: modernStatsTypes[stat.type] || stat.type\n }));\n });\n }\n }\n return stats;\n })\n .then(onSucc, onErr);\n };\n}\n\nexport function shimSenderGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpSender.prototype) {\n return;\n }\n const origGetSenders = window.RTCPeerConnection.prototype.getSenders;\n if (origGetSenders) {\n window.RTCPeerConnection.prototype.getSenders = function getSenders() {\n const senders = origGetSenders.apply(this, []);\n senders.forEach(sender => sender._pc = this);\n return senders;\n };\n }\n\n const origAddTrack = window.RTCPeerConnection.prototype.addTrack;\n if (origAddTrack) {\n window.RTCPeerConnection.prototype.addTrack = function addTrack() {\n const sender = origAddTrack.apply(this, arguments);\n sender._pc = this;\n return sender;\n };\n }\n window.RTCRtpSender.prototype.getStats = function getStats() {\n return this.track ? this._pc.getStats(this.track) :\n Promise.resolve(new Map());\n };\n}\n\nexport function shimReceiverGetStats(window) {\n if (!(typeof window === 'object' && window.RTCPeerConnection &&\n window.RTCRtpSender)) {\n return;\n }\n if (window.RTCRtpSender && 'getStats' in window.RTCRtpReceiver.prototype) {\n return;\n }\n const origGetReceivers = window.RTCPeerConnection.prototype.getReceivers;\n if (origGetReceivers) {\n window.RTCPeerConnection.prototype.getReceivers = function getReceivers() {\n const receivers = origGetReceivers.apply(this, []);\n receivers.forEach(receiver => receiver._pc = this);\n return receivers;\n };\n }\n utils.wrapPeerConnectionEvent(window, 'track', e => {\n e.receiver._pc = e.srcElement;\n return e;\n });\n window.RTCRtpReceiver.prototype.getStats = function getStats() {\n return this._pc.getStats(this.track);\n };\n}\n\nexport function shimRemoveStream(window) {\n if (!window.RTCPeerConnection ||\n 'removeStream' in window.RTCPeerConnection.prototype) {\n return;\n }\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n utils.deprecated('removeStream', 'removeTrack');\n this.getSenders().forEach(sender => {\n if (sender.track && stream.getTracks().includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n}\n\nexport function shimRTCDataChannel(window) {\n // rename DataChannel to RTCDataChannel (native fix in FF60):\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1173851\n if (window.DataChannel && !window.RTCDataChannel) {\n window.RTCDataChannel = window.DataChannel;\n }\n}\n\nexport function shimAddTransceiver(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origAddTransceiver = window.RTCPeerConnection.prototype.addTransceiver;\n if (origAddTransceiver) {\n window.RTCPeerConnection.prototype.addTransceiver =\n function addTransceiver() {\n this.setParametersPromises = [];\n const initParameters = arguments[1];\n const shouldPerformCheck = initParameters &&\n 'sendEncodings' in initParameters;\n if (shouldPerformCheck) {\n // If sendEncodings params are provided, validate grammar\n initParameters.sendEncodings.forEach((encodingParam) => {\n if ('rid' in encodingParam) {\n const ridRegex = /^[a-z0-9]{0,16}$/i;\n if (!ridRegex.test(encodingParam.rid)) {\n throw new TypeError('Invalid RID value provided.');\n }\n }\n if ('scaleResolutionDownBy' in encodingParam) {\n if (!(parseFloat(encodingParam.scaleResolutionDownBy) >= 1.0)) {\n throw new RangeError('scale_resolution_down_by must be >= 1.0');\n }\n }\n if ('maxFramerate' in encodingParam) {\n if (!(parseFloat(encodingParam.maxFramerate) >= 0)) {\n throw new RangeError('max_framerate must be >= 0.0');\n }\n }\n });\n }\n const transceiver = origAddTransceiver.apply(this, arguments);\n if (shouldPerformCheck) {\n // Check if the init options were applied. If not we do this in an\n // asynchronous way and save the promise reference in a global object.\n // This is an ugly hack, but at the same time is way more robust than\n // checking the sender parameters before and after the createOffer\n // Also note that after the createoffer we are not 100% sure that\n // the params were asynchronously applied so we might miss the\n // opportunity to recreate offer.\n const {sender} = transceiver;\n const params = sender.getParameters();\n if (!('encodings' in params) ||\n // Avoid being fooled by patched getParameters() below.\n (params.encodings.length === 1 &&\n Object.keys(params.encodings[0]).length === 0)) {\n params.encodings = initParameters.sendEncodings;\n sender.sendEncodings = initParameters.sendEncodings;\n this.setParametersPromises.push(sender.setParameters(params)\n .then(() => {\n delete sender.sendEncodings;\n }).catch(() => {\n delete sender.sendEncodings;\n })\n );\n }\n }\n return transceiver;\n };\n }\n}\n\nexport function shimGetParameters(window) {\n if (!(typeof window === 'object' && window.RTCRtpSender)) {\n return;\n }\n const origGetParameters = window.RTCRtpSender.prototype.getParameters;\n if (origGetParameters) {\n window.RTCRtpSender.prototype.getParameters =\n function getParameters() {\n const params = origGetParameters.apply(this, arguments);\n if (!('encodings' in params)) {\n params.encodings = [].concat(this.sendEncodings || [{}]);\n }\n return params;\n };\n }\n}\n\nexport function shimCreateOffer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer = function createOffer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateOffer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimCreateAnswer(window) {\n // https://github.com/webrtcHacks/adapter/issues/998#issuecomment-516921647\n // Firefox ignores the init sendEncodings options passed to addTransceiver\n // https://bugzilla.mozilla.org/show_bug.cgi?id=1396918\n if (!(typeof window === 'object' && window.RTCPeerConnection)) {\n return;\n }\n const origCreateAnswer = window.RTCPeerConnection.prototype.createAnswer;\n window.RTCPeerConnection.prototype.createAnswer = function createAnswer() {\n if (this.setParametersPromises && this.setParametersPromises.length) {\n return Promise.all(this.setParametersPromises)\n .then(() => {\n return origCreateAnswer.apply(this, arguments);\n })\n .finally(() => {\n this.setParametersPromises = [];\n });\n }\n return origCreateAnswer.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n'use strict';\nimport * as utils from '../utils';\n\nexport function shimLocalStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getLocalStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getLocalStreams =\n function getLocalStreams() {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n return this._localStreams;\n };\n }\n if (!('addStream' in window.RTCPeerConnection.prototype)) {\n const _addTrack = window.RTCPeerConnection.prototype.addTrack;\n window.RTCPeerConnection.prototype.addStream = function addStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n // Try to emulate Chrome's behaviour of adding in audio-video order.\n // Safari orders by track id.\n stream.getAudioTracks().forEach(track => _addTrack.call(this, track,\n stream));\n stream.getVideoTracks().forEach(track => _addTrack.call(this, track,\n stream));\n };\n\n window.RTCPeerConnection.prototype.addTrack =\n function addTrack(track, ...streams) {\n if (streams) {\n streams.forEach((stream) => {\n if (!this._localStreams) {\n this._localStreams = [stream];\n } else if (!this._localStreams.includes(stream)) {\n this._localStreams.push(stream);\n }\n });\n }\n return _addTrack.apply(this, arguments);\n };\n }\n if (!('removeStream' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.removeStream =\n function removeStream(stream) {\n if (!this._localStreams) {\n this._localStreams = [];\n }\n const index = this._localStreams.indexOf(stream);\n if (index === -1) {\n return;\n }\n this._localStreams.splice(index, 1);\n const tracks = stream.getTracks();\n this.getSenders().forEach(sender => {\n if (tracks.includes(sender.track)) {\n this.removeTrack(sender);\n }\n });\n };\n }\n}\n\nexport function shimRemoteStreamsAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n if (!('getRemoteStreams' in window.RTCPeerConnection.prototype)) {\n window.RTCPeerConnection.prototype.getRemoteStreams =\n function getRemoteStreams() {\n return this._remoteStreams ? this._remoteStreams : [];\n };\n }\n if (!('onaddstream' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'onaddstream', {\n get() {\n return this._onaddstream;\n },\n set(f) {\n if (this._onaddstream) {\n this.removeEventListener('addstream', this._onaddstream);\n this.removeEventListener('track', this._onaddstreampoly);\n }\n this.addEventListener('addstream', this._onaddstream = f);\n this.addEventListener('track', this._onaddstreampoly = (e) => {\n e.streams.forEach(stream => {\n if (!this._remoteStreams) {\n this._remoteStreams = [];\n }\n if (this._remoteStreams.includes(stream)) {\n return;\n }\n this._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n this.dispatchEvent(event);\n });\n });\n }\n });\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n const pc = this;\n if (!this._onaddstreampoly) {\n this.addEventListener('track', this._onaddstreampoly = function(e) {\n e.streams.forEach(stream => {\n if (!pc._remoteStreams) {\n pc._remoteStreams = [];\n }\n if (pc._remoteStreams.indexOf(stream) >= 0) {\n return;\n }\n pc._remoteStreams.push(stream);\n const event = new Event('addstream');\n event.stream = stream;\n pc.dispatchEvent(event);\n });\n });\n }\n return origSetRemoteDescription.apply(pc, arguments);\n };\n }\n}\n\nexport function shimCallbacksAPI(window) {\n if (typeof window !== 'object' || !window.RTCPeerConnection) {\n return;\n }\n const prototype = window.RTCPeerConnection.prototype;\n const origCreateOffer = prototype.createOffer;\n const origCreateAnswer = prototype.createAnswer;\n const setLocalDescription = prototype.setLocalDescription;\n const setRemoteDescription = prototype.setRemoteDescription;\n const addIceCandidate = prototype.addIceCandidate;\n\n prototype.createOffer =\n function createOffer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateOffer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n prototype.createAnswer =\n function createAnswer(successCallback, failureCallback) {\n const options = (arguments.length >= 2) ? arguments[2] : arguments[0];\n const promise = origCreateAnswer.apply(this, [options]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n\n let withCallback = function(description, successCallback, failureCallback) {\n const promise = setLocalDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setLocalDescription = withCallback;\n\n withCallback = function(description, successCallback, failureCallback) {\n const promise = setRemoteDescription.apply(this, [description]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.setRemoteDescription = withCallback;\n\n withCallback = function(candidate, successCallback, failureCallback) {\n const promise = addIceCandidate.apply(this, [candidate]);\n if (!failureCallback) {\n return promise;\n }\n promise.then(successCallback, failureCallback);\n return Promise.resolve();\n };\n prototype.addIceCandidate = withCallback;\n}\n\nexport function shimGetUserMedia(window) {\n const navigator = window && window.navigator;\n\n if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {\n // shim not needed in Safari 12.1\n const mediaDevices = navigator.mediaDevices;\n const _getUserMedia = mediaDevices.getUserMedia.bind(mediaDevices);\n navigator.mediaDevices.getUserMedia = (constraints) => {\n return _getUserMedia(shimConstraints(constraints));\n };\n }\n\n if (!navigator.getUserMedia && navigator.mediaDevices &&\n navigator.mediaDevices.getUserMedia) {\n navigator.getUserMedia = function getUserMedia(constraints, cb, errcb) {\n navigator.mediaDevices.getUserMedia(constraints)\n .then(cb, errcb);\n }.bind(navigator);\n }\n}\n\nexport function shimConstraints(constraints) {\n if (constraints && constraints.video !== undefined) {\n return Object.assign({},\n constraints,\n {video: utils.compactObject(constraints.video)}\n );\n }\n\n return constraints;\n}\n\nexport function shimRTCIceServerUrls(window) {\n if (!window.RTCPeerConnection) {\n return;\n }\n // migrate from non-spec RTCIceServer.url to RTCIceServer.urls\n const OrigPeerConnection = window.RTCPeerConnection;\n window.RTCPeerConnection =\n function RTCPeerConnection(pcConfig, pcConstraints) {\n if (pcConfig && pcConfig.iceServers) {\n const newIceServers = [];\n for (let i = 0; i < pcConfig.iceServers.length; i++) {\n let server = pcConfig.iceServers[i];\n if (!server.hasOwnProperty('urls') &&\n server.hasOwnProperty('url')) {\n utils.deprecated('RTCIceServer.url', 'RTCIceServer.urls');\n server = JSON.parse(JSON.stringify(server));\n server.urls = server.url;\n delete server.url;\n newIceServers.push(server);\n } else {\n newIceServers.push(pcConfig.iceServers[i]);\n }\n }\n pcConfig.iceServers = newIceServers;\n }\n return new OrigPeerConnection(pcConfig, pcConstraints);\n };\n window.RTCPeerConnection.prototype = OrigPeerConnection.prototype;\n // wrap static methods. Currently just generateCertificate.\n if ('generateCertificate' in OrigPeerConnection) {\n Object.defineProperty(window.RTCPeerConnection, 'generateCertificate', {\n get() {\n return OrigPeerConnection.generateCertificate;\n }\n });\n }\n}\n\nexport function shimTrackEventTransceiver(window) {\n // Add event.transceiver member over deprecated event.receiver\n if (typeof window === 'object' && window.RTCTrackEvent &&\n 'receiver' in window.RTCTrackEvent.prototype &&\n !('transceiver' in window.RTCTrackEvent.prototype)) {\n Object.defineProperty(window.RTCTrackEvent.prototype, 'transceiver', {\n get() {\n return {receiver: this.receiver};\n }\n });\n }\n}\n\nexport function shimCreateOfferLegacy(window) {\n const origCreateOffer = window.RTCPeerConnection.prototype.createOffer;\n window.RTCPeerConnection.prototype.createOffer =\n function createOffer(offerOptions) {\n if (offerOptions) {\n if (typeof offerOptions.offerToReceiveAudio !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveAudio =\n !!offerOptions.offerToReceiveAudio;\n }\n const audioTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'audio');\n if (offerOptions.offerToReceiveAudio === false && audioTransceiver) {\n if (audioTransceiver.direction === 'sendrecv') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('sendonly');\n } else {\n audioTransceiver.direction = 'sendonly';\n }\n } else if (audioTransceiver.direction === 'recvonly') {\n if (audioTransceiver.setDirection) {\n audioTransceiver.setDirection('inactive');\n } else {\n audioTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveAudio === true &&\n !audioTransceiver) {\n this.addTransceiver('audio');\n }\n\n if (typeof offerOptions.offerToReceiveVideo !== 'undefined') {\n // support bit values\n offerOptions.offerToReceiveVideo =\n !!offerOptions.offerToReceiveVideo;\n }\n const videoTransceiver = this.getTransceivers().find(transceiver =>\n transceiver.receiver.track.kind === 'video');\n if (offerOptions.offerToReceiveVideo === false && videoTransceiver) {\n if (videoTransceiver.direction === 'sendrecv') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('sendonly');\n } else {\n videoTransceiver.direction = 'sendonly';\n }\n } else if (videoTransceiver.direction === 'recvonly') {\n if (videoTransceiver.setDirection) {\n videoTransceiver.setDirection('inactive');\n } else {\n videoTransceiver.direction = 'inactive';\n }\n }\n } else if (offerOptions.offerToReceiveVideo === true &&\n !videoTransceiver) {\n this.addTransceiver('video');\n }\n }\n return origCreateOffer.apply(this, arguments);\n };\n}\n\nexport function shimAudioContext(window) {\n if (typeof window !== 'object' || window.AudioContext) {\n return;\n }\n window.AudioContext = window.webkitAudioContext;\n}\n","/*\n * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n'use strict';\n\nimport SDPUtils from 'sdp';\nimport * as utils from './utils';\n\nexport function shimRTCIceCandidate(window) {\n // foundation is arbitrarily chosen as an indicator for full support for\n // https://w3c.github.io/webrtc-pc/#rtcicecandidate-interface\n if (!window.RTCIceCandidate || (window.RTCIceCandidate && 'foundation' in\n window.RTCIceCandidate.prototype)) {\n return;\n }\n\n const NativeRTCIceCandidate = window.RTCIceCandidate;\n window.RTCIceCandidate = function RTCIceCandidate(args) {\n // Remove the a= which shouldn't be part of the candidate string.\n if (typeof args === 'object' && args.candidate &&\n args.candidate.indexOf('a=') === 0) {\n args = JSON.parse(JSON.stringify(args));\n args.candidate = args.candidate.substr(2);\n }\n\n if (args.candidate && args.candidate.length) {\n // Augment the native candidate with the parsed fields.\n const nativeCandidate = new NativeRTCIceCandidate(args);\n const parsedCandidate = SDPUtils.parseCandidate(args.candidate);\n const augmentedCandidate = Object.assign(nativeCandidate,\n parsedCandidate);\n\n // Add a serializer that does not serialize the extra attributes.\n augmentedCandidate.toJSON = function toJSON() {\n return {\n candidate: augmentedCandidate.candidate,\n sdpMid: augmentedCandidate.sdpMid,\n sdpMLineIndex: augmentedCandidate.sdpMLineIndex,\n usernameFragment: augmentedCandidate.usernameFragment,\n };\n };\n return augmentedCandidate;\n }\n return new NativeRTCIceCandidate(args);\n };\n window.RTCIceCandidate.prototype = NativeRTCIceCandidate.prototype;\n\n // Hook up the augmented candidate in onicecandidate and\n // addEventListener('icecandidate', ...)\n utils.wrapPeerConnectionEvent(window, 'icecandidate', e => {\n if (e.candidate) {\n Object.defineProperty(e, 'candidate', {\n value: new window.RTCIceCandidate(e.candidate),\n writable: 'false'\n });\n }\n return e;\n });\n}\n\nexport function shimMaxMessageSize(window, browserDetails) {\n if (!window.RTCPeerConnection) {\n return;\n }\n\n if (!('sctp' in window.RTCPeerConnection.prototype)) {\n Object.defineProperty(window.RTCPeerConnection.prototype, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n }\n });\n }\n\n const sctpInDescription = function(description) {\n if (!description || !description.sdp) {\n return false;\n }\n const sections = SDPUtils.splitSections(description.sdp);\n sections.shift();\n return sections.some(mediaSection => {\n const mLine = SDPUtils.parseMLine(mediaSection);\n return mLine && mLine.kind === 'application'\n && mLine.protocol.indexOf('SCTP') !== -1;\n });\n };\n\n const getRemoteFirefoxVersion = function(description) {\n // TODO: Is there a better solution for detecting Firefox?\n const match = description.sdp.match(/mozilla...THIS_IS_SDPARTA-(\\d+)/);\n if (match === null || match.length < 2) {\n return -1;\n }\n const version = parseInt(match[1], 10);\n // Test for NaN (yes, this is ugly)\n return version !== version ? -1 : version;\n };\n\n const getCanSendMaxMessageSize = function(remoteIsFirefox) {\n // Every implementation we know can send at least 64 KiB.\n // Note: Although Chrome is technically able to send up to 256 KiB, the\n // data does not reach the other peer reliably.\n // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8419\n let canSendMaxMessageSize = 65536;\n if (browserDetails.browser === 'firefox') {\n if (browserDetails.version < 57) {\n if (remoteIsFirefox === -1) {\n // FF < 57 will send in 16 KiB chunks using the deprecated PPID\n // fragmentation.\n canSendMaxMessageSize = 16384;\n } else {\n // However, other FF (and RAWRTC) can reassemble PPID-fragmented\n // messages. Thus, supporting ~2 GiB when sending.\n canSendMaxMessageSize = 2147483637;\n }\n } else if (browserDetails.version < 60) {\n // Currently, all FF >= 57 will reset the remote maximum message size\n // to the default value when a data channel is created at a later\n // stage. :(\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n canSendMaxMessageSize =\n browserDetails.version === 57 ? 65535 : 65536;\n } else {\n // FF >= 60 supports sending ~2 GiB\n canSendMaxMessageSize = 2147483637;\n }\n }\n return canSendMaxMessageSize;\n };\n\n const getMaxMessageSize = function(description, remoteIsFirefox) {\n // Note: 65536 bytes is the default value from the SDP spec. Also,\n // every implementation we know supports receiving 65536 bytes.\n let maxMessageSize = 65536;\n\n // FF 57 has a slightly incorrect default remote max message size, so\n // we need to adjust it here to avoid a failure when sending.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1425697\n if (browserDetails.browser === 'firefox'\n && browserDetails.version === 57) {\n maxMessageSize = 65535;\n }\n\n const match = SDPUtils.matchPrefix(description.sdp,\n 'a=max-message-size:');\n if (match.length > 0) {\n maxMessageSize = parseInt(match[0].substr(19), 10);\n } else if (browserDetails.browser === 'firefox' &&\n remoteIsFirefox !== -1) {\n // If the maximum message size is not present in the remote SDP and\n // both local and remote are Firefox, the remote peer can receive\n // ~2 GiB.\n maxMessageSize = 2147483637;\n }\n return maxMessageSize;\n };\n\n const origSetRemoteDescription =\n window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription() {\n this._sctp = null;\n // Chrome decided to not expose .sctp in plan-b mode.\n // As usual, adapter.js has to do an 'ugly worakaround'\n // to cover up the mess.\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 76) {\n const {sdpSemantics} = this.getConfiguration();\n if (sdpSemantics === 'plan-b') {\n Object.defineProperty(this, 'sctp', {\n get() {\n return typeof this._sctp === 'undefined' ? null : this._sctp;\n },\n enumerable: true,\n configurable: true,\n });\n }\n }\n\n if (sctpInDescription(arguments[0])) {\n // Check if the remote is FF.\n const isFirefox = getRemoteFirefoxVersion(arguments[0]);\n\n // Get the maximum message size the local peer is capable of sending\n const canSendMMS = getCanSendMaxMessageSize(isFirefox);\n\n // Get the maximum message size of the remote peer.\n const remoteMMS = getMaxMessageSize(arguments[0], isFirefox);\n\n // Determine final maximum message size\n let maxMessageSize;\n if (canSendMMS === 0 && remoteMMS === 0) {\n maxMessageSize = Number.POSITIVE_INFINITY;\n } else if (canSendMMS === 0 || remoteMMS === 0) {\n maxMessageSize = Math.max(canSendMMS, remoteMMS);\n } else {\n maxMessageSize = Math.min(canSendMMS, remoteMMS);\n }\n\n // Create a dummy RTCSctpTransport object and the 'maxMessageSize'\n // attribute.\n const sctp = {};\n Object.defineProperty(sctp, 'maxMessageSize', {\n get() {\n return maxMessageSize;\n }\n });\n this._sctp = sctp;\n }\n\n return origSetRemoteDescription.apply(this, arguments);\n };\n}\n\nexport function shimSendThrowTypeError(window) {\n if (!(window.RTCPeerConnection &&\n 'createDataChannel' in window.RTCPeerConnection.prototype)) {\n return;\n }\n\n // Note: Although Firefox >= 57 has a native implementation, the maximum\n // message size can be reset for all data channels at a later stage.\n // See: https://bugzilla.mozilla.org/show_bug.cgi?id=1426831\n\n function wrapDcSend(dc, pc) {\n const origDataChannelSend = dc.send;\n dc.send = function send() {\n const data = arguments[0];\n const length = data.length || data.size || data.byteLength;\n if (dc.readyState === 'open' &&\n pc.sctp && length > pc.sctp.maxMessageSize) {\n throw new TypeError('Message too large (can send a maximum of ' +\n pc.sctp.maxMessageSize + ' bytes)');\n }\n return origDataChannelSend.apply(dc, arguments);\n };\n }\n const origCreateDataChannel =\n window.RTCPeerConnection.prototype.createDataChannel;\n window.RTCPeerConnection.prototype.createDataChannel =\n function createDataChannel() {\n const dataChannel = origCreateDataChannel.apply(this, arguments);\n wrapDcSend(dataChannel, this);\n return dataChannel;\n };\n utils.wrapPeerConnectionEvent(window, 'datachannel', e => {\n wrapDcSend(e.channel, e.target);\n return e;\n });\n}\n\n\n/* shims RTCConnectionState by pretending it is the same as iceConnectionState.\n * See https://bugs.chromium.org/p/webrtc/issues/detail?id=6145#c12\n * for why this is a valid hack in Chrome. In Firefox it is slightly incorrect\n * since DTLS failures would be hidden. See\n * https://bugzilla.mozilla.org/show_bug.cgi?id=1265827\n * for the Firefox tracking bug.\n */\nexport function shimConnectionState(window) {\n if (!window.RTCPeerConnection ||\n 'connectionState' in window.RTCPeerConnection.prototype) {\n return;\n }\n const proto = window.RTCPeerConnection.prototype;\n Object.defineProperty(proto, 'connectionState', {\n get() {\n return {\n completed: 'connected',\n checking: 'connecting'\n }[this.iceConnectionState] || this.iceConnectionState;\n },\n enumerable: true,\n configurable: true\n });\n Object.defineProperty(proto, 'onconnectionstatechange', {\n get() {\n return this._onconnectionstatechange || null;\n },\n set(cb) {\n if (this._onconnectionstatechange) {\n this.removeEventListener('connectionstatechange',\n this._onconnectionstatechange);\n delete this._onconnectionstatechange;\n }\n if (cb) {\n this.addEventListener('connectionstatechange',\n this._onconnectionstatechange = cb);\n }\n },\n enumerable: true,\n configurable: true\n });\n\n ['setLocalDescription', 'setRemoteDescription'].forEach((method) => {\n const origMethod = proto[method];\n proto[method] = function() {\n if (!this._connectionstatechangepoly) {\n this._connectionstatechangepoly = e => {\n const pc = e.target;\n if (pc._lastConnectionState !== pc.connectionState) {\n pc._lastConnectionState = pc.connectionState;\n const newEvent = new Event('connectionstatechange', e);\n pc.dispatchEvent(newEvent);\n }\n return e;\n };\n this.addEventListener('iceconnectionstatechange',\n this._connectionstatechangepoly);\n }\n return origMethod.apply(this, arguments);\n };\n });\n}\n\nexport function removeExtmapAllowMixed(window, browserDetails) {\n /* remove a=extmap-allow-mixed for webrtc.org < M71 */\n if (!window.RTCPeerConnection) {\n return;\n }\n if (browserDetails.browser === 'chrome' && browserDetails.version >= 71) {\n return;\n }\n if (browserDetails.browser === 'safari' && browserDetails.version >= 605) {\n return;\n }\n const nativeSRD = window.RTCPeerConnection.prototype.setRemoteDescription;\n window.RTCPeerConnection.prototype.setRemoteDescription =\n function setRemoteDescription(desc) {\n if (desc && desc.sdp && desc.sdp.indexOf('\\na=extmap-allow-mixed') !== -1) {\n const sdp = desc.sdp.split('\\n').filter((line) => {\n return line.trim() !== 'a=extmap-allow-mixed';\n }).join('\\n');\n // Safari enforces read-only-ness of RTCSessionDescription fields.\n if (window.RTCSessionDescription &&\n desc instanceof window.RTCSessionDescription) {\n arguments[0] = new window.RTCSessionDescription({\n type: desc.type,\n sdp,\n });\n } else {\n desc.sdp = sdp;\n }\n }\n return nativeSRD.apply(this, arguments);\n };\n}\n\nexport function shimAddIceCandidateNullOrEmpty(window, browserDetails) {\n // Support for addIceCandidate(null or undefined)\n // as well as addIceCandidate({candidate: \"\", ...})\n // https://bugs.chromium.org/p/chromium/issues/detail?id=978582\n // Note: must be called before other polyfills which change the signature.\n if (!(window.RTCPeerConnection && window.RTCPeerConnection.prototype)) {\n return;\n }\n const nativeAddIceCandidate =\n window.RTCPeerConnection.prototype.addIceCandidate;\n if (!nativeAddIceCandidate || nativeAddIceCandidate.length === 0) {\n return;\n }\n window.RTCPeerConnection.prototype.addIceCandidate =\n function addIceCandidate() {\n if (!arguments[0]) {\n if (arguments[1]) {\n arguments[1].apply(null);\n }\n return Promise.resolve();\n }\n // Firefox 68+ emits and processes {candidate: \"\", ...}, ignore\n // in older versions.\n // Native support for ignoring exists for Chrome M77+.\n // Safari ignores as well, exact version unknown but works in the same\n // version that also ignores addIceCandidate(null).\n if (((browserDetails.browser === 'chrome' && browserDetails.version < 78)\n || (browserDetails.browser === 'firefox'\n && browserDetails.version < 68)\n || (browserDetails.browser === 'safari'))\n && arguments[0] && arguments[0].candidate === '') {\n return Promise.resolve();\n }\n return nativeAddIceCandidate.apply(this, arguments);\n };\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\nimport * as utils from './utils';\n\n // Browser shims.\nimport * as chromeShim from './chrome/chrome_shim';\nimport * as edgeShim from './edge/edge_shim';\nimport * as firefoxShim from './firefox/firefox_shim';\nimport * as safariShim from './safari/safari_shim';\nimport * as commonShim from './common_shim';\n\n// Shimming starts here.\nexport function adapterFactory({window} = {}, options = {\n shimChrome: true,\n shimFirefox: true,\n shimEdge: true,\n shimSafari: true,\n}) {\n // Utils.\n const logging = utils.log;\n const browserDetails = utils.detectBrowser(window);\n\n const adapter = {\n browserDetails,\n commonShim,\n extractVersion: utils.extractVersion,\n disableLog: utils.disableLog,\n disableWarnings: utils.disableWarnings\n };\n\n // Shim browser if found.\n switch (browserDetails.browser) {\n case 'chrome':\n if (!chromeShim || !chromeShim.shimPeerConnection ||\n !options.shimChrome) {\n logging('Chrome shim is not included in this adapter release.');\n return adapter;\n }\n if (browserDetails.version === null) {\n logging('Chrome shim can not determine version, not shimming.');\n return adapter;\n }\n logging('adapter.js shimming chrome.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = chromeShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n chromeShim.shimGetUserMedia(window, browserDetails);\n chromeShim.shimMediaStream(window, browserDetails);\n chromeShim.shimPeerConnection(window, browserDetails);\n chromeShim.shimOnTrack(window, browserDetails);\n chromeShim.shimAddTrackRemoveTrack(window, browserDetails);\n chromeShim.shimGetSendersWithDtmf(window, browserDetails);\n chromeShim.shimGetStats(window, browserDetails);\n chromeShim.shimSenderReceiverGetStats(window, browserDetails);\n chromeShim.fixNegotiationNeeded(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n case 'firefox':\n if (!firefoxShim || !firefoxShim.shimPeerConnection ||\n !options.shimFirefox) {\n logging('Firefox shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming firefox.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = firefoxShim;\n\n // Must be called before shimPeerConnection.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n firefoxShim.shimGetUserMedia(window, browserDetails);\n firefoxShim.shimPeerConnection(window, browserDetails);\n firefoxShim.shimOnTrack(window, browserDetails);\n firefoxShim.shimRemoveStream(window, browserDetails);\n firefoxShim.shimSenderGetStats(window, browserDetails);\n firefoxShim.shimReceiverGetStats(window, browserDetails);\n firefoxShim.shimRTCDataChannel(window, browserDetails);\n firefoxShim.shimAddTransceiver(window, browserDetails);\n firefoxShim.shimGetParameters(window, browserDetails);\n firefoxShim.shimCreateOffer(window, browserDetails);\n firefoxShim.shimCreateAnswer(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimConnectionState(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'edge':\n if (!edgeShim || !edgeShim.shimPeerConnection || !options.shimEdge) {\n logging('MS edge shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming edge.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = edgeShim;\n\n edgeShim.shimGetUserMedia(window, browserDetails);\n edgeShim.shimGetDisplayMedia(window, browserDetails);\n edgeShim.shimPeerConnection(window, browserDetails);\n edgeShim.shimReplaceTrack(window, browserDetails);\n\n // the edge shim implements the full RTCIceCandidate object.\n\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n break;\n case 'safari':\n if (!safariShim || !options.shimSafari) {\n logging('Safari shim is not included in this adapter release.');\n return adapter;\n }\n logging('adapter.js shimming safari.');\n // Export to the adapter global object visible in the browser.\n adapter.browserShim = safariShim;\n\n // Must be called before shimCallbackAPI.\n commonShim.shimAddIceCandidateNullOrEmpty(window, browserDetails);\n\n safariShim.shimRTCIceServerUrls(window, browserDetails);\n safariShim.shimCreateOfferLegacy(window, browserDetails);\n safariShim.shimCallbacksAPI(window, browserDetails);\n safariShim.shimLocalStreamsAPI(window, browserDetails);\n safariShim.shimRemoteStreamsAPI(window, browserDetails);\n safariShim.shimTrackEventTransceiver(window, browserDetails);\n safariShim.shimGetUserMedia(window, browserDetails);\n safariShim.shimAudioContext(window, browserDetails);\n\n commonShim.shimRTCIceCandidate(window, browserDetails);\n commonShim.shimMaxMessageSize(window, browserDetails);\n commonShim.shimSendThrowTypeError(window, browserDetails);\n commonShim.removeExtmapAllowMixed(window, browserDetails);\n break;\n default:\n logging('Unsupported browser!');\n break;\n }\n\n return adapter;\n}\n","/*\n * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.\n *\n * Use of this source code is governed by a BSD-style license\n * that can be found in the LICENSE file in the root of the source\n * tree.\n */\n/* eslint-env node */\n\n'use strict';\n\nimport {adapterFactory} from './adapter_factory.js';\n\nconst adapter =\n adapterFactory({window: typeof window === 'undefined' ? undefined : window});\nexport default adapter;\n","\n'use strict';\nimport * as utils from './utils.js';\nimport * as MediaFormatModule from './mediaformat.js';\nimport adapter from 'webrtc-adapter';\n\n/**\n * @class AudioTrackConstraints\n * @classDesc Constraints for creating an audio MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.AudioSourceInfo} source Source info of this audio track.\n */\nexport class AudioTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.AudioSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Values could be \"mic\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.AudioTrackConstraints\n * @desc Do not provide deviceId if source is not \"mic\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n this.deviceId = undefined;\n }\n}\n\n/**\n * @class VideoTrackConstraints\n * @classDesc Constraints for creating a video MediaStreamTrack.\n * @memberof Owt.Base\n * @constructor\n * @param {Owt.Base.VideoSourceInfo} source Source info of this video track.\n */\nexport class VideoTrackConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(source) {\n if (!Object.values(MediaFormatModule.VideoSourceInfo)\n .some((v) => v === source)) {\n throw new TypeError('Invalid source.');\n }\n /**\n * @member {string} source\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Values could be \"camera\", \"screen-cast\", \"file\" or \"mixed\".\n * @instance\n */\n this.source = source;\n /**\n * @member {string} deviceId\n * @memberof Owt.Base.VideoTrackConstraints\n * @desc Do not provide deviceId if source is not \"camera\".\n * @instance\n * @see https://w3c.github.io/mediacapture-main/#def-constraint-deviceId\n */\n\n this.deviceId = undefined;\n\n /**\n * @member {Owt.Base.Resolution} resolution\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.resolution = undefined;\n\n /**\n * @member {number} frameRate\n * @memberof Owt.Base.VideoTrackConstraints\n * @instance\n */\n this.frameRate = undefined;\n }\n}\n/**\n * @class StreamConstraints\n * @classDesc Constraints for creating a MediaStream from screen mic and camera.\n * @memberof Owt.Base\n * @constructor\n * @param {?Owt.Base.AudioTrackConstraints} audioConstraints\n * @param {?Owt.Base.VideoTrackConstraints} videoConstraints\n */\nexport class StreamConstraints {\n // eslint-disable-next-line require-jsdoc\n constructor(audioConstraints = false, videoConstraints = false) {\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForAudio} audio\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.audio = audioConstraints;\n /**\n * @member {Owt.Base.MediaStreamTrackDeviceConstraintsForVideo} Video\n * @memberof Owt.Base.MediaStreamDeviceConstraints\n * @instance\n */\n this.video = videoConstraints;\n }\n}\n\n// eslint-disable-next-line require-jsdoc\nfunction isVideoConstrainsForScreenCast(constraints) {\n return (typeof constraints.video === 'object' && constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST);\n}\n\n/**\n * @class MediaStreamFactory\n * @classDesc A factory to create MediaStream. You can also create MediaStream by yourself.\n * @memberof Owt.Base\n */\nexport class MediaStreamFactory {\n /**\n * @function createMediaStream\n * @static\n * @desc Create a MediaStream with given constraints. If you want to create a MediaStream for screen cast, please make sure both audio and video's source are \"screen-cast\".\n * @memberof Owt.Base.MediaStreamFactory\n * @return {Promise} Return a promise that is resolved when stream is successfully created, or rejected if one of the following error happened:\n * - One or more parameters cannot be satisfied.\n * - Specified device is busy.\n * - Cannot obtain necessary permission or operation is canceled by user.\n * - Video source is screen cast, while audio source is not.\n * - Audio source is screen cast, while video source is disabled.\n * @param {Owt.Base.StreamConstraints} constraints\n */\n static createMediaStream(constraints) {\n if (typeof constraints !== 'object' ||\n (!constraints.audio && !constraints.video)) {\n return Promise.reject(new TypeError('Invalid constrains'));\n }\n if (!isVideoConstrainsForScreenCast(constraints) &&\n (typeof constraints.audio === 'object') &&\n constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(\n new TypeError('Cannot share screen without video.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) && !utils.isChrome() &&\n !utils.isFirefox()) {\n return Promise.reject(\n new TypeError('Screen sharing only supports Chrome and Firefox.'));\n }\n if (isVideoConstrainsForScreenCast(constraints) &&\n typeof constraints.audio === 'object' &&\n constraints.audio.source !==\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n return Promise.reject(new TypeError(\n 'Cannot capture video from screen cast while capture audio from'\n + ' other source.'));\n }\n\n // Check and convert constraints.\n if (!constraints.audio && !constraints.video) {\n return Promise.reject(new TypeError(\n 'At least one of audio and video must be requested.'));\n }\n const mediaConstraints = Object.create({});\n if (typeof constraints.audio === 'object' &&\n constraints.audio.source === MediaFormatModule.AudioSourceInfo.MIC) {\n mediaConstraints.audio = Object.create({});\n if (utils.isEdge()) {\n mediaConstraints.audio.deviceId = constraints.audio.deviceId;\n } else {\n mediaConstraints.audio.deviceId = {\n exact: constraints.audio.deviceId,\n };\n }\n } else {\n if (constraints.audio.source ===\n MediaFormatModule.AudioSourceInfo.SCREENCAST) {\n mediaConstraints.audio = true;\n } else {\n mediaConstraints.audio = constraints.audio;\n }\n }\n if (typeof constraints.video === 'object') {\n mediaConstraints.video = Object.create({});\n if (typeof constraints.video.frameRate === 'number') {\n mediaConstraints.video.frameRate = constraints.video.frameRate;\n }\n if (constraints.video.resolution &&\n constraints.video.resolution.width &&\n constraints.video.resolution.height) {\n if (constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.width = constraints.video.resolution.width;\n mediaConstraints.video.height = constraints.video.resolution.height;\n } else {\n mediaConstraints.video.width = Object.create({});\n mediaConstraints.video.width.exact =\n constraints.video.resolution.width;\n mediaConstraints.video.height = Object.create({});\n mediaConstraints.video.height.exact =\n constraints.video.resolution.height;\n }\n }\n if (typeof constraints.video.deviceId === 'string') {\n mediaConstraints.video.deviceId = {exact: constraints.video.deviceId};\n }\n if (utils.isFirefox() &&\n constraints.video.source ===\n MediaFormatModule.VideoSourceInfo.SCREENCAST) {\n mediaConstraints.video.mediaSource = 'screen';\n }\n } else {\n mediaConstraints.video = constraints.video;\n }\n\n if (isVideoConstrainsForScreenCast(constraints)) {\n return navigator.mediaDevices.getDisplayMedia(mediaConstraints);\n } else {\n return navigator.mediaDevices.getUserMedia(mediaConstraints);\n }\n }\n}\n","// Copyright (C) <2018> Intel Corporation\n//\n// SPDX-License-Identifier: Apache-2.0\n\n'use strict';\n\nexport * from './mediastream-factory.js';\nexport * from './mediaformat.js';","let logger;\nlet errorLogger;\n\nexport function setLogger() {\n /*eslint-disable */\n logger = console.log;\n errorLogger = console.error;\n /*eslint-enable */\n}\n\nexport function isEnable() {\n return logger != null;\n}\n\nexport function log(message, ...optionalParams) {\n if (logger) {\n logger(message, ...optionalParams);\n }\n}\nexport function error(message, ...optionalParams) {\n if (errorLogger) {\n errorLogger(message, ...optionalParams);\n }\n}\n","export default class Event {\n constructor(type) {\n this.listener = {};\n this.type = type | '';\n }\n\n on(event, fn) {\n if (!this.listener[event]) {\n this.listener[event] = [];\n }\n this.listener[event].push(fn);\n return true;\n }\n\n off(event, fn) {\n if (this.listener[event]) {\n var index = this.listener[event].indexOf(fn);\n if (index > -1) {\n this.listener[event].splice(index, 1);\n }\n return true;\n }\n return false;\n }\n\n offAll() {\n this.listener = {};\n }\n\n dispatch(event, data) {\n if (this.listener[event]) {\n this.listener[event].map((each) => {\n each.apply(null, [data]);\n });\n return true;\n }\n return false;\n }\n}\n","'use strict';\n\nmodule.exports = function bind(fn, thisArg) {\n return function wrap() {\n var args = new Array(arguments.length);\n for (var i = 0; i < args.length; i++) {\n args[i] = arguments[i];\n }\n return fn.apply(thisArg, args);\n };\n};\n","'use strict';\n\nvar bind = require('./helpers/bind');\n\n/*global toString:true*/\n\n// utils is a library of generic helper functions non-specific to axios\n\nvar toString = Object.prototype.toString;\n\n/**\n * Determine if a value is an Array\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Array, otherwise false\n */\nfunction isArray(val) {\n return toString.call(val) === '[object Array]';\n}\n\n/**\n * Determine if a value is undefined\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if the value is undefined, otherwise false\n */\nfunction isUndefined(val) {\n return typeof val === 'undefined';\n}\n\n/**\n * Determine if a value is a Buffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Buffer, otherwise false\n */\nfunction isBuffer(val) {\n return val !== null && !isUndefined(val) && val.constructor !== null && !isUndefined(val.constructor)\n && typeof val.constructor.isBuffer === 'function' && val.constructor.isBuffer(val);\n}\n\n/**\n * Determine if a value is an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an ArrayBuffer, otherwise false\n */\nfunction isArrayBuffer(val) {\n return toString.call(val) === '[object ArrayBuffer]';\n}\n\n/**\n * Determine if a value is a FormData\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an FormData, otherwise false\n */\nfunction isFormData(val) {\n return (typeof FormData !== 'undefined') && (val instanceof FormData);\n}\n\n/**\n * Determine if a value is a view on an ArrayBuffer\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a view on an ArrayBuffer, otherwise false\n */\nfunction isArrayBufferView(val) {\n var result;\n if ((typeof ArrayBuffer !== 'undefined') && (ArrayBuffer.isView)) {\n result = ArrayBuffer.isView(val);\n } else {\n result = (val) && (val.buffer) && (val.buffer instanceof ArrayBuffer);\n }\n return result;\n}\n\n/**\n * Determine if a value is a String\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a String, otherwise false\n */\nfunction isString(val) {\n return typeof val === 'string';\n}\n\n/**\n * Determine if a value is a Number\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Number, otherwise false\n */\nfunction isNumber(val) {\n return typeof val === 'number';\n}\n\n/**\n * Determine if a value is an Object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is an Object, otherwise false\n */\nfunction isObject(val) {\n return val !== null && typeof val === 'object';\n}\n\n/**\n * Determine if a value is a plain Object\n *\n * @param {Object} val The value to test\n * @return {boolean} True if value is a plain Object, otherwise false\n */\nfunction isPlainObject(val) {\n if (toString.call(val) !== '[object Object]') {\n return false;\n }\n\n var prototype = Object.getPrototypeOf(val);\n return prototype === null || prototype === Object.prototype;\n}\n\n/**\n * Determine if a value is a Date\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Date, otherwise false\n */\nfunction isDate(val) {\n return toString.call(val) === '[object Date]';\n}\n\n/**\n * Determine if a value is a File\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a File, otherwise false\n */\nfunction isFile(val) {\n return toString.call(val) === '[object File]';\n}\n\n/**\n * Determine if a value is a Blob\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Blob, otherwise false\n */\nfunction isBlob(val) {\n return toString.call(val) === '[object Blob]';\n}\n\n/**\n * Determine if a value is a Function\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Function, otherwise false\n */\nfunction isFunction(val) {\n return toString.call(val) === '[object Function]';\n}\n\n/**\n * Determine if a value is a Stream\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a Stream, otherwise false\n */\nfunction isStream(val) {\n return isObject(val) && isFunction(val.pipe);\n}\n\n/**\n * Determine if a value is a URLSearchParams object\n *\n * @param {Object} val The value to test\n * @returns {boolean} True if value is a URLSearchParams object, otherwise false\n */\nfunction isURLSearchParams(val) {\n return typeof URLSearchParams !== 'undefined' && val instanceof URLSearchParams;\n}\n\n/**\n * Trim excess whitespace off the beginning and end of a string\n *\n * @param {String} str The String to trim\n * @returns {String} The String freed of excess whitespace\n */\nfunction trim(str) {\n return str.replace(/^\\s*/, '').replace(/\\s*$/, '');\n}\n\n/**\n * Determine if we're running in a standard browser environment\n *\n * This allows axios to run in a web worker, and react-native.\n * Both environments support XMLHttpRequest, but not fully standard globals.\n *\n * web workers:\n * typeof window -> undefined\n * typeof document -> undefined\n *\n * react-native:\n * navigator.product -> 'ReactNative'\n * nativescript\n * navigator.product -> 'NativeScript' or 'NS'\n */\nfunction isStandardBrowserEnv() {\n if (typeof navigator !== 'undefined' && (navigator.product === 'ReactNative' ||\n navigator.product === 'NativeScript' ||\n navigator.product === 'NS')) {\n return false;\n }\n return (\n typeof window !== 'undefined' &&\n typeof document !== 'undefined'\n );\n}\n\n/**\n * Iterate over an Array or an Object invoking a function for each item.\n *\n * If `obj` is an Array callback will be called passing\n * the value, index, and complete array for each item.\n *\n * If 'obj' is an Object callback will be called passing\n * the value, key, and complete object for each property.\n *\n * @param {Object|Array} obj The object to iterate\n * @param {Function} fn The callback to invoke for each item\n */\nfunction forEach(obj, fn) {\n // Don't bother if no value provided\n if (obj === null || typeof obj === 'undefined') {\n return;\n }\n\n // Force an array if not already something iterable\n if (typeof obj !== 'object') {\n /*eslint no-param-reassign:0*/\n obj = [obj];\n }\n\n if (isArray(obj)) {\n // Iterate over array values\n for (var i = 0, l = obj.length; i < l; i++) {\n fn.call(null, obj[i], i, obj);\n }\n } else {\n // Iterate over object keys\n for (var key in obj) {\n if (Object.prototype.hasOwnProperty.call(obj, key)) {\n fn.call(null, obj[key], key, obj);\n }\n }\n }\n}\n\n/**\n * Accepts varargs expecting each argument to be an object, then\n * immutably merges the properties of each object and returns result.\n *\n * When multiple objects contain the same key the later object in\n * the arguments list will take precedence.\n *\n * Example:\n *\n * ```js\n * var result = merge({foo: 123}, {foo: 456});\n * console.log(result.foo); // outputs 456\n * ```\n *\n * @param {Object} obj1 Object to merge\n * @returns {Object} Result of all merge properties\n */\nfunction merge(/* obj1, obj2, obj3, ... */) {\n var result = {};\n function assignValue(val, key) {\n if (isPlainObject(result[key]) && isPlainObject(val)) {\n result[key] = merge(result[key], val);\n } else if (isPlainObject(val)) {\n result[key] = merge({}, val);\n } else if (isArray(val)) {\n result[key] = val.slice();\n } else {\n result[key] = val;\n }\n }\n\n for (var i = 0, l = arguments.length; i < l; i++) {\n forEach(arguments[i], assignValue);\n }\n return result;\n}\n\n/**\n * Extends object a by mutably adding to it the properties of object b.\n *\n * @param {Object} a The object to be extended\n * @param {Object} b The object to copy properties from\n * @param {Object} thisArg The object to bind function to\n * @return {Object} The resulting value of object a\n */\nfunction extend(a, b, thisArg) {\n forEach(b, function assignValue(val, key) {\n if (thisArg && typeof val === 'function') {\n a[key] = bind(val, thisArg);\n } else {\n a[key] = val;\n }\n });\n return a;\n}\n\n/**\n * Remove byte order marker. This catches EF BB BF (the UTF-8 BOM)\n *\n * @param {string} content with BOM\n * @return {string} content value without BOM\n */\nfunction stripBOM(content) {\n if (content.charCodeAt(0) === 0xFEFF) {\n content = content.slice(1);\n }\n return content;\n}\n\nmodule.exports = {\n isArray: isArray,\n isArrayBuffer: isArrayBuffer,\n isBuffer: isBuffer,\n isFormData: isFormData,\n isArrayBufferView: isArrayBufferView,\n isString: isString,\n isNumber: isNumber,\n isObject: isObject,\n isPlainObject: isPlainObject,\n isUndefined: isUndefined,\n isDate: isDate,\n isFile: isFile,\n isBlob: isBlob,\n isFunction: isFunction,\n isStream: isStream,\n isURLSearchParams: isURLSearchParams,\n isStandardBrowserEnv: isStandardBrowserEnv,\n forEach: forEach,\n merge: merge,\n extend: extend,\n trim: trim,\n stripBOM: stripBOM\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction encode(val) {\n return encodeURIComponent(val).\n replace(/%3A/gi, ':').\n replace(/%24/g, '$').\n replace(/%2C/gi, ',').\n replace(/%20/g, '+').\n replace(/%5B/gi, '[').\n replace(/%5D/gi, ']');\n}\n\n/**\n * Build a URL by appending params to the end\n *\n * @param {string} url The base of the url (e.g., http://www.google.com)\n * @param {object} [params] The params to be appended\n * @returns {string} The formatted url\n */\nmodule.exports = function buildURL(url, params, paramsSerializer) {\n /*eslint no-param-reassign:0*/\n if (!params) {\n return url;\n }\n\n var serializedParams;\n if (paramsSerializer) {\n serializedParams = paramsSerializer(params);\n } else if (utils.isURLSearchParams(params)) {\n serializedParams = params.toString();\n } else {\n var parts = [];\n\n utils.forEach(params, function serialize(val, key) {\n if (val === null || typeof val === 'undefined') {\n return;\n }\n\n if (utils.isArray(val)) {\n key = key + '[]';\n } else {\n val = [val];\n }\n\n utils.forEach(val, function parseValue(v) {\n if (utils.isDate(v)) {\n v = v.toISOString();\n } else if (utils.isObject(v)) {\n v = JSON.stringify(v);\n }\n parts.push(encode(key) + '=' + encode(v));\n });\n });\n\n serializedParams = parts.join('&');\n }\n\n if (serializedParams) {\n var hashmarkIndex = url.indexOf('#');\n if (hashmarkIndex !== -1) {\n url = url.slice(0, hashmarkIndex);\n }\n\n url += (url.indexOf('?') === -1 ? '?' : '&') + serializedParams;\n }\n\n return url;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nfunction InterceptorManager() {\n this.handlers = [];\n}\n\n/**\n * Add a new interceptor to the stack\n *\n * @param {Function} fulfilled The function to handle `then` for a `Promise`\n * @param {Function} rejected The function to handle `reject` for a `Promise`\n *\n * @return {Number} An ID used to remove interceptor later\n */\nInterceptorManager.prototype.use = function use(fulfilled, rejected) {\n this.handlers.push({\n fulfilled: fulfilled,\n rejected: rejected\n });\n return this.handlers.length - 1;\n};\n\n/**\n * Remove an interceptor from the stack\n *\n * @param {Number} id The ID that was returned by `use`\n */\nInterceptorManager.prototype.eject = function eject(id) {\n if (this.handlers[id]) {\n this.handlers[id] = null;\n }\n};\n\n/**\n * Iterate over all the registered interceptors\n *\n * This method is particularly useful for skipping over any\n * interceptors that may have become `null` calling `eject`.\n *\n * @param {Function} fn The function to call for each interceptor\n */\nInterceptorManager.prototype.forEach = function forEach(fn) {\n utils.forEach(this.handlers, function forEachHandler(h) {\n if (h !== null) {\n fn(h);\n }\n });\n};\n\nmodule.exports = InterceptorManager;\n","'use strict';\n\nvar utils = require('./../utils');\n\n/**\n * Transform the data for a request or a response\n *\n * @param {Object|String} data The data to be transformed\n * @param {Array} headers The headers for the request or response\n * @param {Array|Function} fns A single function or Array of functions\n * @returns {*} The resulting transformed data\n */\nmodule.exports = function transformData(data, headers, fns) {\n /*eslint no-param-reassign:0*/\n utils.forEach(fns, function transform(fn) {\n data = fn(data, headers);\n });\n\n return data;\n};\n","'use strict';\n\nmodule.exports = function isCancel(value) {\n return !!(value && value.__CANCEL__);\n};\n","'use strict';\n\nvar utils = require('../utils');\n\nmodule.exports = function normalizeHeaderName(headers, normalizedName) {\n utils.forEach(headers, function processHeader(value, name) {\n if (name !== normalizedName && name.toUpperCase() === normalizedName.toUpperCase()) {\n headers[normalizedName] = value;\n delete headers[name];\n }\n });\n};\n","'use strict';\n\n/**\n * Update an Error with the specified config, error code, and response.\n *\n * @param {Error} error The error to update.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The error.\n */\nmodule.exports = function enhanceError(error, config, code, request, response) {\n error.config = config;\n if (code) {\n error.code = code;\n }\n\n error.request = request;\n error.response = response;\n error.isAxiosError = true;\n\n error.toJSON = function toJSON() {\n return {\n // Standard\n message: this.message,\n name: this.name,\n // Microsoft\n description: this.description,\n number: this.number,\n // Mozilla\n fileName: this.fileName,\n lineNumber: this.lineNumber,\n columnNumber: this.columnNumber,\n stack: this.stack,\n // Axios\n config: this.config,\n code: this.code\n };\n };\n return error;\n};\n","'use strict';\n\nvar enhanceError = require('./enhanceError');\n\n/**\n * Create an Error with the specified message, config, error code, request and response.\n *\n * @param {string} message The error message.\n * @param {Object} config The config.\n * @param {string} [code] The error code (for example, 'ECONNABORTED').\n * @param {Object} [request] The request.\n * @param {Object} [response] The response.\n * @returns {Error} The created error.\n */\nmodule.exports = function createError(message, config, code, request, response) {\n var error = new Error(message);\n return enhanceError(error, config, code, request, response);\n};\n","'use strict';\n\nvar createError = require('./createError');\n\n/**\n * Resolve or reject a Promise based on response status.\n *\n * @param {Function} resolve A function that resolves the promise.\n * @param {Function} reject A function that rejects the promise.\n * @param {object} response The response.\n */\nmodule.exports = function settle(resolve, reject, response) {\n var validateStatus = response.config.validateStatus;\n if (!response.status || !validateStatus || validateStatus(response.status)) {\n resolve(response);\n } else {\n reject(createError(\n 'Request failed with status code ' + response.status,\n response.config,\n null,\n response.request,\n response\n ));\n }\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs support document.cookie\n (function standardBrowserEnv() {\n return {\n write: function write(name, value, expires, path, domain, secure) {\n var cookie = [];\n cookie.push(name + '=' + encodeURIComponent(value));\n\n if (utils.isNumber(expires)) {\n cookie.push('expires=' + new Date(expires).toGMTString());\n }\n\n if (utils.isString(path)) {\n cookie.push('path=' + path);\n }\n\n if (utils.isString(domain)) {\n cookie.push('domain=' + domain);\n }\n\n if (secure === true) {\n cookie.push('secure');\n }\n\n document.cookie = cookie.join('; ');\n },\n\n read: function read(name) {\n var match = document.cookie.match(new RegExp('(^|;\\\\s*)(' + name + ')=([^;]*)'));\n return (match ? decodeURIComponent(match[3]) : null);\n },\n\n remove: function remove(name) {\n this.write(name, '', Date.now() - 86400000);\n }\n };\n })() :\n\n // Non standard browser env (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return {\n write: function write() {},\n read: function read() { return null; },\n remove: function remove() {}\n };\n })()\n);\n","'use strict';\n\n/**\n * Determines whether the specified URL is absolute\n *\n * @param {string} url The URL to test\n * @returns {boolean} True if the specified URL is absolute, otherwise false\n */\nmodule.exports = function isAbsoluteURL(url) {\n // A URL is considered absolute if it begins with \"://\" or \"//\" (protocol-relative URL).\n // RFC 3986 defines scheme name as a sequence of characters beginning with a letter and followed\n // by any combination of letters, digits, plus, period, or hyphen.\n return /^([a-z][a-z\\d\\+\\-\\.]*:)?\\/\\//i.test(url);\n};\n","'use strict';\n\n/**\n * Creates a new URL by combining the specified URLs\n *\n * @param {string} baseURL The base URL\n * @param {string} relativeURL The relative URL\n * @returns {string} The combined URL\n */\nmodule.exports = function combineURLs(baseURL, relativeURL) {\n return relativeURL\n ? baseURL.replace(/\\/+$/, '') + '/' + relativeURL.replace(/^\\/+/, '')\n : baseURL;\n};\n","'use strict';\n\nvar isAbsoluteURL = require('../helpers/isAbsoluteURL');\nvar combineURLs = require('../helpers/combineURLs');\n\n/**\n * Creates a new URL by combining the baseURL with the requestedURL,\n * only when the requestedURL is not already an absolute URL.\n * If the requestURL is absolute, this function returns the requestedURL untouched.\n *\n * @param {string} baseURL The base URL\n * @param {string} requestedURL Absolute or relative URL to combine\n * @returns {string} The combined full path\n */\nmodule.exports = function buildFullPath(baseURL, requestedURL) {\n if (baseURL && !isAbsoluteURL(requestedURL)) {\n return combineURLs(baseURL, requestedURL);\n }\n return requestedURL;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\n// Headers whose duplicates are ignored by node\n// c.f. https://nodejs.org/api/http.html#http_message_headers\nvar ignoreDuplicateOf = [\n 'age', 'authorization', 'content-length', 'content-type', 'etag',\n 'expires', 'from', 'host', 'if-modified-since', 'if-unmodified-since',\n 'last-modified', 'location', 'max-forwards', 'proxy-authorization',\n 'referer', 'retry-after', 'user-agent'\n];\n\n/**\n * Parse headers into an object\n *\n * ```\n * Date: Wed, 27 Aug 2014 08:58:49 GMT\n * Content-Type: application/json\n * Connection: keep-alive\n * Transfer-Encoding: chunked\n * ```\n *\n * @param {String} headers Headers needing to be parsed\n * @returns {Object} Headers parsed into an object\n */\nmodule.exports = function parseHeaders(headers) {\n var parsed = {};\n var key;\n var val;\n var i;\n\n if (!headers) { return parsed; }\n\n utils.forEach(headers.split('\\n'), function parser(line) {\n i = line.indexOf(':');\n key = utils.trim(line.substr(0, i)).toLowerCase();\n val = utils.trim(line.substr(i + 1));\n\n if (key) {\n if (parsed[key] && ignoreDuplicateOf.indexOf(key) >= 0) {\n return;\n }\n if (key === 'set-cookie') {\n parsed[key] = (parsed[key] ? parsed[key] : []).concat([val]);\n } else {\n parsed[key] = parsed[key] ? parsed[key] + ', ' + val : val;\n }\n }\n });\n\n return parsed;\n};\n","'use strict';\n\nvar utils = require('./../utils');\n\nmodule.exports = (\n utils.isStandardBrowserEnv() ?\n\n // Standard browser envs have full support of the APIs needed to test\n // whether the request URL is of the same origin as current location.\n (function standardBrowserEnv() {\n var msie = /(msie|trident)/i.test(navigator.userAgent);\n var urlParsingNode = document.createElement('a');\n var originURL;\n\n /**\n * Parse a URL to discover it's components\n *\n * @param {String} url The URL to be parsed\n * @returns {Object}\n */\n function resolveURL(url) {\n var href = url;\n\n if (msie) {\n // IE needs attribute set twice to normalize properties\n urlParsingNode.setAttribute('href', href);\n href = urlParsingNode.href;\n }\n\n urlParsingNode.setAttribute('href', href);\n\n // urlParsingNode provides the UrlUtils interface - http://url.spec.whatwg.org/#urlutils\n return {\n href: urlParsingNode.href,\n protocol: urlParsingNode.protocol ? urlParsingNode.protocol.replace(/:$/, '') : '',\n host: urlParsingNode.host,\n search: urlParsingNode.search ? urlParsingNode.search.replace(/^\\?/, '') : '',\n hash: urlParsingNode.hash ? urlParsingNode.hash.replace(/^#/, '') : '',\n hostname: urlParsingNode.hostname,\n port: urlParsingNode.port,\n pathname: (urlParsingNode.pathname.charAt(0) === '/') ?\n urlParsingNode.pathname :\n '/' + urlParsingNode.pathname\n };\n }\n\n originURL = resolveURL(window.location.href);\n\n /**\n * Determine if a URL shares the same origin as the current location\n *\n * @param {String} requestURL The URL to test\n * @returns {boolean} True if URL shares the same origin, otherwise false\n */\n return function isURLSameOrigin(requestURL) {\n var parsed = (utils.isString(requestURL)) ? resolveURL(requestURL) : requestURL;\n return (parsed.protocol === originURL.protocol &&\n parsed.host === originURL.host);\n };\n })() :\n\n // Non standard browser envs (web workers, react-native) lack needed support.\n (function nonStandardBrowserEnv() {\n return function isURLSameOrigin() {\n return true;\n };\n })()\n);\n","'use strict';\n\nvar utils = require('./../utils');\nvar settle = require('./../core/settle');\nvar cookies = require('./../helpers/cookies');\nvar buildURL = require('./../helpers/buildURL');\nvar buildFullPath = require('../core/buildFullPath');\nvar parseHeaders = require('./../helpers/parseHeaders');\nvar isURLSameOrigin = require('./../helpers/isURLSameOrigin');\nvar createError = require('../core/createError');\n\nmodule.exports = function xhrAdapter(config) {\n return new Promise(function dispatchXhrRequest(resolve, reject) {\n var requestData = config.data;\n var requestHeaders = config.headers;\n\n if (utils.isFormData(requestData)) {\n delete requestHeaders['Content-Type']; // Let the browser set it\n }\n\n var request = new XMLHttpRequest();\n\n // HTTP basic authentication\n if (config.auth) {\n var username = config.auth.username || '';\n var password = config.auth.password ? unescape(encodeURIComponent(config.auth.password)) : '';\n requestHeaders.Authorization = 'Basic ' + btoa(username + ':' + password);\n }\n\n var fullPath = buildFullPath(config.baseURL, config.url);\n request.open(config.method.toUpperCase(), buildURL(fullPath, config.params, config.paramsSerializer), true);\n\n // Set the request timeout in MS\n request.timeout = config.timeout;\n\n // Listen for ready state\n request.onreadystatechange = function handleLoad() {\n if (!request || request.readyState !== 4) {\n return;\n }\n\n // The request errored out and we didn't get a response, this will be\n // handled by onerror instead\n // With one exception: request that using file: protocol, most browsers\n // will return status as 0 even though it's a successful request\n if (request.status === 0 && !(request.responseURL && request.responseURL.indexOf('file:') === 0)) {\n return;\n }\n\n // Prepare the response\n var responseHeaders = 'getAllResponseHeaders' in request ? parseHeaders(request.getAllResponseHeaders()) : null;\n var responseData = !config.responseType || config.responseType === 'text' ? request.responseText : request.response;\n var response = {\n data: responseData,\n status: request.status,\n statusText: request.statusText,\n headers: responseHeaders,\n config: config,\n request: request\n };\n\n settle(resolve, reject, response);\n\n // Clean up request\n request = null;\n };\n\n // Handle browser request cancellation (as opposed to a manual cancellation)\n request.onabort = function handleAbort() {\n if (!request) {\n return;\n }\n\n reject(createError('Request aborted', config, 'ECONNABORTED', request));\n\n // Clean up request\n request = null;\n };\n\n // Handle low level network errors\n request.onerror = function handleError() {\n // Real errors are hidden from us by the browser\n // onerror should only fire if it's a network error\n reject(createError('Network Error', config, null, request));\n\n // Clean up request\n request = null;\n };\n\n // Handle timeout\n request.ontimeout = function handleTimeout() {\n var timeoutErrorMessage = 'timeout of ' + config.timeout + 'ms exceeded';\n if (config.timeoutErrorMessage) {\n timeoutErrorMessage = config.timeoutErrorMessage;\n }\n reject(createError(timeoutErrorMessage, config, 'ECONNABORTED',\n request));\n\n // Clean up request\n request = null;\n };\n\n // Add xsrf header\n // This is only done if running in a standard browser environment.\n // Specifically not if we're in a web worker, or react-native.\n if (utils.isStandardBrowserEnv()) {\n // Add xsrf header\n var xsrfValue = (config.withCredentials || isURLSameOrigin(fullPath)) && config.xsrfCookieName ?\n cookies.read(config.xsrfCookieName) :\n undefined;\n\n if (xsrfValue) {\n requestHeaders[config.xsrfHeaderName] = xsrfValue;\n }\n }\n\n // Add headers to the request\n if ('setRequestHeader' in request) {\n utils.forEach(requestHeaders, function setRequestHeader(val, key) {\n if (typeof requestData === 'undefined' && key.toLowerCase() === 'content-type') {\n // Remove Content-Type if data is undefined\n delete requestHeaders[key];\n } else {\n // Otherwise add header to the request\n request.setRequestHeader(key, val);\n }\n });\n }\n\n // Add withCredentials to request if needed\n if (!utils.isUndefined(config.withCredentials)) {\n request.withCredentials = !!config.withCredentials;\n }\n\n // Add responseType to request if needed\n if (config.responseType) {\n try {\n request.responseType = config.responseType;\n } catch (e) {\n // Expected DOMException thrown by browsers not compatible XMLHttpRequest Level 2.\n // But, this can be suppressed for 'json' type as it can be parsed by default 'transformResponse' function.\n if (config.responseType !== 'json') {\n throw e;\n }\n }\n }\n\n // Handle progress if needed\n if (typeof config.onDownloadProgress === 'function') {\n request.addEventListener('progress', config.onDownloadProgress);\n }\n\n // Not all browsers support upload events\n if (typeof config.onUploadProgress === 'function' && request.upload) {\n request.upload.addEventListener('progress', config.onUploadProgress);\n }\n\n if (config.cancelToken) {\n // Handle cancellation\n config.cancelToken.promise.then(function onCanceled(cancel) {\n if (!request) {\n return;\n }\n\n request.abort();\n reject(cancel);\n // Clean up request\n request = null;\n });\n }\n\n if (!requestData) {\n requestData = null;\n }\n\n // Send the request\n request.send(requestData);\n });\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar normalizeHeaderName = require('./helpers/normalizeHeaderName');\n\nvar DEFAULT_CONTENT_TYPE = {\n 'Content-Type': 'application/x-www-form-urlencoded'\n};\n\nfunction setContentTypeIfUnset(headers, value) {\n if (!utils.isUndefined(headers) && utils.isUndefined(headers['Content-Type'])) {\n headers['Content-Type'] = value;\n }\n}\n\nfunction getDefaultAdapter() {\n var adapter;\n if (typeof XMLHttpRequest !== 'undefined') {\n // For browsers use XHR adapter\n adapter = require('./adapters/xhr');\n } else if (typeof process !== 'undefined' && Object.prototype.toString.call(process) === '[object process]') {\n // For node use HTTP adapter\n adapter = require('./adapters/http');\n }\n return adapter;\n}\n\nvar defaults = {\n adapter: getDefaultAdapter(),\n\n transformRequest: [function transformRequest(data, headers) {\n normalizeHeaderName(headers, 'Accept');\n normalizeHeaderName(headers, 'Content-Type');\n if (utils.isFormData(data) ||\n utils.isArrayBuffer(data) ||\n utils.isBuffer(data) ||\n utils.isStream(data) ||\n utils.isFile(data) ||\n utils.isBlob(data)\n ) {\n return data;\n }\n if (utils.isArrayBufferView(data)) {\n return data.buffer;\n }\n if (utils.isURLSearchParams(data)) {\n setContentTypeIfUnset(headers, 'application/x-www-form-urlencoded;charset=utf-8');\n return data.toString();\n }\n if (utils.isObject(data)) {\n setContentTypeIfUnset(headers, 'application/json;charset=utf-8');\n return JSON.stringify(data);\n }\n return data;\n }],\n\n transformResponse: [function transformResponse(data) {\n /*eslint no-param-reassign:0*/\n if (typeof data === 'string') {\n try {\n data = JSON.parse(data);\n } catch (e) { /* Ignore */ }\n }\n return data;\n }],\n\n /**\n * A timeout in milliseconds to abort a request. If set to 0 (default) a\n * timeout is not created.\n */\n timeout: 0,\n\n xsrfCookieName: 'XSRF-TOKEN',\n xsrfHeaderName: 'X-XSRF-TOKEN',\n\n maxContentLength: -1,\n maxBodyLength: -1,\n\n validateStatus: function validateStatus(status) {\n return status >= 200 && status < 300;\n }\n};\n\ndefaults.headers = {\n common: {\n 'Accept': 'application/json, text/plain, */*'\n }\n};\n\nutils.forEach(['delete', 'get', 'head'], function forEachMethodNoData(method) {\n defaults.headers[method] = {};\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n defaults.headers[method] = utils.merge(DEFAULT_CONTENT_TYPE);\n});\n\nmodule.exports = defaults;\n","'use strict';\n\nvar utils = require('./../utils');\nvar transformData = require('./transformData');\nvar isCancel = require('../cancel/isCancel');\nvar defaults = require('../defaults');\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nfunction throwIfCancellationRequested(config) {\n if (config.cancelToken) {\n config.cancelToken.throwIfRequested();\n }\n}\n\n/**\n * Dispatch a request to the server using the configured adapter.\n *\n * @param {object} config The config that is to be used for the request\n * @returns {Promise} The Promise to be fulfilled\n */\nmodule.exports = function dispatchRequest(config) {\n throwIfCancellationRequested(config);\n\n // Ensure headers exist\n config.headers = config.headers || {};\n\n // Transform request data\n config.data = transformData(\n config.data,\n config.headers,\n config.transformRequest\n );\n\n // Flatten headers\n config.headers = utils.merge(\n config.headers.common || {},\n config.headers[config.method] || {},\n config.headers\n );\n\n utils.forEach(\n ['delete', 'get', 'head', 'post', 'put', 'patch', 'common'],\n function cleanHeaderConfig(method) {\n delete config.headers[method];\n }\n );\n\n var adapter = config.adapter || defaults.adapter;\n\n return adapter(config).then(function onAdapterResolution(response) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n response.data = transformData(\n response.data,\n response.headers,\n config.transformResponse\n );\n\n return response;\n }, function onAdapterRejection(reason) {\n if (!isCancel(reason)) {\n throwIfCancellationRequested(config);\n\n // Transform response data\n if (reason && reason.response) {\n reason.response.data = transformData(\n reason.response.data,\n reason.response.headers,\n config.transformResponse\n );\n }\n }\n\n return Promise.reject(reason);\n });\n};\n","'use strict';\n\nvar utils = require('../utils');\n\n/**\n * Config-specific merge-function which creates a new config-object\n * by merging two configuration objects together.\n *\n * @param {Object} config1\n * @param {Object} config2\n * @returns {Object} New object resulting from merging config2 to config1\n */\nmodule.exports = function mergeConfig(config1, config2) {\n // eslint-disable-next-line no-param-reassign\n config2 = config2 || {};\n var config = {};\n\n var valueFromConfig2Keys = ['url', 'method', 'data'];\n var mergeDeepPropertiesKeys = ['headers', 'auth', 'proxy', 'params'];\n var defaultToConfig2Keys = [\n 'baseURL', 'transformRequest', 'transformResponse', 'paramsSerializer',\n 'timeout', 'timeoutMessage', 'withCredentials', 'adapter', 'responseType', 'xsrfCookieName',\n 'xsrfHeaderName', 'onUploadProgress', 'onDownloadProgress', 'decompress',\n 'maxContentLength', 'maxBodyLength', 'maxRedirects', 'transport', 'httpAgent',\n 'httpsAgent', 'cancelToken', 'socketPath', 'responseEncoding'\n ];\n var directMergeKeys = ['validateStatus'];\n\n function getMergedValue(target, source) {\n if (utils.isPlainObject(target) && utils.isPlainObject(source)) {\n return utils.merge(target, source);\n } else if (utils.isPlainObject(source)) {\n return utils.merge({}, source);\n } else if (utils.isArray(source)) {\n return source.slice();\n }\n return source;\n }\n\n function mergeDeepProperties(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n }\n\n utils.forEach(valueFromConfig2Keys, function valueFromConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n }\n });\n\n utils.forEach(mergeDeepPropertiesKeys, mergeDeepProperties);\n\n utils.forEach(defaultToConfig2Keys, function defaultToConfig2(prop) {\n if (!utils.isUndefined(config2[prop])) {\n config[prop] = getMergedValue(undefined, config2[prop]);\n } else if (!utils.isUndefined(config1[prop])) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n utils.forEach(directMergeKeys, function merge(prop) {\n if (prop in config2) {\n config[prop] = getMergedValue(config1[prop], config2[prop]);\n } else if (prop in config1) {\n config[prop] = getMergedValue(undefined, config1[prop]);\n }\n });\n\n var axiosKeys = valueFromConfig2Keys\n .concat(mergeDeepPropertiesKeys)\n .concat(defaultToConfig2Keys)\n .concat(directMergeKeys);\n\n var otherKeys = Object\n .keys(config1)\n .concat(Object.keys(config2))\n .filter(function filterAxiosKeys(key) {\n return axiosKeys.indexOf(key) === -1;\n });\n\n utils.forEach(otherKeys, mergeDeepProperties);\n\n return config;\n};\n","'use strict';\n\nvar utils = require('./../utils');\nvar buildURL = require('../helpers/buildURL');\nvar InterceptorManager = require('./InterceptorManager');\nvar dispatchRequest = require('./dispatchRequest');\nvar mergeConfig = require('./mergeConfig');\n\n/**\n * Create a new instance of Axios\n *\n * @param {Object} instanceConfig The default config for the instance\n */\nfunction Axios(instanceConfig) {\n this.defaults = instanceConfig;\n this.interceptors = {\n request: new InterceptorManager(),\n response: new InterceptorManager()\n };\n}\n\n/**\n * Dispatch a request\n *\n * @param {Object} config The config specific for this request (merged with this.defaults)\n */\nAxios.prototype.request = function request(config) {\n /*eslint no-param-reassign:0*/\n // Allow for axios('example/url'[, config]) a la fetch API\n if (typeof config === 'string') {\n config = arguments[1] || {};\n config.url = arguments[0];\n } else {\n config = config || {};\n }\n\n config = mergeConfig(this.defaults, config);\n\n // Set config.method\n if (config.method) {\n config.method = config.method.toLowerCase();\n } else if (this.defaults.method) {\n config.method = this.defaults.method.toLowerCase();\n } else {\n config.method = 'get';\n }\n\n // Hook up interceptors middleware\n var chain = [dispatchRequest, undefined];\n var promise = Promise.resolve(config);\n\n this.interceptors.request.forEach(function unshiftRequestInterceptors(interceptor) {\n chain.unshift(interceptor.fulfilled, interceptor.rejected);\n });\n\n this.interceptors.response.forEach(function pushResponseInterceptors(interceptor) {\n chain.push(interceptor.fulfilled, interceptor.rejected);\n });\n\n while (chain.length) {\n promise = promise.then(chain.shift(), chain.shift());\n }\n\n return promise;\n};\n\nAxios.prototype.getUri = function getUri(config) {\n config = mergeConfig(this.defaults, config);\n return buildURL(config.url, config.params, config.paramsSerializer).replace(/^\\?/, '');\n};\n\n// Provide aliases for supported request methods\nutils.forEach(['delete', 'get', 'head', 'options'], function forEachMethodNoData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: (config || {}).data\n }));\n };\n});\n\nutils.forEach(['post', 'put', 'patch'], function forEachMethodWithData(method) {\n /*eslint func-names:0*/\n Axios.prototype[method] = function(url, data, config) {\n return this.request(mergeConfig(config || {}, {\n method: method,\n url: url,\n data: data\n }));\n };\n});\n\nmodule.exports = Axios;\n","'use strict';\n\n/**\n * A `Cancel` is an object that is thrown when an operation is canceled.\n *\n * @class\n * @param {string=} message The message.\n */\nfunction Cancel(message) {\n this.message = message;\n}\n\nCancel.prototype.toString = function toString() {\n return 'Cancel' + (this.message ? ': ' + this.message : '');\n};\n\nCancel.prototype.__CANCEL__ = true;\n\nmodule.exports = Cancel;\n","'use strict';\n\nvar Cancel = require('./Cancel');\n\n/**\n * A `CancelToken` is an object that can be used to request cancellation of an operation.\n *\n * @class\n * @param {Function} executor The executor function.\n */\nfunction CancelToken(executor) {\n if (typeof executor !== 'function') {\n throw new TypeError('executor must be a function.');\n }\n\n var resolvePromise;\n this.promise = new Promise(function promiseExecutor(resolve) {\n resolvePromise = resolve;\n });\n\n var token = this;\n executor(function cancel(message) {\n if (token.reason) {\n // Cancellation has already been requested\n return;\n }\n\n token.reason = new Cancel(message);\n resolvePromise(token.reason);\n });\n}\n\n/**\n * Throws a `Cancel` if cancellation has been requested.\n */\nCancelToken.prototype.throwIfRequested = function throwIfRequested() {\n if (this.reason) {\n throw this.reason;\n }\n};\n\n/**\n * Returns an object that contains a new `CancelToken` and a function that, when called,\n * cancels the `CancelToken`.\n */\nCancelToken.source = function source() {\n var cancel;\n var token = new CancelToken(function executor(c) {\n cancel = c;\n });\n return {\n token: token,\n cancel: cancel\n };\n};\n\nmodule.exports = CancelToken;\n","'use strict';\n\n/**\n * Syntactic sugar for invoking a function and expanding an array for arguments.\n *\n * Common use case would be to use `Function.prototype.apply`.\n *\n * ```js\n * function f(x, y, z) {}\n * var args = [1, 2, 3];\n * f.apply(null, args);\n * ```\n *\n * With `spread` this example can be re-written.\n *\n * ```js\n * spread(function(x, y, z) {})([1, 2, 3]);\n * ```\n *\n * @param {Function} callback\n * @returns {Function}\n */\nmodule.exports = function spread(callback) {\n return function wrap(arr) {\n return callback.apply(null, arr);\n };\n};\n","'use strict';\n\n/**\n * Determines whether the payload is an error thrown by Axios\n *\n * @param {*} payload The value to test\n * @returns {boolean} True if the payload is an error thrown by Axios, otherwise false\n */\nmodule.exports = function isAxiosError(payload) {\n return (typeof payload === 'object') && (payload.isAxiosError === true);\n};\n","'use strict';\n\nvar utils = require('./utils');\nvar bind = require('./helpers/bind');\nvar Axios = require('./core/Axios');\nvar mergeConfig = require('./core/mergeConfig');\nvar defaults = require('./defaults');\n\n/**\n * Create an instance of Axios\n *\n * @param {Object} defaultConfig The default config for the instance\n * @return {Axios} A new instance of Axios\n */\nfunction createInstance(defaultConfig) {\n var context = new Axios(defaultConfig);\n var instance = bind(Axios.prototype.request, context);\n\n // Copy axios.prototype to instance\n utils.extend(instance, Axios.prototype, context);\n\n // Copy context to instance\n utils.extend(instance, context);\n\n return instance;\n}\n\n// Create the default instance to be exported\nvar axios = createInstance(defaults);\n\n// Expose Axios class to allow class inheritance\naxios.Axios = Axios;\n\n// Factory for creating new instances\naxios.create = function create(instanceConfig) {\n return createInstance(mergeConfig(axios.defaults, instanceConfig));\n};\n\n// Expose Cancel & CancelToken\naxios.Cancel = require('./cancel/Cancel');\naxios.CancelToken = require('./cancel/CancelToken');\naxios.isCancel = require('./cancel/isCancel');\n\n// Expose all/spread\naxios.all = function all(promises) {\n return Promise.all(promises);\n};\naxios.spread = require('./helpers/spread');\n\n// Expose isAxiosError\naxios.isAxiosError = require('./helpers/isAxiosError');\n\nmodule.exports = axios;\n\n// Allow use of default import syntax in TypeScript\nmodule.exports.default = axios;\n","module.exports = require('./lib/axios');","\nimport { setLogger } from '../ulity/debug';\nimport * as debug from '../ulity/debug';\nimport Event from '../ulity/event';\nimport Events from '../base/event';\nimport axios from 'axios';\nimport * as Base from '../base/export';\n\nexport default class RTCEndpoint extends Event\n{\n constructor(options)\n {\n super('RTCPusherPlayer');\n this.TAG = '[RTCPusherPlayer]';\n\n let defaults = {\n element: '',// html video element\n debug: false,// if output debug log\n zlmsdpUrl:'',\n simulcast:false,\n useCamera:true,\n audioEnable:true,\n videoEnable:true,\n recvOnly:false,\n resolution:{w:0,h:0}\n };\n \n this.options = Object.assign({}, defaults, options);\n\n if(this.options.debug)\n {\n setLogger();\n }\n\n this.e = {\n onicecandidate:this._onIceCandidate.bind(this),\n ontrack:this._onTrack.bind(this),\n onicecandidateerror:this._onIceCandidateError.bind(this)\n };\n\n this._remoteStream = null;\n this._localStream = null;\n\n this.pc = new RTCPeerConnection(null);\n\n this.pc.onicecandidate = this.e.onicecandidate;\n this.pc.onicecandidateerror = this.e.onicecandidateerror;\n this.pc.ontrack = this.e.ontrack;\n\n if(!this.options.recvOnly && (this.options.audioEnable || this.options.videoEnable))\n this.start();\n else\n this.receive();\n \n }\n\n receive()\n {\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n //debug.error(this.TAG,'this not implement');\n const AudioTransceiverInit = {\n direction: 'recvonly',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'recvonly',\n sendEncodings:[],\n };\n\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n videoTransceiver = this.pc.addTransceiver('video',VideoTransceiverInit);\n \n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/answer exchange \n this.dispatch(Events.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED,ret);\n return;\n }\n let answer = {};\n answer.sdp = ret.sdp;\n answer.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n\n this.pc.setRemoteDescription(answer).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n }\n\n start()\n {\n let videoConstraints = false;\n let audioConstraints = false;\n\n if(this.options.useCamera)\n {\n if(this.options.videoEnable)\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.CAMERA);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n }\n else\n {\n if(this.options.videoEnable)\n {\n videoConstraints = new Base.VideoTrackConstraints(Base.VideoSourceInfo.SCREENCAST);\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.SCREENCAST);\n }\n else\n {\n if(this.options.audioEnable)\n audioConstraints = new Base.AudioTrackConstraints(Base.AudioSourceInfo.MIC);\n else\n {// error shared display media not only audio\n debug.error(this.TAG,'error paramter');\n }\n }\n \n }\n\n if(this.options.resolution.w !=0 && this.options.resolution.h!=0 && typeof videoConstraints == 'object'){\n videoConstraints.resolution = new Base.Resolution(this.options.resolution.w ,this.options.resolution.h);\n }\n\n Base.MediaStreamFactory.createMediaStream(new Base.StreamConstraints(\n audioConstraints, videoConstraints)).then(stream => {\n\n this._localStream = stream;\n\n this.dispatch(Events.WEBRTC_ON_LOCAL_STREAM,stream);\n\n const AudioTransceiverInit = {\n direction: 'sendrecv',\n sendEncodings:[]\n };\n const VideoTransceiverInit= {\n direction: 'sendrecv',\n sendEncodings:[],\n };\n \n if(this.options.simulcast && stream.getVideoTracks().length>0)\n {\n VideoTransceiverInit.sendEncodings = [\n {rid: 'q', active: true, scaleResolutionDownBy: 4.0},\n {rid: 'h', active: true, scaleResolutionDownBy: 2.0},\n {rid: 'f', active: true}\n ];\n }\n let audioTransceiver = null;\n let videoTransceiver = null;\n\n if(stream.getAudioTracks().length>0)\n {\n audioTransceiver = this.pc.addTransceiver(stream.getAudioTracks()[0],\n AudioTransceiverInit);\n }\n else\n {\n AudioTransceiverInit.direction ='recvonly';\n audioTransceiver = this.pc.addTransceiver('audio',AudioTransceiverInit);\n }\n \n if(stream.getVideoTracks().length>0)\n {\n videoTransceiver = this.pc.addTransceiver(stream.getVideoTracks()[0],\n VideoTransceiverInit);\n }\n else\n {\n VideoTransceiverInit.direction = 'recvonly';\n videoTransceiver = this.pc.addTransceiver('video',\n VideoTransceiverInit);\n }\n\n /*\n stream.getTracks().forEach((track,idx)=>{\n debug.log(this.TAG,track);\n this.pc.addTrack(track);\n });\n */\n this.pc.createOffer().then((desc)=>{\n debug.log(this.TAG,'offer:',desc.sdp);\n this.pc.setLocalDescription(desc).then(() => {\n axios({\n method: 'post',\n url:this.options.zlmsdpUrl,\n responseType:'json',\n data:desc.sdp,\n headers:{\n 'Content-Type':'text/plain;charset=utf-8'\n }\n }).then(response=>{\n let ret = response.data;//JSON.parse(response.data);\n if(ret.code != 0)\n {// mean failed for offer/answer exchange \n this.dispatch(Events.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED,ret);\n return;\n }\n let answer = {};\n answer.sdp = ret.sdp;\n answer.type = 'answer';\n debug.log(this.TAG,'answer:',ret.sdp);\n \n this.pc.setRemoteDescription(answer).then(()=>{\n debug.log(this.TAG,'set remote sucess');\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n });\n });\n }).catch(e=>{\n debug.error(this.TAG,e);\n });\n\n }).catch(e=>{\n this.dispatch(Events.CAPTURE_STREAM_FAILED);\n //debug.error(this.TAG,e);\n });\n \n //const offerOptions = {};\n /*\n if (typeof this.pc.addTransceiver === 'function') {\n // |direction| seems not working on Safari.\n this.pc.addTransceiver('audio', { direction: 'recvonly' });\n this.pc.addTransceiver('video', { direction: 'recvonly' });\n } else {\n offerOptions.offerToReceiveAudio = true;\n offerOptions.offerToReceiveVideo = true;\n }\n */\n\n\n\n }\n _onIceCandidate(event) {\n if (event.candidate) { \n debug.log('Remote ICE candidate: \\n ' + event.candidate.candidate);\n // Send the candidate to the remote peer\n }\n else {\n // All ICE candidates have been sent\n }\n }\n\n _onTrack(event){\n if(this.options.element && event.streams && event.streams.length>0)\n {\n this.options.element.srcObject = event.streams[0];\n this._remoteStream = event.streams[0];\n\n this.dispatch(Events.WEBRTC_ON_REMOTE_STREAMS,event);\n }\n else\n {\n debug.error('element pararm is failed');\n }\n }\n\n _onIceCandidateError(event){\n this.dispatch(Events.WEBRTC_ICE_CANDIDATE_ERROR,event);\n }\n\n close()\n {\n if(this.pc)\n {\n this.pc.close();\n this.pc=null;\n }\n\n if(this.options)\n {\n this.options=null;\n }\n\n if(this._localStream)\n {\n this._localStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n\n if(this._remoteStream)\n {\n this._remoteStream.getTracks().forEach((track,idx)=>{\n track.stop();\n });\n }\n }\n\n get remoteStream()\n {\n return this._remoteStream;\n }\n \n get localStream()\n {\n return this._localStream;\n }\n}\n","import * as mediaformat from './mediaformat';\nimport * as MediaFactory from './mediastream-factory';\n\n\nconst quickScan=[\n {\n 'label': '4K(UHD)',\n 'width': 3840,\n 'height': 2160\n },\n {\n 'label': '1080p(FHD)',\n 'width': 1920,\n 'height': 1080\n },\n {\n 'label': 'UXGA',\n 'width': 1600,\n 'height': 1200,\n 'ratio': '4:3'\n },\n {\n 'label': '720p(HD)',\n 'width': 1280,\n 'height': 720\n },\n {\n 'label': 'SVGA',\n 'width': 800,\n 'height': 600\n },\n {\n 'label': 'VGA',\n 'width': 640,\n 'height': 480\n },\n {\n 'label': '360p(nHD)',\n 'width': 640,\n 'height': 360\n },\n {\n 'label': 'CIF',\n 'width': 352,\n 'height': 288\n },\n {\n 'label': 'QVGA',\n 'width': 320,\n 'height': 240\n },\n {\n 'label': 'QCIF',\n 'width': 176,\n 'height': 144\n },\n {\n 'label': 'QQVGA',\n 'width': 160,\n 'height': 120\n }\n];\n\n\n\n\nexport default function GetSupportCameraResolutions(){\n return new Promise(function (resolve, reject) {\n let resolutions = [];\n let ok = 0;\n let err = 0;\n for (let i = 0; i < quickScan.length; ++i) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(quickScan[i].width, quickScan[i].height);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolutions.push(quickScan[i]);\n ok++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n }).catch(e => {\n err++;\n if(ok+err == quickScan.length)\n {\n resolve(resolutions);\n }\n });\n }\n });\n}\n\nexport function GetAllScanResolution()\n{\n return quickScan;\n}\nexport function isSupportResolution(w,h)\n{\n return new Promise(function (resolve, reject) {\n let videoConstraints = new MediaFactory.VideoTrackConstraints(mediaformat.VideoSourceInfo.CAMERA);\n videoConstraints.resolution = new mediaformat.Resolution(w,h);\n\n MediaFactory.MediaStreamFactory.createMediaStream(new MediaFactory.StreamConstraints(\n false, videoConstraints)).then(stream => {\n resolve();\n }).catch(e => {\n reject(e);\n });\n });\n}","import * as events from './base/event';\nimport * as compile from './ulity/version';\nimport * as media from './base/export';\nimport * as endpoint from './endpoint/endpoint';\nimport * as resolution from './base/resolutionfind';\n\n\n\nconsole.log('build date:',compile.BUILD_DATE);\nconsole.log('version:',compile.VERSION);\n\nexport const Events = events.default;\nexport const Media = media;\nexport const Endpoint = endpoint.default;\nexport const GetSupportCameraResolutions = resolution.default;\nexport const GetAllScanResolution = resolution.GetAllScanResolution;\nexport const isSupportResolution = 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\ No newline at end of file diff --git a/www/webrtc/index.html b/www/webrtc/index.html index 4e88139e..9007d34b 100644 --- a/www/webrtc/index.html +++ b/www/webrtc/index.html @@ -26,8 +26,8 @@

- - + +

@@ -46,14 +46,14 @@

- - push - play + + push + play

- -

@@ -69,7 +69,7 @@ var recvOnly = true var resArr = [] - document.getElementsByName("methond").forEach((el,idx)=>{ + document.getElementsByName("method").forEach((el,idx)=>{ el.onclick=function(e){ if(el.value == "play") { @@ -91,14 +91,14 @@ opt = document.createElement('option'); opt.text = r.label +"("+r.width+"x"+r.height+")"; opt.value = r; - document.getElementById("resilution").add(opt,null) + document.getElementById("resolution").add(opt,null) //console.log(opt.text.match(/\d+/g)) }) function start_play(){ - let elr = document.getElementById("resilution"); + let elr = document.getElementById("resolution"); let res = elr.options[elr.selectedIndex].text.match(/\d+/g); let h = parseInt(res.pop()); let w = parseInt(res.pop()); @@ -108,7 +108,7 @@ element: document.getElementById('video'),// video 标签 debug: true,// 是否打印日志 zlmsdpUrl:document.getElementById('streamUrl').value,//流地址 - simulecast:false,//document.getElementById('simulecast').checked, + simulcast:false,//document.getElementById('simulcast').checked, useCamera:document.getElementById('useCamera').checked, audioEnable:document.getElementById('audioEnable').checked, videoEnable:document.getElementById('videoEnable').checked, @@ -127,9 +127,9 @@ console.log('播放成功',e.streams) }); - player.on(ZLMRTCClient.Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED,function(e) - {// offer anwser 交换失败 - console.log('offer anwser 交换失败',e) + player.on(ZLMRTCClient.Events.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED,function(e) + {// offer answer 交换失败 + console.log('offer answer 交换失败',e) stop(); }); @@ -139,7 +139,7 @@ document.getElementById('selfVideo').srcObject=s; document.getElementById('selfVideo').muted = true; - //console.log('offer anwser 交换失败',e) + //console.log('offer answer 交换失败',e) }); player.on(ZLMRTCClient.Events.CAPTURE_STREAM_FAILED,function(s) @@ -152,7 +152,7 @@ function start() { stop(); - let elr = document.getElementById("resilution"); + let elr = document.getElementById("resolution"); let res = elr.options[elr.selectedIndex].text.match(/\d+/g); let h = parseInt(res.pop()); let w = parseInt(res.pop()); From f5786fcba3af4bb507e759d937e12e23731150bc Mon Sep 17 00:00:00 2001 From: wxf Date: Wed, 23 Jun 2021 10:34:02 +0800 Subject: [PATCH 215/218] =?UTF-8?q?=E5=85=BC=E5=AE=B9=20glibc=20<=202.18?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit * https://stackoverflow.com/a/8132440/5218590 * https://sourceware.org/bugzilla/show_bug.cgi?id=15366 --- webrtc/Sdp.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp index 892a9afb..bcf6acb6 100644 --- a/webrtc/Sdp.cpp +++ b/webrtc/Sdp.cpp @@ -10,7 +10,7 @@ #include "Sdp.h" #include "Rtsp/Rtsp.h" -#include +#include using namespace mediakit; using onCreateSdpItem = function; From 549b07f598406edfc02d884cedb6de4055962666 Mon Sep 17 00:00:00 2001 From: wxf Date: Wed, 23 Jun 2021 09:44:37 +0800 Subject: [PATCH 216/218] spec: Enable WebRTC --- package/rpm/ZLMediaKit.spec | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/package/rpm/ZLMediaKit.spec b/package/rpm/ZLMediaKit.spec index 58b2b633..27685f7d 100644 --- a/package/rpm/ZLMediaKit.spec +++ b/package/rpm/ZLMediaKit.spec @@ -2,10 +2,12 @@ %global use_devtoolset 0 %bcond_without faac %bcond_without x264 +%bcond_without webrtc %else %global use_devtoolset 1 %bcond_with faac %bcond_with x264 +%bcond_with webrtc %endif %bcond_without openssl @@ -37,6 +39,10 @@ BuildRequires: faac-devel BuildRequires: x264-devel %endif +%if %{with webrtc} +BuildRequires: libsrtp-devel >= 2.0 +%endif + %if 0%{?use_devtoolset} BuildRequires: devtoolset-8-gcc-c++ %endif @@ -88,6 +94,7 @@ pushd %{_target_platform} -DENABLE_MYSQL:BOOL=%{with mysql} \ -DENABLE_FAAC:BOOL=%{with faac} \ -DENABLE_X264:BOOL=%{with x264} \ + -DENABLE_WEBRTC:BOOL=%{with webrtc} \ -DENABLE_MP4:BOOL=ON \ -DENABLE_RTPPROXY:BOOL=ON \ -DENABLE_API:BOOL=ON \ From b34226b7858d4991e29329e433f251adfca4337a Mon Sep 17 00:00:00 2001 From: wxf Date: Wed, 23 Jun 2021 10:00:46 +0800 Subject: [PATCH 217/218] spec: WebRTC requires openssl >= 1.1 --- package/rpm/ZLMediaKit.spec | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/package/rpm/ZLMediaKit.spec b/package/rpm/ZLMediaKit.spec index 27685f7d..63297d6c 100644 --- a/package/rpm/ZLMediaKit.spec +++ b/package/rpm/ZLMediaKit.spec @@ -24,8 +24,12 @@ URL: https://github.com/xia-chu/ZLMediaKit Source0: %{name}-%{version}.tar.xz %if %{with openssl} +%if 0%{?rhel} <= 7 && %{with webrtc} +BuildRequires: openssl11-devel +%else BuildRequires: openssl-devel %endif +%endif %if %{with mysql} BuildRequires: mysql-devel From 0d4ac4f6557da5ce69c7de4ebfce0f5f053933e0 Mon Sep 17 00:00:00 2001 From: wxf Date: Wed, 23 Jun 2021 10:11:30 +0800 Subject: [PATCH 218/218] spec: Fix openssl directory --- package/rpm/ZLMediaKit.spec | 3 +++ 1 file changed, 3 insertions(+) diff --git a/package/rpm/ZLMediaKit.spec b/package/rpm/ZLMediaKit.spec index 63297d6c..61c5f86b 100644 --- a/package/rpm/ZLMediaKit.spec +++ b/package/rpm/ZLMediaKit.spec @@ -99,6 +99,9 @@ pushd %{_target_platform} -DENABLE_FAAC:BOOL=%{with faac} \ -DENABLE_X264:BOOL=%{with x264} \ -DENABLE_WEBRTC:BOOL=%{with webrtc} \ +%if %{with webrtc} && 0%{?rhel} <= 7 + -DOPENSSL_ROOT_DIR:STRING="/usr/lib64/openssl11;/usr/include/openssl11" \ +%endif -DENABLE_MP4:BOOL=ON \ -DENABLE_RTPPROXY:BOOL=ON \ -DENABLE_API:BOOL=ON \