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<SockInfo &>(*_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<SockInfo &>(*_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