Merge branch 'master' of https://gitee.com/xia-chu/ZLMediaKit into dev

README.md

@@ -13,7 +13,7 @@
 ## 项目特点
 
 - 基于C++11开发，避免使用裸指针，代码稳定可靠，性能优越。
-- 支持多种协议(RTSP/RTMP/HLS/HTTP-FLV/WebSocket-FLV/GB28181/HTTP-TS/WebSocket-TS/HTTP-fMP4/WebSocket-fMP4/MP4),支持协议互转。
+- 支持多种协议(RTSP/RTMP/HLS/HTTP-FLV/WebSocket-FLV/GB28181/HTTP-TS/WebSocket-TS/HTTP-fMP4/WebSocket-fMP4/MP4/WebRTC),支持协议互转。
 - 使用多路复用/多线程/异步网络IO模式开发，并发性能优越，支持海量客户端连接。
 - 代码经过长期大量的稳定性、性能测试，已经在线上商用验证已久。
 - 支持linux、macos、ios、android、windows全平台。
@@ -33,7 +33,7 @@
 
 ## 功能清单
 ### 功能一览
-<img width="800" alt="图片" src="https://user-images.githubusercontent.com/11495632/102689561-09824200-423a-11eb-9cf9-b39d1378ef68.png">
+<img width="800" alt="功能一览" src="https://user-images.githubusercontent.com/11495632/114176523-d50fce80-996d-11eb-81f8-0a2e2715ba7b.png">
 
 - RTSP[S]
   - RTSP[S] 服务器，支持RTMP/MP4/HLS转RTSP[S],支持亚马逊echo show这样的设备
@@ -92,6 +92,10 @@
   - RTSP/RTMP/HTTP-FLV/WS-FLV支持MP4文件点播，支持seek
   - 支持H264/H265/AAC/G711/OPUS编码
   
+- WebRTC(体验,请使用dev分支)
+  - 支持WebRTC推流，支持转其他协议
+  - 支持WebRTC播放，支持其他协议转WebRTC     
+  
 - 其他
   - 支持丰富的restful api以及web hook事件 
   - 支持简单的telnet调试

README_en.md

@@ -11,7 +11,7 @@
 
 ## Why ZLMediaKit?
 - Developed based on C++ 11, the code is stable and reliable, avoiding the use of raw pointers, cross-platform porting is simple and convenient, and the code is clear and concise.
-- Support rich streaming media protocols(`RTSP/RTMP/HLS/HTTP-FLV/WebSocket-flv/HTTP-TS/WebSocket-TS/HTTP-fMP4/Websocket-fMP4/MP4`),and support Inter-protocol conversion.
+- Support rich streaming media protocols(`RTSP/RTMP/HLS/HTTP-FLV/WebSocket-flv/HTTP-TS/WebSocket-TS/HTTP-fMP4/Websocket-fMP4/MP4/WebRTC`),and support Inter-protocol conversion.
 - Multiplexing asynchronous network IO based on epoll and multi thread，extreme performance.
 - Well performance and stable test,can be used commercially.
 - Support linux, macos, ios, android, Windows Platforms.
@@ -55,6 +55,10 @@
   - WebSocket Server and Client.
   - File access authentication.
   
+- WebRTC(experiential, dev branch)
+  - Support webrtc push stream and transfer to other protocols
+  - Support webrtc play, support other protocol to webrtc
+
 - Others
   - Support stream proxy by ffmpeg.
   - RESTful http api and http hook event api.

server/WebApi.cpp

@@ -303,7 +303,7 @@ Value makeMediaSourceJson(MediaSource &media){
         item["originSock"] = Json::nullValue;
     }
 
-    for(auto &track : media.getTracks()){
+    for(auto &track : media.getTracks(false)){
         Value obj;
         auto codec_type = track->getTrackType();
         obj["codec_id"] = track->getCodecId();

server/WebHook.cpp

@@ -301,14 +301,15 @@ void installWebHook(){
             return;
         }
         ArgsType body;
-        body["regist"] = bRegist;
         if (bRegist) {
             body = makeMediaSourceJson(sender);
+            body["regist"] = bRegist;
         } else {
             body["schema"] = sender.getSchema();
             body["vhost"] = sender.getVhost();
             body["app"] = sender.getApp();
             body["stream"] = sender.getId();
+            body["regist"] = bRegist;
         }
         //执行hook
         do_http_hook(hook_stream_chaned,body, nullptr);

src/Rtsp/RtpReceiver.cpp

@@ -53,13 +53,23 @@ bool RtpReceiver::handleOneRtp(int index, TrackType type, int sample_rate, uint8
     auto ssrc = ntohl(header->ssrc);
 
     if (!_ssrc[index]) {
-        //保存SSRC至track对象
+        //记录并锁定ssrc
         _ssrc[index] = ssrc;
-    } else if (_ssrc[index] != ssrc) {
+        _ssrc_alive[index].resetTime();
+    } else if (_ssrc[index] == ssrc) {
+        //ssrc匹配正确,刷新计时器
+        _ssrc_alive[index].resetTime();
+    } else {
         //ssrc错误
-        WarnL << "ssrc错误:" << ssrc << " != " << _ssrc[index];
+        if (_ssrc_alive[index].elapsedTime() < 10 * 1000) {
+            //接受正确ssrc的rtp在10秒内，那么我们认为存在多路rtp,忽略掉ssrc不匹配的rtp
+            WarnL << "ssrc比匹配,rtp已丢弃:" << ssrc << " != " << _ssrc[index];
             return false;
         }
+        InfoL << "rtp流ssrc切换:" << _ssrc[index] << " -> " << ssrc;
+        _ssrc[index] = ssrc;
+        _ssrc_alive[index].resetTime();
+    }
 
     auto rtp = RtpPacket::create();
     //需要添加4个字节的rtp over tcp头

src/Rtsp/RtpReceiver.h

@@ -198,6 +198,7 @@ protected:
 
 private:
     uint32_t _ssrc[2] = {0, 0};
+    Ticker _ssrc_alive[2];
     //rtp排序缓存，根据seq排序
     PacketSortor<RtpPacket::Ptr> _rtp_sortor[2];
 };