diff --git a/README.md b/README.md
index 73ba90eb..1322dfa5 100644
--- a/README.md
+++ b/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
-
+
- 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
diff --git a/README_en.md b/README_en.md
index a89d0ded..b37074c3 100644
--- a/README_en.md
+++ b/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.
diff --git a/server/WebApi.cpp b/server/WebApi.cpp
index 7d69df6a..aae6380c 100755
--- a/server/WebApi.cpp
+++ b/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();
diff --git a/server/WebHook.cpp b/server/WebHook.cpp
index 1bd47ffb..d1df3308 100755
--- a/server/WebHook.cpp
+++ b/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);
diff --git a/src/Rtsp/RtpReceiver.cpp b/src/Rtsp/RtpReceiver.cpp
index 711ce9e8..c33c2b5a 100644
--- a/src/Rtsp/RtpReceiver.cpp
+++ b/src/Rtsp/RtpReceiver.cpp
@@ -53,12 +53,22 @@ 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];
- return false;
+ 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();
diff --git a/src/Rtsp/RtpReceiver.h b/src/Rtsp/RtpReceiver.h
index 99cba9d3..f8d2fde9 100644
--- a/src/Rtsp/RtpReceiver.h
+++ b/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 _rtp_sortor[2];
};