startSendRtp接口支持rtcp接收超时主动停止

ZLToolKit @ 658271fd

-Subproject commit 4e127f53a93dd1b7665f04a0ec7312d67a34ab6f
+Subproject commit 658271fdf4fb497b0665c06544627789844a003a

postman/ZLMediaKit.postman_collection.json

@@ -1562,6 +1562,12 @@
 							"value": "1",
 							"description": "rtp es方式打包时，是否只打包音频，该参数非必选参数",
 							"disabled": true
+						},
+						{
+							"key": "udp_rtcp_timeout",
+							"value": "0",
+							"description": "udp方式推流时，是否开启rtcp发送和rtcp接收超时判断，开启后(默认关闭)，如果接收rr rtcp超时，将导致主动停止rtp发送",
+							"disabled": true
 						}
 					]
 				}

server/WebApi.cpp

@@ -1147,7 +1147,8 @@ void installWebApi() {
         args.src_port = allArgs["src_port"];
         args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
         args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
-        args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
+        args.only_audio = allArgs["only_audio"].as<bool>();
+        args.udp_rtcp_timeout = allArgs["udp_rtcp_timeout"];
         TraceL << "startSendRtp, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
 
         src->getOwnerPoller()->async([=]() mutable {
@@ -1178,7 +1179,7 @@ void installWebApi() {
         args.src_port = allArgs["src_port"];
         args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
         args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
-        args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
+        args.only_audio = allArgs["only_audio"].as<bool>();
         TraceL << "startSendRtpPassive, pt " << int(args.pt) << " ps " << args.use_ps << " audio " <<  args.only_audio;
 
         src->getOwnerPoller()->async([=]() mutable {

src/Common/MediaSource.h

@@ -111,6 +111,15 @@ public:
         uint16_t dst_port;
         // 发送目标主机地址，可以是ip或域名
         std::string dst_url;
+
+        //udp发送时，是否开启rr rtcp接收超时判断
+        bool udp_rtcp_timeout = false;
+        //tcp被动发送服务器延时关闭事件，单位毫秒
+        uint32_t tcp_passive_close_delay_ms = 5 * 1000;
+        //udp 发送时，rr rtcp包接收超时时间，单位毫秒
+        uint32_t rtcp_timeout_ms = 30 * 1000;
+        //udp 发送时，发送sr rtcp包间隔，单位毫秒
+        uint32_t rtcp_send_interval_ms = 5 * 1000;
     };
 
     // 开始发送ps-rtp

src/Common/MultiMediaSourceMuxer.cpp

@@ -223,7 +223,7 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const MediaSourceEvent::
 #if defined(ENABLE_RTPPROXY)
     auto rtp_sender = std::make_shared<RtpSender>();
     weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
-    rtp_sender->startSend(args, [args, weak_self, rtp_sender, cb](uint16_t local_port, const SockException &ex) {
+    rtp_sender->startSend(args, [args, weak_self, rtp_sender, cb](uint16_t local_port, const SockException &ex) mutable {
         cb(local_port, ex);
         auto strong_self = weak_self.lock();
         if (!strong_self || ex) {
@@ -233,7 +233,15 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const MediaSourceEvent::
             rtp_sender->addTrack(track);
         }
         rtp_sender->addTrackCompleted();
-        strong_self->_rtp_sender[args.ssrc] = rtp_sender;
+
+        auto ssrc = args.ssrc;
+        rtp_sender->setOnClose([weak_self, ssrc]() {
+            if (auto strong_self = weak_self.lock()) {
+                strong_self->_rtp_sender.erase(ssrc);
+                WarnL << "stream:" << strong_self->_get_origin_url() << " stop send rtp:" << ssrc;
+            }
+        });
+        strong_self->_rtp_sender[args.ssrc] = std::move(rtp_sender);
     });
 #else
     cb(0, SockException(Err_other, "该功能未启用，编译时请打开ENABLE_RTPPROXY宏"));

src/Rtp/RtpSender.h

@@ -13,6 +13,7 @@
 #if defined(ENABLE_RTPPROXY)
 #include "PSEncoder.h"
 #include "Extension/CommonRtp.h"
+#include "Rtcp/RtcpContext.h"
 
 namespace mediakit{
 
@@ -53,6 +54,8 @@ public:
      */
     virtual void resetTracks() override;
 
+    void setOnClose(std::function<void()> on_close);
+
 private:
     //合并写输出
     void onFlushRtpList(std::shared_ptr<toolkit::List<toolkit::Buffer::Ptr> > rtp_list);
@@ -60,14 +63,23 @@ private:
     void onConnect();
     //异常断开socket事件
     void onErr(const toolkit::SockException &ex, bool is_connect = false);
+    void createRtcpSocket();
+    void onRecvRtcp(RtcpHeader *rtcp);
+    void onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check);
+    void onClose();
 
 private:
     bool _is_connect = false;
     MediaSourceEvent::SendRtpArgs _args;
-    toolkit::Socket::Ptr _socket;
+    toolkit::Socket::Ptr _socket_rtp;
+    toolkit::Socket::Ptr _socket_rtcp;
     toolkit::EventPoller::Ptr _poller;
     toolkit::Timer::Ptr _connect_timer;
     MediaSinkInterface::Ptr _interface;
+    std::shared_ptr<RtcpContext> _rtcp_context;
+    toolkit::Ticker _rtcp_send_ticker;
+    toolkit::Ticker _rtcp_recv_ticker;
+    std::function<void()> _on_close;
 };
 
 }//namespace mediakit