startSendRtp接口支持rtcp接收超时主动停止

9f0c15a4 · xiongziliang · b7ef7667 · 658271fd · 9f0c15a4 · 9f0c15a4
Commit 9f0c15a4 authored Aug 20, 2022 by xiongziliang
--- a/ZLToolKit @ 658271fd
+++ b/ZLToolKit @ 658271fd
-Subproject commit 4e127f53a93dd1b7665f04a0ec7312d67a34ab6f
+Subproject commit 658271fdf4fb497b0665c06544627789844a003a
--- a/postman/ZLMediaKit.postman_collection.json
+++ b/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
 						}
 					]
 				}

--- a/server/WebApi.cpp
+++ b/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 {

--- a/src/Common/MediaSource.h
+++ b/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

--- a/src/Common/MultiMediaSourceMuxer.cpp
+++ b/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"));

--- a/src/Rtp/RtpSender.cpp
+++ b/src/Rtp/RtpSender.cpp
--- a/src/Rtp/RtpSender.h
+++ b/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