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https://github.com/ZLMediaKit/ZLMediaKit.git
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46842e6f29
因WebRtcPlayer中使用RtspMediaSource的共享指针，特定情况下引起媒体注销无法触发的问题。
- 重现步骤
在ZL的webrtc demo页面推流
浏览器打开如下html
webrtc.html
关闭推流器页面,推流器停止推流
webrtc.htm浏览器console->network将观察到：即使推流停止，但webrtc sdp请求一直能成功获取sdp，且流媒体一直不注销
- 原因
因为每个WebRtc 播放 SDP请求都会产生 WebRtcPlayer，产生RtspMediaSource的共享指针，产生强引用。
而DTLS超时释放需要一定的时间，WebRtcPlayer销毁需要超时。如果请求sdp的时间足够短，强引用会一直存在。将永远无法触发媒体注销
- 场景
webrtc播放存在重试，但是udp不通。DTLS无法创建
有人对ZLM执行恶意攻击，短时间内不断请求SDP但是不建立WebRTC通信
101 lines
3.7 KiB
C++
101 lines
3.7 KiB
C++
/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include "WebRtcPlayer.h"
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#include "Common/config.h"
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using namespace std;
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namespace mediakit {
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WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info,
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bool preferred_tcp) {
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WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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ret->onCreate();
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return ret;
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}
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WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info,
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bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
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_media_info = info;
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_play_src = src;
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CHECK(src);
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}
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void WebRtcPlayer::onStartWebRTC() {
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auto playSrc = _play_src.lock();
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if(!playSrc){
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onShutdown(SockException(Err_shutdown, "rtsp media source was shutdown"));
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return ;
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}
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WebRtcTransportImp::onStartWebRTC();
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if (canSendRtp()) {
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playSrc->pause(false);
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_reader = playSrc->getRing()->attach(getPoller(), true);
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weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
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weak_ptr<Session> weak_session = getSession();
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_reader->setGetInfoCB([weak_session]() { return weak_session.lock(); });
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_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
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auto strong_self = weak_self.lock();
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if (!strong_self) {
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return;
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}
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size_t i = 0;
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
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strong_self->onSendRtp(rtp, ++i == pkt->size());
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});
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});
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_reader->setDetachCB([weak_self]() {
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auto strong_self = weak_self.lock();
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if (!strong_self) {
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return;
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}
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strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
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});
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}
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}
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void WebRtcPlayer::onDestory() {
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WebRtcTransportImp::onDestory();
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auto duration = getDuration();
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auto bytes_usage = getBytesUsage();
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//流量统计事件广播
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GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
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if (_reader && getSession()) {
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WarnL << "RTC播放器("
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<< _media_info.shortUrl()
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<< ")结束播放,耗时(s):" << duration;
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if (bytes_usage >= iFlowThreshold * 1024) {
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NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration,
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true, static_cast<SockInfo &>(*getSession()));
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}
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}
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}
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void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
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auto playSrc = _play_src.lock();
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if(!playSrc){
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return ;
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}
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WebRtcTransportImp::onRtcConfigure(configure);
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//这是播放
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configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
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configure.setPlayRtspInfo(playSrc->getSdp());
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}
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}// namespace mediakit
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