mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-26 04:31:37 +08:00
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++
/*
|
|
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
|
|
*
|
|
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
|
|
*
|
|
* Use of this source code is governed by MIT license that can be found in the
|
|
* LICENSE file in the root of the source tree. All contributing project authors
|
|
* may be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "WebRtcPlayer.h"
|
|
#include "Common/config.h"
|
|
|
|
using namespace std;
|
|
|
|
namespace mediakit {
|
|
|
|
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info,
|
|
bool preferred_tcp) {
|
|
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
|
|
ptr->onDestory();
|
|
delete ptr;
|
|
});
|
|
ret->onCreate();
|
|
return ret;
|
|
}
|
|
|
|
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info,
|
|
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
|
|
_media_info = info;
|
|
_play_src = src;
|
|
CHECK(src);
|
|
}
|
|
|
|
void WebRtcPlayer::onStartWebRTC() {
|
|
auto playSrc = _play_src.lock();
|
|
if(!playSrc){
|
|
onShutdown(SockException(Err_shutdown, "rtsp media source was shutdown"));
|
|
return ;
|
|
}
|
|
WebRtcTransportImp::onStartWebRTC();
|
|
if (canSendRtp()) {
|
|
playSrc->pause(false);
|
|
_reader = playSrc->getRing()->attach(getPoller(), true);
|
|
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
|
|
weak_ptr<Session> weak_session = getSession();
|
|
_reader->setGetInfoCB([weak_session]() { return weak_session.lock(); });
|
|
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
|
|
auto strong_self = weak_self.lock();
|
|
if (!strong_self) {
|
|
return;
|
|
}
|
|
size_t i = 0;
|
|
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
|
|
//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
|
|
strong_self->onSendRtp(rtp, ++i == pkt->size());
|
|
});
|
|
});
|
|
_reader->setDetachCB([weak_self]() {
|
|
auto strong_self = weak_self.lock();
|
|
if (!strong_self) {
|
|
return;
|
|
}
|
|
strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
|
|
});
|
|
}
|
|
}
|
|
void WebRtcPlayer::onDestory() {
|
|
WebRtcTransportImp::onDestory();
|
|
|
|
auto duration = getDuration();
|
|
auto bytes_usage = getBytesUsage();
|
|
//流量统计事件广播
|
|
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
|
|
if (_reader && getSession()) {
|
|
WarnL << "RTC播放器("
|
|
<< _media_info.shortUrl()
|
|
<< ")结束播放,耗时(s):" << duration;
|
|
if (bytes_usage >= iFlowThreshold * 1024) {
|
|
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration,
|
|
true, static_cast<SockInfo &>(*getSession()));
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
|
|
auto playSrc = _play_src.lock();
|
|
if(!playSrc){
|
|
return ;
|
|
}
|
|
WebRtcTransportImp::onRtcConfigure(configure);
|
|
//这是播放
|
|
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
|
|
configure.setPlayRtspInfo(playSrc->getSdp());
|
|
}
|
|
|
|
}// namespace mediakit
|