什么是网站改版,招标投标公共服务平台,wordpress优化加速缓存中国,沈阳哪个医院人流好一点RTT指 round-trip time#xff0c;即计算AB两端的往返时延 这里可以分成两个问题#xff1a; 如何在A端估算A和B之间的RTT时间? 如何在B端估算A和B之间的RTT时间? 本文参考资料:rfc 3550rfc 3611webrtc issue https://code.google.com/p/webrtc/issues/detail?id1613以及解…RTT指 round-trip time即计算AB两端的往返时延 这里可以分成两个问题 如何在A端估算A和B之间的RTT时间? 如何在B端估算A和B之间的RTT时间? 本文参考资料:rfc 3550rfc 3611webrtc issue https://code.google.com/p/webrtc/issues/detail?id1613以及解决版本https://code.google.com/p/webrtc/source/detail?r4898https://code.google.com/p/webrtc/source/detail?r5063 一、假设A - B 发送视频. 那么如何在A端估算A-B之间的RTT时间? RFC 3550 http://tools.ietf.org/html/rfc3550#section-6.4.16.4.1 SR: Sender Report RTCP Packet 中描述了如何在发送端计算RTT时间. 大概过程如下: A 发送 SR 包, 并记录SR包的发送时间. 记为send_timeB 接收到 A的SR包后, 记录下最后一次接受到SR包的时间. 记为last_recv_time... (B等待发送rtcp包)B 发送 RR包, 计算从[last_recv_time] 到 当前时间的延时. 记录为delay_since_last_SR. 附加到RR包中. A 收到 B的RR包后, 计算RTT RTT send_time - delay_since_last_SR - last_recv_time 对应到webrtc中的实现. 何时发送rtcp ? ModuleRtpRtcpImpl::Process if (rtcp_sender_.TimeToSendRTCPReport()) { rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); }在RTCPSender::TimeToSendRTCPReport 详细说明了RTCP的发送频率. 每次发送RTCP时都会计算出下一次发送rtcp的时间, 即_nextTimeToSendRTCP. 对于 RR和SR包. 计算如下. RTCPSender::PrepareRTCP.... if( rtcpPacketTypeFlags kRtcpRr || rtcpPacketTypeFlags kRtcpSr) { // generate next time to send a RTCP report // seeded from RTP constructor int32_t random rand() % 1000; int32_t timeToNext RTCP_INTERVAL_AUDIO_MS; if(_audio) { timeToNext (RTCP_INTERVAL_AUDIO_MS/2) (RTCP_INTERVAL_AUDIO_MS*random/1000); }else { uint32_t minIntervalMs RTCP_INTERVAL_AUDIO_MS; if(_sending) { // Calculate bandwidth for video; 360 / send bandwidth in kbit/s. uint32_t send_bitrate_kbit feedback_state.send_bitrate / 1000; if (send_bitrate_kbit ! 0) minIntervalMs 360000 / send_bitrate_kbit; } if(minIntervalMs RTCP_INTERVAL_VIDEO_MS) { minIntervalMs RTCP_INTERVAL_VIDEO_MS; } timeToNext (minIntervalMs/2) (minIntervalMs*random/1000); } _nextTimeToSendRTCP _clock-TimeInMilliseconds() timeToNext; }依赖于随机值, 而且音视频的时间也不同. A - 发送SR包. ModuleRtpRtcpImpl::Process RTCPSender::SendRTCP RTCPSender::PrepareRTCP RTCPSender::BuildSRPrepareRTCP 中通过_sending(是否是发送端) 状态判定发送SR或RR. SR包中含有发送时的NTP时间戳. BuildSR中_lastSendReport 记录NTP时间的中间32位. 可以标识SR包, 也就是B回应RR包中report block的LSR字段(last SR timestamp ), 通过LSR可以查找_lastRTCPTime._lastRTCPTime记录RTCP_NUMBER_OF_SR个数的SR发送时间.这两个数组是一一对应的. _lastRTCPTime[0] Clock::NtpToMs(NTPsec, NTPfrac); _lastSendReport[0] (NTPsec 16) (NTPfrac 16); 最后SendToNetwork. B - 接收到SR包. ModuleRtpRtcpImpl::IncomingRtcpPacket RTCPReceiver::IncomingRTCPPacket RTCPReceiver::HandleSenderReceiverReport在HandleSenderReceiverReport 中保存 SR包中的NTP时间戳 _remoteSenderInfo.NTPseconds rtcpPacket.SR.NTPMostSignificant; _remoteSenderInfo.NTPfraction rtcpPacket.SR.NTPLeastSignificant;并记录SR包接到时的NTP时间戳_clock-CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); B - 发送RR包获取回馈状态, 并发送给A ModuleRtpRtcpImpl::Process() if (rtcp_sender_.TimeToSendRTCPReport()) { rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); } ModuleRtpRtcpImpl::GetFeedbackState() ModuleRtpRtcpImpl::LastReceivedNTP state.last_rr_ntp_secs 和state.last_rr_ntp_frac即为上一次接收到SR包时, 记录的_clock-CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); 时间戳. state.remote_sr 通过_remoteSenderInfo.NTPseconds 和 _remoteSenderInfo.NTPfraction, 取中间32位算出. RTCPSender::PrepareReport 在这里计算延时, 填充到report block中. // get our NTP as late as possible to avoid a race _clock-CurrentNtp(*ntp_secs, *ntp_frac); // Delay since last received report uint32_t delaySinceLastReceivedSR 0; if ((feedback_state.last_rr_ntp_secs ! 0) || (feedback_state.last_rr_ntp_frac ! 0)) { // get the 16 lowest bits of seconds and the 16 higest bits of fractions uint32_t now*ntp_secs0x0000FFFF; now 16; now (*ntp_frac0xffff0000)16; uint32_t receiveTime feedback_state.last_rr_ntp_secs0x0000FFFF; receiveTime 16; receiveTime (feedback_state.last_rr_ntp_frac0xffff0000)16; delaySinceLastReceivedSR now-receiveTime; } report_block-delaySinceLastSR delaySinceLastReceivedSR; report_block-lastSR feedback_state.remote_sr; report_block-delaySinceLastSR 即为 从接到SR包到发送RR包之间的延时. report_block-lastSR 即SR包中NTP时间戳的中间32位. (在A端_lastSendReport数组中记录). A 收到 B的RR包ModuleRtpRtcpImpl::IncomingRtcpPacket RTCPReceiver::IncomingRTCPPacket RTCPReceiver::HandleSenderReceiverReport RTCPReceiver::HandleReportBlock通过 lastSR 到sender模块中取出SR包的发送时间. uint32_t sendTimeMS _rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR); 计算RTT . uint32_t delaySinceLastSendReport rtcpPacket.ReportBlockItem.DelayLastSR; // local NTP time when we received this uint32_t lastReceivedRRNTPsecs 0; uint32_t lastReceivedRRNTPfrac 0; _clock-CurrentNtp(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); // time when we received this in MS uint32_t receiveTimeMS Clock::NtpToMs(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); // Estimate RTT uint32_t d (delaySinceLastSendReport 0x0000ffff) * 1000; d / 65536; d ((delaySinceLastSendReport 0xffff0000) 16) * 1000; int32_t RTT 0; if (sendTimeMS 0) { RTT receiveTimeMS - d - sendTimeMS;.... } 注意delay since last SR (DLSR) 的单位是1/65536秒. 二、另外一个问题, 那么如何在B端估算A和B之间的RTT时间? 如果是互相视频聊天的话, A和B都是发送端, 自然都可以计算出RTT. 但是B如果仅仅是接收者的话, 仅仅依靠RFC3550协议是无法计算RTT时间的.需要参考rfc 3611协议, 实现section4.5 的 DLRR Report Block 即可. http://tools.ietf.org/html/rfc3611#section-4.5 webrtc 在bug 1613 https://code.google.com/p/webrtc/issues/detail?id1613 中讨论该问题. 并在版本 https://code.google.com/p/webrtc/source/detail?r4898和https://code.google.com/p/webrtc/source/detail?r5063 中修复. 具体实现和SR非常类似. 1. 开启XR协议 ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(true) webrtc的示例loopback 程序中可以这样启动 receive_config.rtp.rtcp_xr.receiver_reference_time_report true; 接受者(B)发送RTCP时, 附加kRtcpXrReceiverReferenceTime发送者(A)发送RTCP时, 附加kRtcpXrDlrrReportBlock RTCPSender::PrepareRTCP if (xrSendReceiverReferenceTimeEnabled_ !_sending) { rtcpPacketTypeFlags | kRtcpXrReceiverReferenceTime; } if (feedback_state.has_last_xr_rr) { rtcpPacketTypeFlags | kRtcpXrDlrrReportBlock; } B在发送kRtcpXrReceiverReferenceTime, 在last_xr_rr_ map中记录 NTP时间戳中间32位(key) 和 发送时间(value). A 收到XR_RR包后在处理kRtcpXrReceiverReferenceTimeCodeRTCPReceiver::HandleXrReceiveReferenceTime _remoteXRReceiveTimeInfo.lastRR RTCPUtility::MidNtp( packet.XRReceiverReferenceTimeItem.NTPMostSignificant, packet.XRReceiverReferenceTimeItem.NTPLeastSignificant); _clock-CurrentNtp(_lastReceivedXRNTPsecs, _lastReceivedXRNTPfrac); 记录lastRR和收到XR_RR包的时间. A 发送RTCP时, 会检查是否收到过xr_rr包.ModuleRtpRtcpImpl::GetFeedbackState() state.has_last_xr_rr LastReceivedXrReferenceTimeInfo(state.last_xr_rr); bool RTCPReceiver::LastReceivedXrReferenceTimeInfo( RtcpReceiveTimeInfo* info) const { assert(info); CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if (_lastReceivedXRNTPsecs 0 _lastReceivedXRNTPfrac 0) { return false; } info-sourceSSRC _remoteXRReceiveTimeInfo.sourceSSRC; info-lastRR _remoteXRReceiveTimeInfo.lastRR; // Get the delay since last received report (RFC 3611). uint32_t receive_time RTCPUtility::MidNtp(_lastReceivedXRNTPsecs, _lastReceivedXRNTPfrac); uint32_t ntp_sec 0; uint32_t ntp_frac 0; _clock-CurrentNtp(ntp_sec, ntp_frac); uint32_t now RTCPUtility::MidNtp(ntp_sec, ntp_frac); info-delaySinceLastRR now - receive_time; return true;}计算出 从接到last_xr_rr 到当前的延时. 然后发送 kRtcpXrDlrrReportBlock 出去. B 收到XR_SR后RTCPReceiver::HandleXrDlrrReportBlock计算出RTT时间. 保存在xr_rr_rtt_ms_ rtp_rtcp_impl_unittest.cc 测试程序.TEST_F(RtpRtcpImplTest, RttForReceiverOnly) 转载于:https://www.cnblogs.com/lingdhox/p/5746210.html
