[webrtc] rtcp模块中rtt时间计算

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 3550
rfc 3611
webrtc issue https://code.google.com/p/webrtc/issues/detail?id=1613
以及解决版本
https://code.google.com/p/webrtc/source/detail?r=4898
https://code.google.com/p/webrtc/source/detail?r=5063

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-6.4.1
6.4.1 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_time
B 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::BuildSR
PrepareRTCP 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_secs&0x0000FFFF;
now <<=16;
now += (*ntp_frac&0xffff0000)>>16;

uint32_t receiveTime = feedback_state.last_rr_ntp_secs&0x0000FFFF;
receiveTime <<=16;
receiveTime += (feedback_state.last_rr_ntp_frac&0xffff0000)>>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?id=1613
中讨论该问题. 并在版本 https://code.google.com/p/webrtc/source/detail?r=4898
和https://code.google.com/p/webrtc/source/detail?r=5063 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kRtcpXrReceiverReferenceTimeCode
RTCPReceiver::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)