RTT指 round-trip time，即計算AB兩端的往返時延
這里可以分成兩個問題：
如何在A端估算A和B之間的RTT時間?
如何在B端估算A和B之間的RTT時間?
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本文參考資料:
rfc 3550
rfc 3611
webrtc issue https://code.google.com/p/webrtc/issues/detail?id=1613
以及解決版本
https://code.google.com/p/webrtc/source/detail?r=4898
https://code.google.com/p/webrtc/source/detail?r=5063
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一、假設A -> B 發送視頻. 那么如何在A端估算A->B之間的RTT時間?
RFC 3550 http://tools.ietf.org/html/rfc3550#section-6.4.1
6.4.1 SR: Sender Report RTCP Packet
中描述了如何在發送端計算RTT時間.
大概過程如下:
A 發送 SR 包, 并記錄SR包的發送時間. 記為send_time
B 接收到 A的SR包后, 記錄下最后一次接受到SR包的時間. 記為last_recv_time
... (B等待發送rtcp包)
B 發送 RR包, 計算從[last_recv_time] 到 當前時間的延時. 記錄為delay_since_last_SR. 附加到RR包中.
A 收到 B的RR包后, 計算RTT
RTT = send_time - delay_since_last_SR - last_recv_time
對應到webrtc中的實現.
何時發送rtcp ?
ModuleRtpRtcpImpl::Process
if (rtcp_sender_.TimeToSendRTCPReport()) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
}
在RTCPSender::TimeToSendRTCPReport 詳細說明了RTCP的發送頻率.
每次發送RTCP時都會計算出下一次發送rtcp的時間, 即_nextTimeToSendRTCP.
對于 RR和SR包. 計算如下.
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;
}
依賴于隨機值, 而且音視頻的時間也不同.
A -> 發送SR包.
ModuleRtpRtcpImpl::Process
RTCPSender::SendRTCP
RTCPSender::PrepareRTCP
RTCPSender::BuildSR
PrepareRTCP 中通過_sending(是否是發送端) 狀態判定發送SR或RR. SR包中含有發送時的NTP時間戳.
BuildSR中
_lastSendReport 記錄NTP時間的中間32位. 可以標識SR包, 也就是B回應RR包中report block的LSR字段(last SR timestamp ), 通過LSR可以查找_lastRTCPTime.
_lastRTCPTime記錄RTCP_NUMBER_OF_SR個數的SR發送時間.
這兩個數組是一一對應的.
_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時間戳
_remoteSenderInfo.NTPseconds = rtcpPacket.SR.NTPMostSignificant;
_remoteSenderInfo.NTPfraction = rtcpPacket.SR.NTPLeastSignificant;
并記錄SR包接到時的NTP時間戳
_clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
B -> 發送RR包
獲取回饋狀態, 并發送給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
即為上一次接收到SR包時, 記錄的_clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); 時間戳.
state.remote_sr 通過_remoteSenderInfo.NTPseconds 和 _remoteSenderInfo.NTPfraction, 取中間32位算出.
RTCPSender::PrepareReport
在這里計算延時, 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 即為 從接到SR包到發送RR包之間的延時.
report_block->lastSR 即SR包中NTP時間戳的中間32位. (在A端_lastSendReport數組中記錄).
A 收到 B的RR包
ModuleRtpRtcpImpl::IncomingRtcpPacket
RTCPReceiver::IncomingRTCPPacket
RTCPReceiver::HandleSenderReceiverReport
RTCPReceiver::HandleReportBlock
通過 lastSR 到sender模塊中取出SR包的發送時間.
uint32_t sendTimeMS =
_rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR);
計算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) 的單位是1/65536秒.
二、另外一個問題, 那么如何在B端估算A和B之間的RTT時間?
如果是互相視頻聊天的話, A和B都是發送端, 自然都可以計算出RTT.
但是B如果僅僅是接收者的話, 僅僅依靠RFC3550協議是無法計算RTT時間的.
需要參考rfc 3611協議, 實現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
中討論該問題. 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 中修復.
具體實現和SR非常類似.
1. 開啟XR協議 ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(true)
webrtc的示例loopback 程序中可以這樣啟動
receive_config.rtp.rtcp_xr.receiver_reference_time_report = true;
接受者(B)發送RTCP時, 附加kRtcpXrReceiverReferenceTime
發送者(A)發送RTCP時, 附加kRtcpXrDlrrReportBlock
RTCPSender::PrepareRTCP
if (xrSendReceiverReferenceTimeEnabled_ && !_sending)
{
rtcpPacketTypeFlags |= kRtcpXrReceiverReferenceTime;
}
if (feedback_state.has_last_xr_rr)
{
rtcpPacketTypeFlags |= kRtcpXrDlrrReportBlock;
}
B在發送kRtcpXrReceiverReferenceTime, 在last_xr_rr_ map中記錄 NTP時間戳中間32位(key) 和 發送時間(value).
A 收到XR_RR包后
在處理kRtcpXrReceiverReferenceTimeCode
RTCPReceiver::HandleXrReceiveReferenceTime
_remoteXRReceiveTimeInfo.lastRR = RTCPUtility::MidNtp(
packet.XRReceiverReferenceTimeItem.NTPMostSignificant,
packet.XRReceiverReferenceTimeItem.NTPLeastSignificant);
_clock->CurrentNtp(_lastReceivedXRNTPsecs, _lastReceivedXRNTPfrac);
記錄lastRR和收到XR_RR包的時間.
A 發送RTCP時, 會檢查是否收到過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;
}
計算出 從接到last_xr_rr 到當前的延時.
然后發送 kRtcpXrDlrrReportBlock 出去.
B 收到XR_SR后
RTCPReceiver::HandleXrDlrrReportBlock
計算出RTT時間. 保存在xr_rr_rtt_ms_
rtp_rtcp_impl_unittest.cc 測試程序.
TEST_F(RtpRtcpImplTest, RttForReceiverOnly)
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