webrtc音视频及流媒体WebRTC

WebRTC RTP/RTCP协议分析(一)

2020-05-17  本文已影响0人  JeffreyLau

1 rtp/rtcp通道创建流程

RtpTransportInternal为webrtc 网络rtp以及rtcp传输层的接口,rtp和rtcp数据经该api层讲数据发送到PacketTransportInternal它们之间的关系如下图:

RtpTranspor.png

1.1 Transport的创建

Transport_create.png

1.2 RtpTransportInternal的初始化

// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
#pc/peer_connection.cc
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
    const std::string& mid) {
  RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
  MediaTransportConfig media_transport_config =
      transport_controller_->GetMediaTransportConfig(mid);

  cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
      call_ptr_, configuration_.media_config, rtp_transport,
      media_transport_config, signaling_thread(), mid, SrtpRequired(),
      GetCryptoOptions(), &ssrc_generator_, video_options_,
      video_bitrate_allocator_factory_.get());
  if (!video_channel) {
    return nullptr;
  }
  video_channel->SignalDtlsSrtpSetupFailure.connect(
      this, &PeerConnection::OnDtlsSrtpSetupFailure);
  video_channel->SignalSentPacket.connect(this,
                                          &PeerConnection::OnSentPacket_w);
  video_channel->SetRtpTransport(rtp_transport);

  return video_channel;
}

#pc/channel.cc
bool BaseChannel::ConnectToRtpTransport() {
  RTC_DCHECK(rtp_transport_);
  if (!RegisterRtpDemuxerSink()) {
    return false;
  }
  rtp_transport_->SignalReadyToSend.connect(
      this, &BaseChannel::OnTransportReadyToSend);
  rtp_transport_->SignalRtcpPacketReceived.connect(
      this, &BaseChannel::OnRtcpPacketReceived);

  // TODO(bugs.webrtc.org/9719): Media transport should also be used to provide
  // 'writable' state here.
  rtp_transport_->SignalWritableState.connect(this,
                                              &BaseChannel::OnWritableState);
  rtp_transport_->SignalSentPacket.connect(this,
                                           &BaseChannel::SignalSentPacket_n);
  return true;
}

#pc/rtp_transport_internal.h
class RtpTransportInternal : public sigslot::has_slots<> {
 public:
  virtual ~RtpTransportInternal() = default;

  // Called whenever a transport's ready-to-send state changes. The argument
  // is true if all used transports are ready to send. This is more specific
  // than just "writable"; it means the last send didn't return ENOTCONN.
  sigslot::signal1<bool> SignalReadyToSend;

  // Called whenever an RTCP packet is received. There is no equivalent signal
  // for RTP packets because they would be forwarded to the BaseChannel through
  // the RtpDemuxer callback.
  sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;

  // Called whenever a transport's writable state might change. The argument is
  // true if the transport is writable, otherwise it is false.
  sigslot::signal1<bool> SignalWritableState;

  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;

  virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
                                      RtpPacketSinkInterface* sink) = 0;

  virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
}

// This class represents a receiver of already parsed RTP packets.
#call/rtp_packet_sink_interface.h
class RtpPacketSinkInterface {
 public:
  virtual ~RtpPacketSinkInterface() = default;
  virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
};

#pc/channel.cc
class BaseChannel : public sigslot::has_slots<>,
                    public webrtc::RtpPacketSinkInterface{
                        
  // RtpPacketSinkInterface overrides.
  void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
                        
};

#pc/channel.cc
bool BaseChannel::RegisterRtpDemuxerSink() {
  RTC_DCHECK(rtp_transport_);
  return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this] {
    return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
  });
}

#pc/rtp_transport.h
bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
                                          RtpPacketSinkInterface* sink) {
  rtp_demuxer_.RemoveSink(sink);
  if (!rtp_demuxer_.AddSink(criteria, sink)) {
    RTC_LOG(LS_ERROR) << "Failed to register the sink for RTP demuxer.";
    return false;
  }
  return true;
}

2 rtp/rtcp数据发送

2.1 rtp包的数据发送流程

2.2 rtcp包的数据发送流程

#modules/rtp_rtcp/source/rtp_rtcp_impl.cc
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
  return std::max<int64_t>(0,
                           next_process_time_ - clock_->TimeInMilliseconds());
}

#modules/rtp_rtcp/source/rtp_rtcp_impl.cc
// Process any pending tasks such as timeouts (non time critical events).
void ModuleRtpRtcpImpl::Process() {
  const int64_t now = clock_->TimeInMilliseconds();
  next_process_time_ = now + kRtpRtcpMaxIdleTimeProcessMs;
  .......省略
  if (rtcp_sender_.TimeToSendRTCPReport())
    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);

  if (TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) {
    rtcp_receiver_.NotifyTmmbrUpdated();
  }
}

#modules/rtp_rtcp/source/rtp_rtcp_impl.cc
// TODO(pbos): Handle media and RTX streams separately (separate RTCP
// feedbacks).
RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
  RTCPSender::FeedbackState state;
  // This is called also when receiver_only is true. Hence below
  // checks that rtp_sender_ exists.
  if (rtp_sender_) {
    StreamDataCounters rtp_stats;
    StreamDataCounters rtx_stats;
    rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
    state.packets_sent =
        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
    state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
                             rtx_stats.transmitted.payload_bytes;
    state.send_bitrate = rtp_sender_->BitrateSent();
  }
  state.module = this;

  LastReceivedNTP(&state.last_rr_ntp_secs, &state.last_rr_ntp_frac,
                  &state.remote_sr);

  state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();

  return state;
}

#modules/rtp_rtcp/source/rtp_rtcp_impl.cc
bool ModuleRtpRtcpImpl::LastReceivedNTP(
    uint32_t* rtcp_arrival_time_secs,  // When we got the last report.
    uint32_t* rtcp_arrival_time_frac,
    uint32_t* remote_sr) const {
  // Remote SR: NTP inside the last received (mid 16 bits from sec and frac).
  uint32_t ntp_secs = 0;
  uint32_t ntp_frac = 0;

  if (!rtcp_receiver_.NTP(&ntp_secs, &ntp_frac, rtcp_arrival_time_secs,
                          rtcp_arrival_time_frac, NULL)) {
    return false;
  }
  *remote_sr =
      ((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16);
  return true;
}

int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                             RTCPPacketType packetType,
                             int32_t nack_size,
                             const uint16_t* nack_list) {
  return SendCompoundRTCP(
      feedback_state, std::set<RTCPPacketType>(&packetType, &packetType + 1),
      nack_size, nack_list);
}

int32_t RTCPSender::SendCompoundRTCP(
    const FeedbackState& feedback_state,
    const std::set<RTCPPacketType>& packet_types,
    int32_t nack_size,
    const uint16_t* nack_list) {
  PacketContainer container(transport_, event_log_);
  size_t max_packet_size;

  {
    rtc::CritScope lock(&critical_section_rtcp_sender_);
      
    // Add all flags as volatile. Non volatile entries will not be overwritten.
    // All new volatile flags added will be consumed by the end of this call.
    SetFlags(packet_types, true);
    .....
    // We need to send our NTP even if we haven't received any reports.
    RtcpContext context(feedback_state, nack_size, nack_list,
                        clock_->TimeInMicroseconds());
    
    PrepareReport(feedback_state);
    
    std::unique_ptr<rtcp::RtcpPacket> packet_bye;
    
    auto it = report_flags_.begin();
    while (it != report_flags_.end()) {
      auto builder_it = builders_.find(it->type);
      RTC_DCHECK(builder_it != builders_.end())
          << "Could not find builder for packet type " << it->type;
      if (it->is_volatile) {
        report_flags_.erase(it++);
      } else {
        ++it;
      }
    
      BuilderFunc func = builder_it->second;
      std::unique_ptr<rtcp::RtcpPacket> packet = (this->*func)(context);
      if (packet == nullptr)
        return -1;
      // If there is a BYE, don't append now - save it and append it
      // at the end later.
      if (builder_it->first == kRtcpBye) {
        packet_bye = std::move(packet);
      } else {
        container.Append(packet.release());
      }
    }
    
    // Append the BYE now at the end
    if (packet_bye) {
      container.Append(packet_bye.release());
    }
    
    if (packet_type_counter_observer_ != nullptr) {
      packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
          remote_ssrc_, packet_type_counter_);
    }
    
    RTC_DCHECK(AllVolatileFlagsConsumed());
    max_packet_size = max_packet_size_;

  }

  size_t bytes_sent = container.SendPackets(max_packet_size);
  return bytes_sent == 0 ? -1 : 0;
}

协议地址:
http://tools.ietf.org/html/rfc3550#section-6.4.1
rtcp_sr.png
Fields explain
NTP timestamp 64 bits 网络时间戳,用于不同源之间的同步,如音频和视频
RTP timestamp 32 bits RTP包发送的相对时间戳,该frame从编码器中编码出来的时间
sender's packet count 32 bits 发送者总发送的包数,SSRC发生变化时会被重置
sender's octet count 32 bits 发送者总发送的字节数=每个包的字节数X总的发包数
Fields Explain
SSRC n 32 bits, source identifier，接收到的每个媒体源,如音频和视频,n表示第几个
fraction lost: 8 bits 上一次报告之后到本次报告之间的丢包比列
number of packets lost 24 bits 自接收开始丢包总数,迟到的包不算
highest sequence number 32 bits 低16位表示收到的最大seq,高16位表示seq的循环次数
interarrival jitter 32 bits 估算的RTP包到达时间间隔的统计方差,延迟大小
last SR timestamp (LSR) 32 bits 上一次接收到的SR的NTP时间戳(remote ntp time),取值为:ntp_msw&0xffff + ntp_lsw>>16（取ntp_msw的低16位和ntp_lsw的高16位）
delay since last SR (DLSR) 32 bits 从接收到上一个SR包到发送此接收报告块之间的延时,以1/65536秒为单位. 
std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
  // Timestamp shouldn't be estimated before first media frame.
  RTC_DCHECK_GE(last_frame_capture_time_ms_, 0);
  // The timestamp of this RTCP packet should be estimated as the timestamp of
  // the frame being captured at this moment. We are calculating that
  // timestamp as the last frame's timestamp + the time since the last frame
  // was captured.
  int rtp_rate = rtp_clock_rates_khz_[last_payload_type_];
  if (rtp_rate <= 0) {
    rtp_rate =
        (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) /
        1000;
  }
  // Round now_us_ to the closest millisecond, because Ntp time is rounded
  // when converted to milliseconds,
  // 同一帧数据的rtp_timestamp的时间戳是一样的
  uint32_t rtp_timestamp =
      timestamp_offset_ + last_rtp_timestamp_ +
      ((ctx.now_us_ + 500) / 1000 - last_frame_capture_time_ms_) * rtp_rate;

  rtcp::SenderReport* report = new rtcp::SenderReport();
  report->SetSenderSsrc(ssrc_);
  report->SetNtp(TimeMicrosToNtp(ctx.now_us_));
  report->SetRtpTimestamp(rtp_timestamp);
  report->SetPacketCount(ctx.feedback_state_.packets_sent);
  report->SetOctetCount(ctx.feedback_state_.media_bytes_sent);
  report->SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));

  return std::unique_ptr<rtcp::RtcpPacket>(report);
}

std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
    const FeedbackState& feedback_state) {
  std::vector<rtcp::ReportBlock> result;
  if (!receive_statistics_)
    return result;

  // TODO(danilchap): Support sending more than |RTCP_MAX_REPORT_BLOCKS| per
  // compound rtcp packet when single rtcp module is used for multiple media
  // streams.
  result = receive_statistics_->RtcpReportBlocks(RTCP_MAX_REPORT_BLOCKS);

  if (!result.empty() && ((feedback_state.last_rr_ntp_secs != 0) ||
                          (feedback_state.last_rr_ntp_frac != 0))) {
    // Get our NTP as late as possible to avoid a race.
    uint32_t now = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));

    uint32_t receive_time = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
    receive_time <<= 16;
    receive_time += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;

    uint32_t delay_since_last_sr = now - receive_time;
    // TODO(danilchap): Instead of setting same value on all report blocks,
    // set only when media_ssrc match sender ssrc of the sender report
    // remote times were taken from.
    for (auto& report_block : result) {
      report_block.SetLastSr(feedback_state.remote_sr);
      report_block.SetDelayLastSr(delay_since_last_sr);
    }
  }
  return result;
}
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