| /* |
| * Copyright 2018 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/datagram_rtp_transport.h" |
| |
| #include <algorithm> |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/strings/string_view.h" |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/rtc_error.h" |
| #include "media/base/rtp_utils.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "p2p/base/dtls_transport_internal.h" |
| #include "p2p/base/packet_transport_internal.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/message_queue.h" |
| #include "rtc_base/rtc_certificate.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/stream.h" |
| #include "rtc_base/thread.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| // Field trials. |
| // Disable datagram to RTCP feedback translation and enable RTCP feedback loop |
| // on top of datagram feedback loop. Note that two |
| // feedback loops add unneccesary overhead, so it's preferable to use feedback |
| // loop provided by datagram transport and convert datagram ACKs to RTCP ACKs, |
| // but enabling RTCP feedback loop may be useful in tests and experiments. |
| const char kDisableDatagramToRtcpFeebackTranslationFieldTrial[] = |
| "WebRTC-kDisableDatagramToRtcpFeebackTranslation"; |
| |
| } // namespace |
| |
| // Maximum packet size of RTCP feedback packet for allocation. We re-create RTCP |
| // feedback packets when we get ACK notifications from datagram transport. Our |
| // rtcp feedback packets contain only 1 ACK, so they are much smaller than 1250. |
| constexpr size_t kMaxRtcpFeedbackPacketSize = 1250; |
| |
| DatagramRtpTransport::DatagramRtpTransport( |
| const std::vector<RtpExtension>& rtp_header_extensions, |
| cricket::IceTransportInternal* ice_transport, |
| DatagramTransportInterface* datagram_transport) |
| : ice_transport_(ice_transport), |
| datagram_transport_(datagram_transport), |
| disable_datagram_to_rtcp_feeback_translation_(field_trial::IsEnabled( |
| kDisableDatagramToRtcpFeebackTranslationFieldTrial)) { |
| // Save extension map for parsing RTP packets (we only need transport |
| // sequence numbers). |
| const RtpExtension* transport_sequence_number_extension = |
| RtpExtension::FindHeaderExtensionByUri(rtp_header_extensions, |
| TransportSequenceNumber::kUri); |
| |
| if (transport_sequence_number_extension != nullptr) { |
| rtp_header_extension_map_.Register<TransportSequenceNumber>( |
| transport_sequence_number_extension->id); |
| } else { |
| RTC_LOG(LS_ERROR) << "Transport sequence numbers are not supported in " |
| "datagram transport connection"; |
| } |
| |
| // TODO(sukhanov): Add CHECK to make sure that field trial |
| // WebRTC-ExcludeTransportSequenceNumberFromFecFieldTrial is enabled. |
| // If feedback loop is translation is enabled, FEC packets must exclude |
| // transport sequence numbers, otherwise recovered packets will be corrupt. |
| |
| RTC_DCHECK(ice_transport_); |
| RTC_DCHECK(datagram_transport_); |
| |
| ice_transport_->SignalNetworkRouteChanged.connect( |
| this, &DatagramRtpTransport::OnNetworkRouteChanged); |
| // Subscribe to DatagramTransport to read incoming packets. |
| datagram_transport_->SetDatagramSink(this); |
| datagram_transport_->SetTransportStateCallback(this); |
| } |
| |
| DatagramRtpTransport::~DatagramRtpTransport() { |
| // Unsubscribe from DatagramTransport sinks. |
| datagram_transport_->SetDatagramSink(nullptr); |
| datagram_transport_->SetTransportStateCallback(nullptr); |
| } |
| |
| bool DatagramRtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Assign and increment datagram_id. |
| const DatagramId datagram_id = current_datagram_id_++; |
| |
| // Send as is (without extracting transport sequence number) for |
| // RTP packets if we are not doing datagram => RTCP feedback translation. |
| if (disable_datagram_to_rtcp_feeback_translation_) { |
| // Even if we are not extracting transport sequence number we need to |
| // propagate "Sent" notification for both RTP and RTCP packets. For this |
| // reason we need save options.packet_id in packet map. |
| sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| |
| return SendDatagram(*packet, datagram_id); |
| } |
| |
| // Parse RTP packet. |
| RtpPacket rtp_packet(&rtp_header_extension_map_); |
| // TODO(mellem): Verify that this doesn't mangle something (it shouldn't). |
| if (!rtp_packet.Parse(*packet)) { |
| RTC_NOTREACHED() << "Failed to parse outgoing RtpPacket, len=" |
| << packet->size() |
| << ", options.packet_id=" << options.packet_id; |
| return -1; |
| } |
| |
| // Try to get transport sequence number. |
| uint16_t transport_senquence_number; |
| if (!rtp_packet.GetExtension<TransportSequenceNumber>( |
| &transport_senquence_number)) { |
| // Save packet info without transport sequence number. |
| sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| |
| RTC_LOG(LS_VERBOSE) |
| << "Sending rtp packet without transport sequence number, packet=" |
| << rtp_packet.ToString(); |
| |
| return SendDatagram(*packet, datagram_id); |
| } |
| |
| // Save packet info with sequence number and ssrc so we could reconstruct |
| // RTCP feedback packet when we receive datagram ACK. |
| sent_rtp_packet_map_[datagram_id] = SentPacketInfo( |
| options.packet_id, rtp_packet.Ssrc(), transport_senquence_number); |
| |
| // Since datagram transport provides feedback and timestamps, we do not need |
| // to send transport sequence number, so we remove it from RTP packet. Later |
| // when we get Ack for sent datagram, we will re-create RTCP feedback packet. |
| if (!rtp_packet.RemoveExtension(TransportSequenceNumber::kId)) { |
| RTC_NOTREACHED() << "Failed to remove transport sequence number, packet=" |
| << rtp_packet.ToString(); |
| return -1; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "Removed transport_senquence_number=" |
| << transport_senquence_number |
| << " from packet=" << rtp_packet.ToString() |
| << ", saved bytes=" << packet->size() - rtp_packet.size(); |
| |
| return SendDatagram( |
| rtc::ArrayView<const uint8_t>(rtp_packet.data(), rtp_packet.size()), |
| datagram_id); |
| } |
| |
| bool DatagramRtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Assign and increment datagram_id. |
| const DatagramId datagram_id = current_datagram_id_++; |
| |
| // Even if we are not extracting transport sequence number we need to |
| // propagate "Sent" notification for both RTP and RTCP packets. For this |
| // reason we need save options.packet_id in packet map. |
| sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| return SendDatagram(*packet, datagram_id); |
| } |
| |
| bool DatagramRtpTransport::SendDatagram(rtc::ArrayView<const uint8_t> data, |
| DatagramId datagram_id) { |
| return datagram_transport_->SendDatagram(data, datagram_id).ok(); |
| } |
| |
| void DatagramRtpTransport::OnDatagramReceived( |
| rtc::ArrayView<const uint8_t> data) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| rtc::ArrayView<const char> cdata(reinterpret_cast<const char*>(data.data()), |
| data.size()); |
| if (cricket::InferRtpPacketType(cdata) == cricket::RtpPacketType::kRtcp) { |
| rtc::CopyOnWriteBuffer buffer(data.data(), data.size()); |
| SignalRtcpPacketReceived(&buffer, /*packet_time_us=*/-1); |
| return; |
| } |
| |
| // TODO(sukhanov): I am not filling out time, but on my video quality |
| // test in WebRTC the time was not set either and higher layers of the stack |
| // overwrite -1 with current current rtc time. Leaveing comment for now to |
| // make sure it works as expected. |
| RtpPacketReceived parsed_packet(&rtp_header_extension_map_); |
| if (!parsed_packet.Parse(data)) { |
| RTC_LOG(LS_ERROR) << "Failed to parse incoming RTP packet"; |
| return; |
| } |
| if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) { |
| RTC_LOG(LS_WARNING) << "Failed to demux RTP packet: " |
| << RtpDemuxer::DescribePacket(parsed_packet); |
| } |
| } |
| |
| void DatagramRtpTransport::OnDatagramSent(DatagramId datagram_id) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Find packet_id and propagate OnPacketSent notification. |
| const auto& it = sent_rtp_packet_map_.find(datagram_id); |
| if (it == sent_rtp_packet_map_.end()) { |
| RTC_NOTREACHED() << "Did not find sent packet info for sent datagram_id=" |
| << datagram_id; |
| return; |
| } |
| |
| // Also see how DatagramRtpTransport::OnSentPacket handles OnSentPacket |
| // notification from ICE in bypass mode. |
| rtc::SentPacket sent_packet(/*packet_id=*/it->second.packet_id, |
| rtc::TimeMillis()); |
| |
| SignalSentPacket(sent_packet); |
| } |
| |
| bool DatagramRtpTransport::GetAndRemoveSentPacketInfo( |
| DatagramId datagram_id, |
| SentPacketInfo* sent_packet_info) { |
| RTC_CHECK(sent_packet_info != nullptr); |
| |
| const auto& it = sent_rtp_packet_map_.find(datagram_id); |
| if (it == sent_rtp_packet_map_.end()) { |
| return false; |
| } |
| |
| *sent_packet_info = it->second; |
| sent_rtp_packet_map_.erase(it); |
| return true; |
| } |
| |
| void DatagramRtpTransport::OnDatagramAcked(const DatagramAck& ack) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| SentPacketInfo sent_packet_info; |
| if (!GetAndRemoveSentPacketInfo(ack.datagram_id, &sent_packet_info)) { |
| // TODO(sukhanov): If OnDatagramAck() can come after OnDatagramLost(), |
| // datagram_id is already deleted and we may need to relax the CHECK below. |
| // It's probably OK to ignore such datagrams, because it's been a few RTTs |
| // anyway since they were sent. |
| RTC_NOTREACHED() << "Did not find sent packet info for datagram_id=" |
| << ack.datagram_id; |
| return; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "Datagram acked, ack.datagram_id=" << ack.datagram_id |
| << ", sent_packet_info.packet_id=" |
| << sent_packet_info.packet_id |
| << ", sent_packet_info.transport_sequence_number=" |
| << sent_packet_info.transport_sequence_number.value_or(-1) |
| << ", sent_packet_info.ssrc=" |
| << sent_packet_info.ssrc.value_or(-1) |
| << ", receive_timestamp_ms=" |
| << ack.receive_timestamp.ms(); |
| |
| // If transport sequence number was not present in RTP packet, we do not need |
| // to propagate RTCP feedback. |
| if (!sent_packet_info.transport_sequence_number) { |
| return; |
| } |
| |
| // TODO(sukhanov): We noticed that datagram transport implementations can |
| // return zero timestamps in the middle of the call. This is workaround to |
| // avoid propagating zero timestamps, but we need to understand why we have |
| // them in the first place. |
| int64_t receive_timestamp_us = ack.receive_timestamp.us(); |
| |
| if (receive_timestamp_us == 0) { |
| receive_timestamp_us = previous_nonzero_timestamp_us_; |
| } else { |
| previous_nonzero_timestamp_us_ = receive_timestamp_us; |
| } |
| |
| // Ssrc must be provided in packet info if transport sequence number is set, |
| // which is guaranteed by SentPacketInfo constructor. |
| RTC_CHECK(sent_packet_info.ssrc); |
| |
| // Recreate RTCP feedback packet. |
| rtcp::TransportFeedback feedback_packet; |
| feedback_packet.SetMediaSsrc(*sent_packet_info.ssrc); |
| |
| const uint16_t transport_sequence_number = |
| sent_packet_info.transport_sequence_number.value(); |
| |
| feedback_packet.SetBase(transport_sequence_number, receive_timestamp_us); |
| feedback_packet.AddReceivedPacket(transport_sequence_number, |
| receive_timestamp_us); |
| |
| rtc::CopyOnWriteBuffer buffer(kMaxRtcpFeedbackPacketSize); |
| size_t index = 0; |
| if (!feedback_packet.Create(buffer.data(), &index, buffer.capacity(), |
| nullptr)) { |
| RTC_NOTREACHED() << "Failed to create RTCP feedback packet"; |
| return; |
| } |
| |
| RTC_CHECK_GT(index, 0); |
| RTC_CHECK_LE(index, kMaxRtcpFeedbackPacketSize); |
| |
| // Propagage created RTCP packet as normal incoming packet. |
| buffer.SetSize(index); |
| SignalRtcpPacketReceived(&buffer, /*packet_time_us=*/-1); |
| } |
| |
| void DatagramRtpTransport::OnDatagramLost(DatagramId datagram_id) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| RTC_LOG(LS_INFO) << "Datagram lost, datagram_id=" << datagram_id; |
| |
| SentPacketInfo sent_packet_info; |
| if (!GetAndRemoveSentPacketInfo(datagram_id, &sent_packet_info)) { |
| RTC_NOTREACHED() << "Did not find sent packet info for lost datagram_id=" |
| << datagram_id; |
| } |
| } |
| |
| void DatagramRtpTransport::OnStateChanged(MediaTransportState state) { |
| state_ = state; |
| SignalWritableState(state_ == MediaTransportState::kWritable); |
| if (state_ == MediaTransportState::kWritable) { |
| SignalReadyToSend(true); |
| } |
| } |
| |
| const std::string& DatagramRtpTransport::transport_name() const { |
| return ice_transport_->transport_name(); |
| } |
| |
| int DatagramRtpTransport::SetRtpOption(rtc::Socket::Option opt, int value) { |
| return ice_transport_->SetOption(opt, value); |
| } |
| |
| int DatagramRtpTransport::SetRtcpOption(rtc::Socket::Option opt, int value) { |
| return -1; |
| } |
| |
| bool DatagramRtpTransport::IsReadyToSend() const { |
| return state_ == MediaTransportState::kWritable; |
| } |
| |
| bool DatagramRtpTransport::IsWritable(bool /*rtcp*/) const { |
| return state_ == MediaTransportState::kWritable; |
| } |
| |
| void DatagramRtpTransport::UpdateRtpHeaderExtensionMap( |
| const cricket::RtpHeaderExtensions& header_extensions) { |
| rtp_header_extension_map_ = RtpHeaderExtensionMap(header_extensions); |
| } |
| |
| bool DatagramRtpTransport::RegisterRtpDemuxerSink( |
| const RtpDemuxerCriteria& criteria, |
| RtpPacketSinkInterface* sink) { |
| rtp_demuxer_.RemoveSink(sink); |
| return rtp_demuxer_.AddSink(criteria, sink); |
| } |
| |
| bool DatagramRtpTransport::UnregisterRtpDemuxerSink( |
| RtpPacketSinkInterface* sink) { |
| return rtp_demuxer_.RemoveSink(sink); |
| } |
| |
| void DatagramRtpTransport::OnNetworkRouteChanged( |
| absl::optional<rtc::NetworkRoute> network_route) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| SignalNetworkRouteChanged(network_route); |
| } |
| |
| } // namespace webrtc |