| /* |
| * Copyright 2017 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/rtp_transport.h" |
| |
| #include <errno.h> |
| |
| #include <cstdint> |
| #include <utility> |
| |
| #include "api/array_view.h" |
| #include "api/units/timestamp.h" |
| #include "media/base/rtp_utils.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace webrtc { |
| |
| void RtpTransport::SetRtcpMuxEnabled(bool enable) { |
| rtcp_mux_enabled_ = enable; |
| MaybeSignalReadyToSend(); |
| } |
| |
| const std::string& RtpTransport::transport_name() const { |
| return rtp_packet_transport_->transport_name(); |
| } |
| |
| int RtpTransport::SetRtpOption(rtc::Socket::Option opt, int value) { |
| return rtp_packet_transport_->SetOption(opt, value); |
| } |
| |
| int RtpTransport::SetRtcpOption(rtc::Socket::Option opt, int value) { |
| if (rtcp_packet_transport_) { |
| return rtcp_packet_transport_->SetOption(opt, value); |
| } |
| return -1; |
| } |
| |
| void RtpTransport::SetRtpPacketTransport( |
| rtc::PacketTransportInternal* new_packet_transport) { |
| if (new_packet_transport == rtp_packet_transport_) { |
| return; |
| } |
| if (rtp_packet_transport_) { |
| rtp_packet_transport_->SignalReadyToSend.disconnect(this); |
| rtp_packet_transport_->DeregisterReceivedPacketCallback(this); |
| rtp_packet_transport_->SignalNetworkRouteChanged.disconnect(this); |
| rtp_packet_transport_->SignalWritableState.disconnect(this); |
| rtp_packet_transport_->SignalSentPacket.disconnect(this); |
| // Reset the network route of the old transport. |
| SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>()); |
| } |
| if (new_packet_transport) { |
| new_packet_transport->SignalReadyToSend.connect( |
| this, &RtpTransport::OnReadyToSend); |
| new_packet_transport->RegisterReceivedPacketCallback( |
| this, [&](rtc::PacketTransportInternal* transport, |
| const rtc::ReceivedPacket& packet) { |
| OnReadPacket(transport, packet); |
| }); |
| new_packet_transport->SignalNetworkRouteChanged.connect( |
| this, &RtpTransport::OnNetworkRouteChanged); |
| new_packet_transport->SignalWritableState.connect( |
| this, &RtpTransport::OnWritableState); |
| new_packet_transport->SignalSentPacket.connect(this, |
| &RtpTransport::OnSentPacket); |
| // Set the network route for the new transport. |
| SendNetworkRouteChanged(new_packet_transport->network_route()); |
| } |
| |
| rtp_packet_transport_ = new_packet_transport; |
| SetReadyToSend(false, |
| rtp_packet_transport_ && rtp_packet_transport_->writable()); |
| } |
| |
| void RtpTransport::SetRtcpPacketTransport( |
| rtc::PacketTransportInternal* new_packet_transport) { |
| if (new_packet_transport == rtcp_packet_transport_) { |
| return; |
| } |
| if (rtcp_packet_transport_) { |
| rtcp_packet_transport_->SignalReadyToSend.disconnect(this); |
| rtcp_packet_transport_->DeregisterReceivedPacketCallback(this); |
| rtcp_packet_transport_->SignalNetworkRouteChanged.disconnect(this); |
| rtcp_packet_transport_->SignalWritableState.disconnect(this); |
| rtcp_packet_transport_->SignalSentPacket.disconnect(this); |
| // Reset the network route of the old transport. |
| SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>()); |
| } |
| if (new_packet_transport) { |
| new_packet_transport->SignalReadyToSend.connect( |
| this, &RtpTransport::OnReadyToSend); |
| new_packet_transport->RegisterReceivedPacketCallback( |
| this, [&](rtc::PacketTransportInternal* transport, |
| const rtc::ReceivedPacket& packet) { |
| OnReadPacket(transport, packet); |
| }); |
| new_packet_transport->SignalNetworkRouteChanged.connect( |
| this, &RtpTransport::OnNetworkRouteChanged); |
| new_packet_transport->SignalWritableState.connect( |
| this, &RtpTransport::OnWritableState); |
| new_packet_transport->SignalSentPacket.connect(this, |
| &RtpTransport::OnSentPacket); |
| // Set the network route for the new transport. |
| SendNetworkRouteChanged(new_packet_transport->network_route()); |
| } |
| rtcp_packet_transport_ = new_packet_transport; |
| |
| // Assumes the transport is ready to send if it is writable. |
| SetReadyToSend(true, |
| rtcp_packet_transport_ && rtcp_packet_transport_->writable()); |
| } |
| |
| bool RtpTransport::IsWritable(bool rtcp) const { |
| rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_ |
| ? rtcp_packet_transport_ |
| : rtp_packet_transport_; |
| return transport && transport->writable(); |
| } |
| |
| bool RtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| return SendPacket(false, packet, options, flags); |
| } |
| |
| bool RtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| return SendPacket(true, packet, options, flags); |
| } |
| |
| bool RtpTransport::SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_ |
| ? rtcp_packet_transport_ |
| : rtp_packet_transport_; |
| int ret = transport->SendPacket(packet->cdata<char>(), packet->size(), |
| options, flags); |
| if (ret != static_cast<int>(packet->size())) { |
| if (set_ready_to_send_false_if_send_fail_) { |
| // TODO: webrtc:361124449 - Remove SetReadyToSend if field trial |
| // WebRTC-SetReadyToSendFalseIfSendFail succeed 2024-12-01. |
| if (transport->GetError() == ENOTCONN) { |
| RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport."; |
| SetReadyToSend(rtcp, false); |
| } |
| } |
| return false; |
| } |
| return true; |
| } |
| |
| void RtpTransport::UpdateRtpHeaderExtensionMap( |
| const cricket::RtpHeaderExtensions& header_extensions) { |
| header_extension_map_ = RtpHeaderExtensionMap(header_extensions); |
| } |
| |
| 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; |
| } |
| |
| bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) { |
| if (!rtp_demuxer_.RemoveSink(sink)) { |
| RTC_LOG(LS_ERROR) << "Failed to unregister the sink for RTP demuxer."; |
| return false; |
| } |
| return true; |
| } |
| |
| flat_set<uint32_t> RtpTransport::GetSsrcsForSink(RtpPacketSinkInterface* sink) { |
| return rtp_demuxer_.GetSsrcsForSink(sink); |
| } |
| |
| void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet, |
| webrtc::Timestamp arrival_time, |
| rtc::EcnMarking ecn) { |
| RtpPacketReceived parsed_packet(&header_extension_map_); |
| parsed_packet.set_arrival_time(arrival_time); |
| parsed_packet.set_ecn(ecn); |
| |
| if (!parsed_packet.Parse(std::move(packet))) { |
| RTC_LOG(LS_ERROR) |
| << "Failed to parse the incoming RTP packet before demuxing. Drop it."; |
| return; |
| } |
| |
| if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) { |
| RTC_LOG(LS_VERBOSE) << "Failed to demux RTP packet: " |
| << RtpDemuxer::DescribePacket(parsed_packet); |
| NotifyUnDemuxableRtpPacketReceived(parsed_packet); |
| } |
| } |
| |
| bool RtpTransport::IsTransportWritable() { |
| auto rtcp_packet_transport = |
| rtcp_mux_enabled_ ? nullptr : rtcp_packet_transport_; |
| return rtp_packet_transport_ && rtp_packet_transport_->writable() && |
| (!rtcp_packet_transport || rtcp_packet_transport->writable()); |
| } |
| |
| void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) { |
| SetReadyToSend(transport == rtcp_packet_transport_, true); |
| } |
| |
| void RtpTransport::OnNetworkRouteChanged( |
| absl::optional<rtc::NetworkRoute> network_route) { |
| SendNetworkRouteChanged(network_route); |
| } |
| |
| void RtpTransport::OnWritableState( |
| rtc::PacketTransportInternal* packet_transport) { |
| RTC_DCHECK(packet_transport == rtp_packet_transport_ || |
| packet_transport == rtcp_packet_transport_); |
| SendWritableState(IsTransportWritable()); |
| } |
| |
| void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport, |
| const rtc::SentPacket& sent_packet) { |
| RTC_DCHECK(packet_transport == rtp_packet_transport_ || |
| packet_transport == rtcp_packet_transport_); |
| if (processing_sent_packet_) { |
| TaskQueueBase::Current()->PostTask(SafeTask( |
| safety_.flag(), [this, sent_packet] { SendSentPacket(sent_packet); })); |
| return; |
| } |
| processing_sent_packet_ = true; |
| SendSentPacket(sent_packet); |
| processing_sent_packet_ = false; |
| } |
| |
| void RtpTransport::OnRtpPacketReceived( |
| const rtc::ReceivedPacket& received_packet) { |
| rtc::CopyOnWriteBuffer payload(received_packet.payload()); |
| DemuxPacket( |
| payload, |
| received_packet.arrival_time().value_or(Timestamp::MinusInfinity()), |
| received_packet.ecn()); |
| } |
| |
| void RtpTransport::OnRtcpPacketReceived( |
| const rtc::ReceivedPacket& received_packet) { |
| rtc::CopyOnWriteBuffer payload(received_packet.payload()); |
| // TODO(bugs.webrtc.org/15368): Propagate timestamp and maybe received packet |
| // further. |
| SendRtcpPacketReceived(&payload, received_packet.arrival_time() |
| ? received_packet.arrival_time()->us() |
| : -1); |
| } |
| |
| void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport, |
| const rtc::ReceivedPacket& received_packet) { |
| TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket"); |
| |
| // When using RTCP multiplexing we might get RTCP packets on the RTP |
| // transport. We check the RTP payload type to determine if it is RTCP. |
| cricket::RtpPacketType packet_type = |
| cricket::InferRtpPacketType(received_packet.payload()); |
| // Filter out the packet that is neither RTP nor RTCP. |
| if (packet_type == cricket::RtpPacketType::kUnknown) { |
| return; |
| } |
| |
| // Protect ourselves against crazy data. |
| if (!cricket::IsValidRtpPacketSize(packet_type, |
| received_packet.payload().size())) { |
| RTC_LOG(LS_ERROR) << "Dropping incoming " |
| << cricket::RtpPacketTypeToString(packet_type) |
| << " packet: wrong size=" |
| << received_packet.payload().size(); |
| return; |
| } |
| |
| if (packet_type == cricket::RtpPacketType::kRtcp) { |
| OnRtcpPacketReceived(received_packet); |
| } else { |
| OnRtpPacketReceived(received_packet); |
| } |
| } |
| |
| void RtpTransport::SetReadyToSend(bool rtcp, bool ready) { |
| if (rtcp) { |
| rtcp_ready_to_send_ = ready; |
| } else { |
| rtp_ready_to_send_ = ready; |
| } |
| |
| MaybeSignalReadyToSend(); |
| } |
| |
| void RtpTransport::MaybeSignalReadyToSend() { |
| bool ready_to_send = |
| rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_); |
| if (ready_to_send != ready_to_send_) { |
| if (processing_ready_to_send_) { |
| // Delay ReadyToSend processing until current operation is finished. |
| // Note that this may not cause a signal, since ready_to_send may |
| // have a new value by the time this executes. |
| TaskQueueBase::Current()->PostTask( |
| SafeTask(safety_.flag(), [this] { MaybeSignalReadyToSend(); })); |
| return; |
| } |
| ready_to_send_ = ready_to_send; |
| processing_ready_to_send_ = true; |
| SendReadyToSend(ready_to_send); |
| processing_ready_to_send_ = false; |
| } |
| } |
| |
| } // namespace webrtc |
