Bjorn A Mellem | 364b267 | 2019-08-20 16:58:03 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2018 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include "pc/datagram_rtp_transport.h" |
| 12 | |
| 13 | #include <algorithm> |
| 14 | #include <memory> |
| 15 | #include <utility> |
| 16 | |
| 17 | #include "absl/memory/memory.h" |
| 18 | #include "absl/strings/string_view.h" |
| 19 | #include "absl/types/optional.h" |
| 20 | #include "api/array_view.h" |
| 21 | #include "api/rtc_error.h" |
| 22 | #include "media/base/rtp_utils.h" |
Bjorn A Mellem | 364b267 | 2019-08-20 16:58:03 -0700 | [diff] [blame] | 23 | #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| 24 | #include "modules/rtp_rtcp/source/rtp_packet.h" |
| 25 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| 26 | #include "p2p/base/dtls_transport_internal.h" |
| 27 | #include "p2p/base/packet_transport_internal.h" |
| 28 | #include "rtc_base/buffer.h" |
| 29 | #include "rtc_base/checks.h" |
| 30 | #include "rtc_base/dscp.h" |
| 31 | #include "rtc_base/logging.h" |
Bjorn A Mellem | 364b267 | 2019-08-20 16:58:03 -0700 | [diff] [blame] | 32 | #include "rtc_base/rtc_certificate.h" |
| 33 | #include "rtc_base/ssl_stream_adapter.h" |
| 34 | #include "rtc_base/stream.h" |
| 35 | #include "rtc_base/thread.h" |
| 36 | #include "system_wrappers/include/field_trial.h" |
| 37 | |
| 38 | namespace webrtc { |
| 39 | |
| 40 | namespace { |
| 41 | |
| 42 | // Field trials. |
| 43 | // Disable datagram to RTCP feedback translation and enable RTCP feedback loop |
| 44 | // on top of datagram feedback loop. Note that two |
| 45 | // feedback loops add unneccesary overhead, so it's preferable to use feedback |
| 46 | // loop provided by datagram transport and convert datagram ACKs to RTCP ACKs, |
| 47 | // but enabling RTCP feedback loop may be useful in tests and experiments. |
| 48 | const char kDisableDatagramToRtcpFeebackTranslationFieldTrial[] = |
| 49 | "WebRTC-kDisableDatagramToRtcpFeebackTranslation"; |
| 50 | |
| 51 | } // namespace |
| 52 | |
| 53 | // Maximum packet size of RTCP feedback packet for allocation. We re-create RTCP |
| 54 | // feedback packets when we get ACK notifications from datagram transport. Our |
| 55 | // rtcp feedback packets contain only 1 ACK, so they are much smaller than 1250. |
| 56 | constexpr size_t kMaxRtcpFeedbackPacketSize = 1250; |
| 57 | |
| 58 | DatagramRtpTransport::DatagramRtpTransport( |
| 59 | const std::vector<RtpExtension>& rtp_header_extensions, |
| 60 | cricket::IceTransportInternal* ice_transport, |
| 61 | DatagramTransportInterface* datagram_transport) |
| 62 | : ice_transport_(ice_transport), |
| 63 | datagram_transport_(datagram_transport), |
| 64 | disable_datagram_to_rtcp_feeback_translation_(field_trial::IsEnabled( |
| 65 | kDisableDatagramToRtcpFeebackTranslationFieldTrial)) { |
| 66 | // Save extension map for parsing RTP packets (we only need transport |
| 67 | // sequence numbers). |
| 68 | const RtpExtension* transport_sequence_number_extension = |
| 69 | RtpExtension::FindHeaderExtensionByUri(rtp_header_extensions, |
| 70 | TransportSequenceNumber::kUri); |
| 71 | |
| 72 | if (transport_sequence_number_extension != nullptr) { |
| 73 | rtp_header_extension_map_.Register<TransportSequenceNumber>( |
| 74 | transport_sequence_number_extension->id); |
| 75 | } else { |
| 76 | RTC_LOG(LS_ERROR) << "Transport sequence numbers are not supported in " |
| 77 | "datagram transport connection"; |
| 78 | } |
| 79 | |
| 80 | // TODO(sukhanov): Add CHECK to make sure that field trial |
| 81 | // WebRTC-ExcludeTransportSequenceNumberFromFecFieldTrial is enabled. |
| 82 | // If feedback loop is translation is enabled, FEC packets must exclude |
| 83 | // transport sequence numbers, otherwise recovered packets will be corrupt. |
| 84 | |
| 85 | RTC_DCHECK(ice_transport_); |
| 86 | RTC_DCHECK(datagram_transport_); |
| 87 | |
| 88 | ice_transport_->SignalNetworkRouteChanged.connect( |
| 89 | this, &DatagramRtpTransport::OnNetworkRouteChanged); |
| 90 | // Subscribe to DatagramTransport to read incoming packets. |
| 91 | datagram_transport_->SetDatagramSink(this); |
| 92 | datagram_transport_->SetTransportStateCallback(this); |
| 93 | } |
| 94 | |
| 95 | DatagramRtpTransport::~DatagramRtpTransport() { |
| 96 | // Unsubscribe from DatagramTransport sinks. |
| 97 | datagram_transport_->SetDatagramSink(nullptr); |
| 98 | datagram_transport_->SetTransportStateCallback(nullptr); |
| 99 | } |
| 100 | |
| 101 | bool DatagramRtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
| 102 | const rtc::PacketOptions& options, |
| 103 | int flags) { |
| 104 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 105 | |
| 106 | // Assign and increment datagram_id. |
| 107 | const DatagramId datagram_id = current_datagram_id_++; |
| 108 | |
| 109 | // Send as is (without extracting transport sequence number) for |
| 110 | // RTP packets if we are not doing datagram => RTCP feedback translation. |
| 111 | if (disable_datagram_to_rtcp_feeback_translation_) { |
| 112 | // Even if we are not extracting transport sequence number we need to |
| 113 | // propagate "Sent" notification for both RTP and RTCP packets. For this |
| 114 | // reason we need save options.packet_id in packet map. |
| 115 | sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| 116 | |
| 117 | return SendDatagram(*packet, datagram_id); |
| 118 | } |
| 119 | |
| 120 | // Parse RTP packet. |
| 121 | RtpPacket rtp_packet(&rtp_header_extension_map_); |
| 122 | // TODO(mellem): Verify that this doesn't mangle something (it shouldn't). |
| 123 | if (!rtp_packet.Parse(*packet)) { |
| 124 | RTC_NOTREACHED() << "Failed to parse outgoing RtpPacket, len=" |
| 125 | << packet->size() |
| 126 | << ", options.packet_id=" << options.packet_id; |
| 127 | return -1; |
| 128 | } |
| 129 | |
| 130 | // Try to get transport sequence number. |
| 131 | uint16_t transport_senquence_number; |
| 132 | if (!rtp_packet.GetExtension<TransportSequenceNumber>( |
| 133 | &transport_senquence_number)) { |
| 134 | // Save packet info without transport sequence number. |
| 135 | sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| 136 | |
| 137 | RTC_LOG(LS_VERBOSE) |
| 138 | << "Sending rtp packet without transport sequence number, packet=" |
| 139 | << rtp_packet.ToString(); |
| 140 | |
| 141 | return SendDatagram(*packet, datagram_id); |
| 142 | } |
| 143 | |
| 144 | // Save packet info with sequence number and ssrc so we could reconstruct |
| 145 | // RTCP feedback packet when we receive datagram ACK. |
| 146 | sent_rtp_packet_map_[datagram_id] = SentPacketInfo( |
| 147 | options.packet_id, rtp_packet.Ssrc(), transport_senquence_number); |
| 148 | |
| 149 | // Since datagram transport provides feedback and timestamps, we do not need |
| 150 | // to send transport sequence number, so we remove it from RTP packet. Later |
| 151 | // when we get Ack for sent datagram, we will re-create RTCP feedback packet. |
| 152 | if (!rtp_packet.RemoveExtension(TransportSequenceNumber::kId)) { |
| 153 | RTC_NOTREACHED() << "Failed to remove transport sequence number, packet=" |
| 154 | << rtp_packet.ToString(); |
| 155 | return -1; |
| 156 | } |
| 157 | |
| 158 | RTC_LOG(LS_VERBOSE) << "Removed transport_senquence_number=" |
| 159 | << transport_senquence_number |
| 160 | << " from packet=" << rtp_packet.ToString() |
| 161 | << ", saved bytes=" << packet->size() - rtp_packet.size(); |
| 162 | |
| 163 | return SendDatagram( |
| 164 | rtc::ArrayView<const uint8_t>(rtp_packet.data(), rtp_packet.size()), |
| 165 | datagram_id); |
| 166 | } |
| 167 | |
| 168 | bool DatagramRtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
| 169 | const rtc::PacketOptions& options, |
| 170 | int flags) { |
| 171 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 172 | |
| 173 | // Assign and increment datagram_id. |
| 174 | const DatagramId datagram_id = current_datagram_id_++; |
| 175 | |
| 176 | // Even if we are not extracting transport sequence number we need to |
| 177 | // propagate "Sent" notification for both RTP and RTCP packets. For this |
| 178 | // reason we need save options.packet_id in packet map. |
| 179 | sent_rtp_packet_map_[datagram_id] = SentPacketInfo(options.packet_id); |
| 180 | return SendDatagram(*packet, datagram_id); |
| 181 | } |
| 182 | |
| 183 | bool DatagramRtpTransport::SendDatagram(rtc::ArrayView<const uint8_t> data, |
| 184 | DatagramId datagram_id) { |
| 185 | return datagram_transport_->SendDatagram(data, datagram_id).ok(); |
| 186 | } |
| 187 | |
| 188 | void DatagramRtpTransport::OnDatagramReceived( |
| 189 | rtc::ArrayView<const uint8_t> data) { |
| 190 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 191 | |
| 192 | rtc::ArrayView<const char> cdata(reinterpret_cast<const char*>(data.data()), |
| 193 | data.size()); |
| 194 | if (cricket::InferRtpPacketType(cdata) == cricket::RtpPacketType::kRtcp) { |
| 195 | rtc::CopyOnWriteBuffer buffer(data.data(), data.size()); |
| 196 | SignalRtcpPacketReceived(&buffer, /*packet_time_us=*/-1); |
| 197 | return; |
| 198 | } |
| 199 | |
| 200 | // TODO(sukhanov): I am not filling out time, but on my video quality |
| 201 | // test in WebRTC the time was not set either and higher layers of the stack |
| 202 | // overwrite -1 with current current rtc time. Leaveing comment for now to |
| 203 | // make sure it works as expected. |
| 204 | RtpPacketReceived parsed_packet(&rtp_header_extension_map_); |
| 205 | if (!parsed_packet.Parse(data)) { |
| 206 | RTC_LOG(LS_ERROR) << "Failed to parse incoming RTP packet"; |
| 207 | return; |
| 208 | } |
| 209 | if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) { |
| 210 | RTC_LOG(LS_WARNING) << "Failed to demux RTP packet: " |
| 211 | << RtpDemuxer::DescribePacket(parsed_packet); |
| 212 | } |
| 213 | } |
| 214 | |
| 215 | void DatagramRtpTransport::OnDatagramSent(DatagramId datagram_id) { |
| 216 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 217 | |
| 218 | // Find packet_id and propagate OnPacketSent notification. |
| 219 | const auto& it = sent_rtp_packet_map_.find(datagram_id); |
| 220 | if (it == sent_rtp_packet_map_.end()) { |
| 221 | RTC_NOTREACHED() << "Did not find sent packet info for sent datagram_id=" |
| 222 | << datagram_id; |
| 223 | return; |
| 224 | } |
| 225 | |
| 226 | // Also see how DatagramRtpTransport::OnSentPacket handles OnSentPacket |
| 227 | // notification from ICE in bypass mode. |
| 228 | rtc::SentPacket sent_packet(/*packet_id=*/it->second.packet_id, |
| 229 | rtc::TimeMillis()); |
| 230 | |
| 231 | SignalSentPacket(sent_packet); |
| 232 | } |
| 233 | |
| 234 | bool DatagramRtpTransport::GetAndRemoveSentPacketInfo( |
| 235 | DatagramId datagram_id, |
| 236 | SentPacketInfo* sent_packet_info) { |
| 237 | RTC_CHECK(sent_packet_info != nullptr); |
| 238 | |
| 239 | const auto& it = sent_rtp_packet_map_.find(datagram_id); |
| 240 | if (it == sent_rtp_packet_map_.end()) { |
| 241 | return false; |
| 242 | } |
| 243 | |
| 244 | *sent_packet_info = it->second; |
| 245 | sent_rtp_packet_map_.erase(it); |
| 246 | return true; |
| 247 | } |
| 248 | |
| 249 | void DatagramRtpTransport::OnDatagramAcked(const DatagramAck& ack) { |
| 250 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 251 | |
| 252 | SentPacketInfo sent_packet_info; |
| 253 | if (!GetAndRemoveSentPacketInfo(ack.datagram_id, &sent_packet_info)) { |
| 254 | // TODO(sukhanov): If OnDatagramAck() can come after OnDatagramLost(), |
| 255 | // datagram_id is already deleted and we may need to relax the CHECK below. |
| 256 | // It's probably OK to ignore such datagrams, because it's been a few RTTs |
| 257 | // anyway since they were sent. |
| 258 | RTC_NOTREACHED() << "Did not find sent packet info for datagram_id=" |
| 259 | << ack.datagram_id; |
| 260 | return; |
| 261 | } |
| 262 | |
| 263 | RTC_LOG(LS_VERBOSE) << "Datagram acked, ack.datagram_id=" << ack.datagram_id |
| 264 | << ", sent_packet_info.packet_id=" |
| 265 | << sent_packet_info.packet_id |
| 266 | << ", sent_packet_info.transport_sequence_number=" |
| 267 | << sent_packet_info.transport_sequence_number.value_or(-1) |
| 268 | << ", sent_packet_info.ssrc=" |
| 269 | << sent_packet_info.ssrc.value_or(-1) |
| 270 | << ", receive_timestamp_ms=" |
| 271 | << ack.receive_timestamp.ms(); |
| 272 | |
| 273 | // If transport sequence number was not present in RTP packet, we do not need |
| 274 | // to propagate RTCP feedback. |
| 275 | if (!sent_packet_info.transport_sequence_number) { |
| 276 | return; |
| 277 | } |
| 278 | |
| 279 | // TODO(sukhanov): We noticed that datagram transport implementations can |
| 280 | // return zero timestamps in the middle of the call. This is workaround to |
| 281 | // avoid propagating zero timestamps, but we need to understand why we have |
| 282 | // them in the first place. |
| 283 | int64_t receive_timestamp_us = ack.receive_timestamp.us(); |
| 284 | |
| 285 | if (receive_timestamp_us == 0) { |
| 286 | receive_timestamp_us = previous_nonzero_timestamp_us_; |
| 287 | } else { |
| 288 | previous_nonzero_timestamp_us_ = receive_timestamp_us; |
| 289 | } |
| 290 | |
| 291 | // Ssrc must be provided in packet info if transport sequence number is set, |
| 292 | // which is guaranteed by SentPacketInfo constructor. |
| 293 | RTC_CHECK(sent_packet_info.ssrc); |
| 294 | |
| 295 | // Recreate RTCP feedback packet. |
| 296 | rtcp::TransportFeedback feedback_packet; |
| 297 | feedback_packet.SetMediaSsrc(*sent_packet_info.ssrc); |
| 298 | |
| 299 | const uint16_t transport_sequence_number = |
| 300 | sent_packet_info.transport_sequence_number.value(); |
| 301 | |
| 302 | feedback_packet.SetBase(transport_sequence_number, receive_timestamp_us); |
| 303 | feedback_packet.AddReceivedPacket(transport_sequence_number, |
| 304 | receive_timestamp_us); |
| 305 | |
| 306 | rtc::CopyOnWriteBuffer buffer(kMaxRtcpFeedbackPacketSize); |
| 307 | size_t index = 0; |
| 308 | if (!feedback_packet.Create(buffer.data(), &index, buffer.capacity(), |
| 309 | nullptr)) { |
| 310 | RTC_NOTREACHED() << "Failed to create RTCP feedback packet"; |
| 311 | return; |
| 312 | } |
| 313 | |
| 314 | RTC_CHECK_GT(index, 0); |
| 315 | RTC_CHECK_LE(index, kMaxRtcpFeedbackPacketSize); |
| 316 | |
| 317 | // Propagage created RTCP packet as normal incoming packet. |
| 318 | buffer.SetSize(index); |
| 319 | SignalRtcpPacketReceived(&buffer, /*packet_time_us=*/-1); |
| 320 | } |
| 321 | |
| 322 | void DatagramRtpTransport::OnDatagramLost(DatagramId datagram_id) { |
| 323 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 324 | |
| 325 | RTC_LOG(LS_INFO) << "Datagram lost, datagram_id=" << datagram_id; |
| 326 | |
| 327 | SentPacketInfo sent_packet_info; |
| 328 | if (!GetAndRemoveSentPacketInfo(datagram_id, &sent_packet_info)) { |
| 329 | RTC_NOTREACHED() << "Did not find sent packet info for lost datagram_id=" |
| 330 | << datagram_id; |
| 331 | } |
| 332 | } |
| 333 | |
| 334 | void DatagramRtpTransport::OnStateChanged(MediaTransportState state) { |
| 335 | state_ = state; |
| 336 | SignalWritableState(state_ == MediaTransportState::kWritable); |
| 337 | if (state_ == MediaTransportState::kWritable) { |
| 338 | SignalReadyToSend(true); |
| 339 | } |
| 340 | } |
| 341 | |
| 342 | const std::string& DatagramRtpTransport::transport_name() const { |
| 343 | return ice_transport_->transport_name(); |
| 344 | } |
| 345 | |
| 346 | int DatagramRtpTransport::SetRtpOption(rtc::Socket::Option opt, int value) { |
| 347 | return ice_transport_->SetOption(opt, value); |
| 348 | } |
| 349 | |
| 350 | int DatagramRtpTransport::SetRtcpOption(rtc::Socket::Option opt, int value) { |
| 351 | return -1; |
| 352 | } |
| 353 | |
| 354 | bool DatagramRtpTransport::IsReadyToSend() const { |
| 355 | return state_ == MediaTransportState::kWritable; |
| 356 | } |
| 357 | |
| 358 | bool DatagramRtpTransport::IsWritable(bool /*rtcp*/) const { |
| 359 | return state_ == MediaTransportState::kWritable; |
| 360 | } |
| 361 | |
| 362 | void DatagramRtpTransport::UpdateRtpHeaderExtensionMap( |
| 363 | const cricket::RtpHeaderExtensions& header_extensions) { |
| 364 | rtp_header_extension_map_ = RtpHeaderExtensionMap(header_extensions); |
| 365 | } |
| 366 | |
| 367 | bool DatagramRtpTransport::RegisterRtpDemuxerSink( |
| 368 | const RtpDemuxerCriteria& criteria, |
| 369 | RtpPacketSinkInterface* sink) { |
| 370 | rtp_demuxer_.RemoveSink(sink); |
| 371 | return rtp_demuxer_.AddSink(criteria, sink); |
| 372 | } |
| 373 | |
| 374 | bool DatagramRtpTransport::UnregisterRtpDemuxerSink( |
| 375 | RtpPacketSinkInterface* sink) { |
| 376 | return rtp_demuxer_.RemoveSink(sink); |
| 377 | } |
| 378 | |
| 379 | void DatagramRtpTransport::OnNetworkRouteChanged( |
| 380 | absl::optional<rtc::NetworkRoute> network_route) { |
| 381 | RTC_DCHECK_RUN_ON(&thread_checker_); |
| 382 | SignalNetworkRouteChanged(network_route); |
| 383 | } |
| 384 | |
| 385 | } // namespace webrtc |