| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #include <utility> | 11 #include <utility> |
| 12 | 12 |
| 13 #include "webrtc/pc/channel.h" | 13 #include "webrtc/pc/channel.h" |
| 14 | 14 |
| 15 #include "webrtc/api/call/audio_sink.h" | 15 #include "webrtc/api/call/audio_sink.h" |
| 16 #include "webrtc/base/bind.h" | 16 #include "webrtc/base/bind.h" |
| 17 #include "webrtc/base/byteorder.h" | 17 #include "webrtc/base/byteorder.h" |
| 18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
| 19 #include "webrtc/base/common.h" | 19 #include "webrtc/base/common.h" |
| 20 #include "webrtc/base/copyonwritebuffer.h" | 20 #include "webrtc/base/copyonwritebuffer.h" |
| 21 #include "webrtc/base/dscp.h" | 21 #include "webrtc/base/dscp.h" |
| 22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
| 23 #include "webrtc/base/networkroute.h" | 23 #include "webrtc/base/networkroute.h" |
| 24 #include "webrtc/base/trace_event.h" | 24 #include "webrtc/base/trace_event.h" |
| 25 #include "webrtc/media/base/mediaconstants.h" | 25 #include "webrtc/media/base/mediaconstants.h" |
| 26 #include "webrtc/media/base/rtputils.h" | 26 #include "webrtc/media/base/rtputils.h" |
| 27 #include "webrtc/p2p/base/packettransportinterface.h" | 27 #include "webrtc/p2p/base/packettransportinterface.h" |
| 28 #include "webrtc/p2p/base/transportchannel.h" | |
| 29 #include "webrtc/pc/channelmanager.h" | 28 #include "webrtc/pc/channelmanager.h" |
| 30 | 29 |
| 31 namespace cricket { | 30 namespace cricket { |
| 32 using rtc::Bind; | 31 using rtc::Bind; |
| 33 | 32 |
| 34 namespace { | 33 namespace { |
| 35 // See comment below for why we need to use a pointer to a unique_ptr. | 34 // See comment below for why we need to use a pointer to a unique_ptr. |
| 36 bool SetRawAudioSink_w(VoiceMediaChannel* channel, | 35 bool SetRawAudioSink_w(VoiceMediaChannel* channel, |
| 37 uint32_t ssrc, | 36 uint32_t ssrc, |
| 38 std::unique_ptr<webrtc::AudioSinkInterface>* sink) { | 37 std::unique_ptr<webrtc::AudioSinkInterface>* sink) { |
| (...skipping 152 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 191 delete media_channel_; | 190 delete media_channel_; |
| 192 LOG(LS_INFO) << "Destroyed channel: " << content_name_; | 191 LOG(LS_INFO) << "Destroyed channel: " << content_name_; |
| 193 } | 192 } |
| 194 | 193 |
| 195 void BaseChannel::DisconnectTransportChannels_n() { | 194 void BaseChannel::DisconnectTransportChannels_n() { |
| 196 // Send any outstanding RTCP packets. | 195 // Send any outstanding RTCP packets. |
| 197 FlushRtcpMessages_n(); | 196 FlushRtcpMessages_n(); |
| 198 | 197 |
| 199 // Stop signals from transport channels, but keep them alive because | 198 // Stop signals from transport channels, but keep them alive because |
| 200 // media_channel may use them from a different thread. | 199 // media_channel may use them from a different thread. |
| 201 if (rtp_transport_) { | 200 if (rtp_dtls_transport_) { |
| 202 DisconnectFromTransportChannel(rtp_transport_); | 201 DisconnectFromTransport(rtp_dtls_transport_); |
| 203 } | 202 } |
| 204 if (rtcp_transport_) { | 203 if (rtcp_dtls_transport_) { |
| 205 DisconnectFromTransportChannel(rtcp_transport_); | 204 DisconnectFromTransport(rtcp_dtls_transport_); |
| 206 } | 205 } |
| 207 | 206 |
| 208 // Clear pending read packets/messages. | 207 // Clear pending read packets/messages. |
| 209 network_thread_->Clear(&invoker_); | 208 network_thread_->Clear(&invoker_); |
| 210 network_thread_->Clear(this); | 209 network_thread_->Clear(this); |
| 211 } | 210 } |
| 212 | 211 |
| 213 bool BaseChannel::Init_w(TransportChannel* rtp_transport, | 212 bool BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| 214 TransportChannel* rtcp_transport) { | 213 DtlsTransportInternal* rtcp_dtls_transport) { |
| 215 if (!network_thread_->Invoke<bool>( | 214 if (!network_thread_->Invoke<bool>( |
| 216 RTC_FROM_HERE, Bind(&BaseChannel::InitNetwork_n, this, rtp_transport, | 215 RTC_FROM_HERE, Bind(&BaseChannel::InitNetwork_n, this, |
| 217 rtcp_transport))) { | 216 rtp_dtls_transport, rtcp_dtls_transport))) { |
| 218 return false; | 217 return false; |
| 219 } | 218 } |
| 220 | 219 |
| 221 // Both RTP and RTCP channels are set, we can call SetInterface on | 220 // Both RTP and RTCP channels are set, we can call SetInterface on |
| 222 // media channel and it can set network options. | 221 // media channel and it can set network options. |
| 223 RTC_DCHECK(worker_thread_->IsCurrent()); | 222 RTC_DCHECK(worker_thread_->IsCurrent()); |
| 224 media_channel_->SetInterface(this); | 223 media_channel_->SetInterface(this); |
| 225 return true; | 224 return true; |
| 226 } | 225 } |
| 227 | 226 |
| 228 bool BaseChannel::InitNetwork_n(TransportChannel* rtp_transport, | 227 bool BaseChannel::InitNetwork_n(DtlsTransportInternal* rtp_dtls_transport, |
| 229 TransportChannel* rtcp_transport) { | 228 DtlsTransportInternal* rtcp_dtls_transport) { |
| 230 RTC_DCHECK(network_thread_->IsCurrent()); | 229 RTC_DCHECK(network_thread_->IsCurrent()); |
| 231 SetTransports_n(rtp_transport, rtcp_transport); | 230 SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport); |
| 232 | 231 |
| 233 if (!SetDtlsSrtpCryptoSuites_n(rtp_transport_, false)) { | 232 if (!SetDtlsSrtpCryptoSuites_n(rtp_dtls_transport_, false)) { |
| 234 return false; | 233 return false; |
| 235 } | 234 } |
| 236 if (rtcp_transport_ && !SetDtlsSrtpCryptoSuites_n(rtcp_transport_, true)) { | 235 if (rtcp_dtls_transport_ && |
| 236 !SetDtlsSrtpCryptoSuites_n(rtcp_dtls_transport_, true)) { |
| 237 return false; | 237 return false; |
| 238 } | 238 } |
| 239 if (rtcp_mux_required_) { | 239 if (rtcp_mux_required_) { |
| 240 rtcp_mux_filter_.SetActive(); | 240 rtcp_mux_filter_.SetActive(); |
| 241 } | 241 } |
| 242 return true; | 242 return true; |
| 243 } | 243 } |
| 244 | 244 |
| 245 void BaseChannel::Deinit() { | 245 void BaseChannel::Deinit() { |
| 246 RTC_DCHECK(worker_thread_->IsCurrent()); | 246 RTC_DCHECK(worker_thread_->IsCurrent()); |
| 247 media_channel_->SetInterface(NULL); | 247 media_channel_->SetInterface(NULL); |
| 248 // Packets arrive on the network thread, processing packets calls virtual | 248 // Packets arrive on the network thread, processing packets calls virtual |
| 249 // functions, so need to stop this process in Deinit that is called in | 249 // functions, so need to stop this process in Deinit that is called in |
| 250 // derived classes destructor. | 250 // derived classes destructor. |
| 251 network_thread_->Invoke<void>( | 251 network_thread_->Invoke<void>( |
| 252 RTC_FROM_HERE, Bind(&BaseChannel::DisconnectTransportChannels_n, this)); | 252 RTC_FROM_HERE, Bind(&BaseChannel::DisconnectTransportChannels_n, this)); |
| 253 } | 253 } |
| 254 | 254 |
| 255 void BaseChannel::SetTransports(TransportChannel* rtp_transport, | 255 void BaseChannel::SetTransports(DtlsTransportInternal* rtp_dtls_transport, |
| 256 TransportChannel* rtcp_transport) { | 256 DtlsTransportInternal* rtcp_dtls_transport) { |
| 257 network_thread_->Invoke<void>( | 257 network_thread_->Invoke<void>(RTC_FROM_HERE, |
| 258 RTC_FROM_HERE, | 258 Bind(&BaseChannel::SetTransports_n, this, |
| 259 Bind(&BaseChannel::SetTransports_n, this, rtp_transport, rtcp_transport)); | 259 rtp_dtls_transport, rtcp_dtls_transport)); |
| 260 } | 260 } |
| 261 | 261 |
| 262 void BaseChannel::SetTransports_n(TransportChannel* rtp_transport, | 262 void BaseChannel::SetTransports_n(DtlsTransportInternal* rtp_dtls_transport, |
| 263 TransportChannel* rtcp_transport) { | 263 DtlsTransportInternal* rtcp_dtls_transport) { |
| 264 RTC_DCHECK(network_thread_->IsCurrent()); | 264 RTC_DCHECK(network_thread_->IsCurrent()); |
| 265 // Verify some assumptions (as described in the comment above SetTransport). | 265 if (!rtp_dtls_transport && !rtcp_dtls_transport) { |
| 266 RTC_DCHECK(rtp_transport); | 266 LOG(LS_ERROR) << "Setting nullptr to RTP Transport and RTCP Transport."; |
| 267 RTC_DCHECK(NeedsRtcpTransport() == (rtcp_transport != nullptr)); | 267 return; |
| 268 if (rtcp_transport) { | |
| 269 RTC_DCHECK(rtp_transport->transport_name() == | |
| 270 rtcp_transport->transport_name()); | |
| 271 } | 268 } |
| 272 | 269 |
| 273 if (rtp_transport->transport_name() == transport_name_) { | 270 if (rtp_dtls_transport && rtcp_dtls_transport) { |
| 271 RTC_DCHECK(rtp_dtls_transport->transport_name() == |
| 272 rtcp_dtls_transport->transport_name()); |
| 273 RTC_DCHECK(NeedsRtcpTransport()); |
| 274 } |
| 275 |
| 276 std::string transport_name = rtp_dtls_transport |
| 277 ? rtp_dtls_transport->transport_name() |
| 278 : rtcp_dtls_transport->transport_name(); |
| 279 if (transport_name == transport_name_) { |
| 274 // Nothing to do if transport name isn't changing. | 280 // Nothing to do if transport name isn't changing. |
| 275 return; | 281 return; |
| 276 } | 282 } |
| 277 | 283 |
| 278 transport_name_ = rtp_transport->transport_name(); | 284 transport_name_ = transport_name; |
| 279 | 285 |
| 280 // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport | 286 // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport |
| 281 // changes and wait until the DTLS handshake is complete to set the newly | 287 // changes and wait until the DTLS handshake is complete to set the newly |
| 282 // negotiated parameters. | 288 // negotiated parameters. |
| 283 if (ShouldSetupDtlsSrtp_n()) { | 289 if (ShouldSetupDtlsSrtp_n()) { |
| 284 // Set |writable_| to false such that UpdateWritableState_w can set up | 290 // Set |writable_| to false such that UpdateWritableState_w can set up |
| 285 // DTLS-SRTP when |writable_| becomes true again. | 291 // DTLS-SRTP when |writable_| becomes true again. |
| 286 writable_ = false; | 292 writable_ = false; |
| 287 srtp_filter_.ResetParams(); | 293 srtp_filter_.ResetParams(); |
| 288 } | 294 } |
| 289 | 295 |
| 290 // If this BaseChannel doesn't require RTCP mux and we haven't fully | 296 // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| 291 // negotiated RTCP mux, we need an RTCP transport. | 297 // negotiated RTCP mux, we need an RTCP transport. |
| 292 if (NeedsRtcpTransport()) { | 298 if (NeedsRtcpTransport()) { |
| 293 LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() << " on " | 299 LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() << " on " |
| 294 << transport_name() << " transport " << rtcp_transport; | 300 << transport_name << " transport " << rtcp_dtls_transport; |
| 295 SetTransportChannel_n(true, rtcp_transport); | 301 SetTransport_n(true, rtcp_dtls_transport); |
| 296 RTC_DCHECK(rtcp_transport_); | 302 RTC_DCHECK(rtcp_dtls_transport_); |
| 297 } | 303 } |
| 298 | 304 |
| 299 LOG(LS_INFO) << "Setting non-RTCP Transport for " << content_name() << " on " | 305 LOG(LS_INFO) << "Setting non-RTCP Transport for " << content_name() << " on " |
| 300 << transport_name() << " transport " << rtp_transport; | 306 << transport_name << " transport " << rtp_dtls_transport; |
| 301 SetTransportChannel_n(false, rtp_transport); | 307 SetTransport_n(false, rtp_dtls_transport); |
| 302 RTC_DCHECK(rtp_transport_); | 308 RTC_DCHECK(rtp_dtls_transport_); |
| 303 | 309 |
| 304 // Update aggregate writable/ready-to-send state between RTP and RTCP upon | 310 // Update aggregate writable/ready-to-send state between RTP and RTCP upon |
| 305 // setting new transport channels. | 311 // setting new transport channels. |
| 306 UpdateWritableState_n(); | 312 UpdateWritableState_n(); |
| 307 // We can only update ready-to-send after updating writability. | 313 // We can only update ready-to-send after updating writability. |
| 308 // | 314 // |
| 309 // On setting a new channel, assume it's ready to send if it's writable, | 315 // On setting a new channel, assume it's ready to send if it's writable, |
| 310 // because we have no way of knowing otherwise (the channel doesn't give us | 316 // because we have no way of knowing otherwise (the channel doesn't give us |
| 311 // "was last send successful?"). | 317 // "was last send successful?"). |
| 312 // | 318 // |
| 313 // This won't always be accurate (the last SendPacket call from another | 319 // This won't always be accurate (the last SendPacket call from another |
| 314 // BaseChannel could have resulted in an error), but even so, we'll just | 320 // BaseChannel could have resulted in an error), but even so, we'll just |
| 315 // encounter the error again and update "ready to send" accordingly. | 321 // encounter the error again and update "ready to send" accordingly. |
| 316 SetTransportChannelReadyToSend(false, | |
| 317 rtp_transport_ && rtp_transport_->writable()); | |
| 318 SetTransportChannelReadyToSend( | 322 SetTransportChannelReadyToSend( |
| 319 true, rtcp_transport_ && rtcp_transport_->writable()); | 323 false, rtp_dtls_transport_ && rtp_dtls_transport_->writable()); |
| 324 SetTransportChannelReadyToSend( |
| 325 true, rtcp_dtls_transport_ && rtcp_dtls_transport_->writable()); |
| 320 } | 326 } |
| 321 | 327 |
| 322 void BaseChannel::SetTransportChannel_n(bool rtcp, | 328 void BaseChannel::SetTransport_n(bool rtcp, |
| 323 TransportChannel* new_transport) { | 329 DtlsTransportInternal* new_transport) { |
| 324 RTC_DCHECK(network_thread_->IsCurrent()); | 330 RTC_DCHECK(network_thread_->IsCurrent()); |
| 325 TransportChannel*& old_transport = rtcp ? rtcp_transport_ : rtp_transport_; | 331 DtlsTransportInternal*& old_transport = |
| 332 rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
| 333 |
| 326 if (!old_transport && !new_transport) { | 334 if (!old_transport && !new_transport) { |
| 327 // Nothing to do. | 335 // Nothing to do. |
| 328 return; | 336 return; |
| 329 } | 337 } |
| 338 |
| 330 RTC_DCHECK(old_transport != new_transport); | 339 RTC_DCHECK(old_transport != new_transport); |
| 331 if (old_transport) { | 340 if (old_transport) { |
| 332 DisconnectFromTransportChannel(old_transport); | 341 DisconnectFromTransport(old_transport); |
| 333 } | 342 } |
| 334 | 343 |
| 335 old_transport = new_transport; | 344 old_transport = new_transport; |
| 336 | 345 |
| 337 if (new_transport) { | 346 if (new_transport) { |
| 338 if (rtcp) { | 347 if (rtcp) { |
| 339 RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_filter_.IsActive())) | 348 RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_filter_.IsActive())) |
| 340 << "Setting RTCP for DTLS/SRTP after SrtpFilter is active " | 349 << "Setting RTCP for DTLS/SRTP after SrtpFilter is active " |
| 341 << "should never happen."; | 350 << "should never happen."; |
| 342 } | 351 } |
| 343 ConnectToTransportChannel(new_transport); | 352 ConnectToTransport(new_transport); |
| 344 auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_; | 353 auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_; |
| 345 for (const auto& pair : socket_options) { | 354 for (const auto& pair : socket_options) { |
| 346 new_transport->SetOption(pair.first, pair.second); | 355 new_transport->SetOption(pair.first, pair.second); |
| 347 } | 356 } |
| 348 } | 357 } |
| 349 } | 358 } |
| 350 | 359 |
| 351 void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) { | 360 void BaseChannel::ConnectToTransport(DtlsTransportInternal* transport) { |
| 352 RTC_DCHECK(network_thread_->IsCurrent()); | 361 RTC_DCHECK(network_thread_->IsCurrent()); |
| 353 | 362 |
| 354 tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState); | 363 transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| 355 tc->SignalReadPacket.connect(this, &BaseChannel::OnPacketRead); | 364 transport->SignalReadPacket.connect(this, &BaseChannel::OnPacketRead); |
| 356 tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend); | 365 transport->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend); |
| 357 tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState); | 366 transport->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState); |
| 358 tc->SignalSelectedCandidatePairChanged.connect( | 367 transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| 368 transport->ice_transport()->SignalSelectedCandidatePairChanged.connect( |
| 359 this, &BaseChannel::OnSelectedCandidatePairChanged); | 369 this, &BaseChannel::OnSelectedCandidatePairChanged); |
| 360 tc->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); | |
| 361 } | 370 } |
| 362 | 371 |
| 363 void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) { | 372 void BaseChannel::DisconnectFromTransport(DtlsTransportInternal* transport) { |
| 364 RTC_DCHECK(network_thread_->IsCurrent()); | 373 RTC_DCHECK(network_thread_->IsCurrent()); |
| 365 OnSelectedCandidatePairChanged(tc, nullptr, -1, false); | 374 OnSelectedCandidatePairChanged(transport->ice_transport(), nullptr, -1, |
| 375 false); |
| 366 | 376 |
| 367 tc->SignalWritableState.disconnect(this); | 377 transport->SignalWritableState.disconnect(this); |
| 368 tc->SignalReadPacket.disconnect(this); | 378 transport->SignalReadPacket.disconnect(this); |
| 369 tc->SignalReadyToSend.disconnect(this); | 379 transport->SignalReadyToSend.disconnect(this); |
| 370 tc->SignalDtlsState.disconnect(this); | 380 transport->SignalDtlsState.disconnect(this); |
| 371 tc->SignalSelectedCandidatePairChanged.disconnect(this); | 381 transport->SignalSentPacket.disconnect(this); |
| 372 tc->SignalSentPacket.disconnect(this); | 382 transport->ice_transport()->SignalSelectedCandidatePairChanged.disconnect( |
| 383 this); |
| 373 } | 384 } |
| 374 | 385 |
| 375 bool BaseChannel::Enable(bool enable) { | 386 bool BaseChannel::Enable(bool enable) { |
| 376 worker_thread_->Invoke<void>( | 387 worker_thread_->Invoke<void>( |
| 377 RTC_FROM_HERE, | 388 RTC_FROM_HERE, |
| 378 Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w, | 389 Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w, |
| 379 this)); | 390 this)); |
| 380 return true; | 391 return true; |
| 381 } | 392 } |
| 382 | 393 |
| (...skipping 27 matching lines...) Expand all Loading... |
| 410 | 421 |
| 411 bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, | 422 bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, |
| 412 ContentAction action, | 423 ContentAction action, |
| 413 std::string* error_desc) { | 424 std::string* error_desc) { |
| 414 TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); | 425 TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); |
| 415 return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, | 426 return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, |
| 416 this, content, action, error_desc)); | 427 this, content, action, error_desc)); |
| 417 } | 428 } |
| 418 | 429 |
| 419 void BaseChannel::StartConnectionMonitor(int cms) { | 430 void BaseChannel::StartConnectionMonitor(int cms) { |
| 420 // We pass in the BaseChannel instead of the rtp_transport_ | 431 // We pass in the BaseChannel instead of the rtp_dtls_transport_ |
| 421 // because if the rtp_transport_ changes, the ConnectionMonitor | 432 // because if the rtp_dtls_transport_ changes, the ConnectionMonitor |
| 422 // would be pointing to the wrong TransportChannel. | 433 // would be pointing to the wrong TransportChannel. |
| 423 // We pass in the network thread because on that thread connection monitor | 434 // We pass in the network thread because on that thread connection monitor |
| 424 // will call BaseChannel::GetConnectionStats which must be called on the | 435 // will call BaseChannel::GetConnectionStats which must be called on the |
| 425 // network thread. | 436 // network thread. |
| 426 connection_monitor_.reset( | 437 connection_monitor_.reset( |
| 427 new ConnectionMonitor(this, network_thread(), rtc::Thread::Current())); | 438 new ConnectionMonitor(this, network_thread(), rtc::Thread::Current())); |
| 428 connection_monitor_->SignalUpdate.connect( | 439 connection_monitor_->SignalUpdate.connect( |
| 429 this, &BaseChannel::OnConnectionMonitorUpdate); | 440 this, &BaseChannel::OnConnectionMonitorUpdate); |
| 430 connection_monitor_->Start(cms); | 441 connection_monitor_->Start(cms); |
| 431 } | 442 } |
| 432 | 443 |
| 433 void BaseChannel::StopConnectionMonitor() { | 444 void BaseChannel::StopConnectionMonitor() { |
| 434 if (connection_monitor_) { | 445 if (connection_monitor_) { |
| 435 connection_monitor_->Stop(); | 446 connection_monitor_->Stop(); |
| 436 connection_monitor_.reset(); | 447 connection_monitor_.reset(); |
| 437 } | 448 } |
| 438 } | 449 } |
| 439 | 450 |
| 440 bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) { | 451 bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) { |
| 441 RTC_DCHECK(network_thread_->IsCurrent()); | 452 RTC_DCHECK(network_thread_->IsCurrent()); |
| 442 return rtp_transport_->GetStats(infos); | 453 return rtp_dtls_transport_->ice_transport()->GetStats(infos); |
| 443 } | 454 } |
| 444 | 455 |
| 445 bool BaseChannel::NeedsRtcpTransport() { | 456 bool BaseChannel::NeedsRtcpTransport() { |
| 446 // If this BaseChannel doesn't require RTCP mux and we haven't fully | 457 // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| 447 // negotiated RTCP mux, we need an RTCP transport. | 458 // negotiated RTCP mux, we need an RTCP transport. |
| 448 return !rtcp_mux_required_ && !rtcp_mux_filter_.IsFullyActive(); | 459 return !rtcp_mux_required_ && !rtcp_mux_filter_.IsFullyActive(); |
| 449 } | 460 } |
| 450 | 461 |
| 451 bool BaseChannel::IsReadyToReceiveMedia_w() const { | 462 bool BaseChannel::IsReadyToReceiveMedia_w() const { |
| 452 // Receive data if we are enabled and have local content, | 463 // Receive data if we are enabled and have local content, |
| (...skipping 28 matching lines...) Expand all Loading... |
| 481 int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, | 492 int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, |
| 482 int value) { | 493 int value) { |
| 483 return network_thread_->Invoke<int>( | 494 return network_thread_->Invoke<int>( |
| 484 RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value)); | 495 RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value)); |
| 485 } | 496 } |
| 486 | 497 |
| 487 int BaseChannel::SetOption_n(SocketType type, | 498 int BaseChannel::SetOption_n(SocketType type, |
| 488 rtc::Socket::Option opt, | 499 rtc::Socket::Option opt, |
| 489 int value) { | 500 int value) { |
| 490 RTC_DCHECK(network_thread_->IsCurrent()); | 501 RTC_DCHECK(network_thread_->IsCurrent()); |
| 491 TransportChannel* channel = nullptr; | 502 DtlsTransportInternal* transport = nullptr; |
| 492 switch (type) { | 503 switch (type) { |
| 493 case ST_RTP: | 504 case ST_RTP: |
| 494 channel = rtp_transport_; | 505 transport = rtp_dtls_transport_; |
| 495 socket_options_.push_back( | 506 socket_options_.push_back( |
| 496 std::pair<rtc::Socket::Option, int>(opt, value)); | 507 std::pair<rtc::Socket::Option, int>(opt, value)); |
| 497 break; | 508 break; |
| 498 case ST_RTCP: | 509 case ST_RTCP: |
| 499 channel = rtcp_transport_; | 510 transport = rtcp_dtls_transport_; |
| 500 rtcp_socket_options_.push_back( | 511 rtcp_socket_options_.push_back( |
| 501 std::pair<rtc::Socket::Option, int>(opt, value)); | 512 std::pair<rtc::Socket::Option, int>(opt, value)); |
| 502 break; | 513 break; |
| 503 } | 514 } |
| 504 return channel ? channel->SetOption(opt, value) : -1; | 515 return transport ? transport->ice_transport()->SetOption(opt, value) : -1; |
| 505 } | 516 } |
| 506 | 517 |
| 507 bool BaseChannel::SetCryptoOptions(const rtc::CryptoOptions& crypto_options) { | 518 bool BaseChannel::SetCryptoOptions(const rtc::CryptoOptions& crypto_options) { |
| 508 crypto_options_ = crypto_options; | 519 crypto_options_ = crypto_options; |
| 509 return true; | 520 return true; |
| 510 } | 521 } |
| 511 | 522 |
| 512 void BaseChannel::OnWritableState(rtc::PacketTransportInterface* transport) { | 523 void BaseChannel::OnWritableState(rtc::PacketTransportInterface* transport) { |
| 513 RTC_DCHECK(transport == rtp_transport_ || transport == rtcp_transport_); | 524 RTC_DCHECK(transport == rtp_dtls_transport_ || |
| 525 transport == rtcp_dtls_transport_); |
| 514 RTC_DCHECK(network_thread_->IsCurrent()); | 526 RTC_DCHECK(network_thread_->IsCurrent()); |
| 515 UpdateWritableState_n(); | 527 UpdateWritableState_n(); |
| 516 } | 528 } |
| 517 | 529 |
| 518 void BaseChannel::OnPacketRead(rtc::PacketTransportInterface* transport, | 530 void BaseChannel::OnPacketRead(rtc::PacketTransportInterface* transport, |
| 519 const char* data, | 531 const char* data, |
| 520 size_t len, | 532 size_t len, |
| 521 const rtc::PacketTime& packet_time, | 533 const rtc::PacketTime& packet_time, |
| 522 int flags) { | 534 int flags) { |
| 523 TRACE_EVENT0("webrtc", "BaseChannel::OnPacketRead"); | 535 TRACE_EVENT0("webrtc", "BaseChannel::OnPacketRead"); |
| 524 // OnPacketRead gets called from P2PSocket; now pass data to MediaEngine | 536 // OnPacketRead gets called from P2PSocket; now pass data to MediaEngine |
| 525 RTC_DCHECK(network_thread_->IsCurrent()); | 537 RTC_DCHECK(network_thread_->IsCurrent()); |
| 526 | 538 |
| 527 // When using RTCP multiplexing we might get RTCP packets on the RTP | 539 // When using RTCP multiplexing we might get RTCP packets on the RTP |
| 528 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP. | 540 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP. |
| 529 bool rtcp = PacketIsRtcp(transport, data, len); | 541 bool rtcp = PacketIsRtcp(transport, data, len); |
| 530 rtc::CopyOnWriteBuffer packet(data, len); | 542 rtc::CopyOnWriteBuffer packet(data, len); |
| 531 HandlePacket(rtcp, &packet, packet_time); | 543 HandlePacket(rtcp, &packet, packet_time); |
| 532 } | 544 } |
| 533 | 545 |
| 534 void BaseChannel::OnReadyToSend(rtc::PacketTransportInterface* transport) { | 546 void BaseChannel::OnReadyToSend(rtc::PacketTransportInterface* transport) { |
| 535 RTC_DCHECK(transport == rtp_transport_ || transport == rtcp_transport_); | 547 RTC_DCHECK(transport == rtp_dtls_transport_ || |
| 536 SetTransportChannelReadyToSend(transport == rtcp_transport_, true); | 548 transport == rtcp_dtls_transport_); |
| 549 SetTransportChannelReadyToSend(transport == rtcp_dtls_transport_, true); |
| 537 } | 550 } |
| 538 | 551 |
| 539 void BaseChannel::OnDtlsState(TransportChannel* channel, | 552 void BaseChannel::OnDtlsState(DtlsTransportInternal* transport, |
| 540 DtlsTransportState state) { | 553 DtlsTransportState state) { |
| 541 if (!ShouldSetupDtlsSrtp_n()) { | 554 if (!ShouldSetupDtlsSrtp_n()) { |
| 542 return; | 555 return; |
| 543 } | 556 } |
| 544 | 557 |
| 545 // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED | 558 // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED |
| 546 // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to | 559 // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to |
| 547 // cover other scenarios like the whole channel is writable (not just this | 560 // cover other scenarios like the whole transport is writable (not just this |
| 548 // TransportChannel) or when TransportChannel is attached after DTLS is | 561 // TransportChannel) or when TransportChannel is attached after DTLS is |
| 549 // negotiated. | 562 // negotiated. |
| 550 if (state != DTLS_TRANSPORT_CONNECTED) { | 563 if (state != DTLS_TRANSPORT_CONNECTED) { |
| 551 srtp_filter_.ResetParams(); | 564 srtp_filter_.ResetParams(); |
| 552 } | 565 } |
| 553 } | 566 } |
| 554 | 567 |
| 555 void BaseChannel::OnSelectedCandidatePairChanged( | 568 void BaseChannel::OnSelectedCandidatePairChanged( |
| 556 TransportChannel* channel, | 569 IceTransportInternal* ice_transport, |
| 557 CandidatePairInterface* selected_candidate_pair, | 570 CandidatePairInterface* selected_candidate_pair, |
| 558 int last_sent_packet_id, | 571 int last_sent_packet_id, |
| 559 bool ready_to_send) { | 572 bool ready_to_send) { |
| 560 RTC_DCHECK(channel == rtp_transport_ || channel == rtcp_transport_); | 573 RTC_DCHECK(ice_transport == rtp_dtls_transport_->ice_transport() || |
| 574 ice_transport == rtcp_dtls_transport_->ice_transport()); |
| 561 RTC_DCHECK(network_thread_->IsCurrent()); | 575 RTC_DCHECK(network_thread_->IsCurrent()); |
| 562 selected_candidate_pair_ = selected_candidate_pair; | 576 selected_candidate_pair_ = selected_candidate_pair; |
| 563 std::string transport_name = channel->transport_name(); | 577 std::string transport_name = ice_transport->transport_name(); |
| 564 rtc::NetworkRoute network_route; | 578 rtc::NetworkRoute network_route; |
| 565 if (selected_candidate_pair) { | 579 if (selected_candidate_pair) { |
| 566 network_route = rtc::NetworkRoute( | 580 network_route = rtc::NetworkRoute( |
| 567 ready_to_send, selected_candidate_pair->local_candidate().network_id(), | 581 ready_to_send, selected_candidate_pair->local_candidate().network_id(), |
| 568 selected_candidate_pair->remote_candidate().network_id(), | 582 selected_candidate_pair->remote_candidate().network_id(), |
| 569 last_sent_packet_id); | 583 last_sent_packet_id); |
| 570 | 584 |
| 571 UpdateTransportOverhead(); | 585 UpdateTransportOverhead(); |
| 572 } | 586 } |
| 573 invoker_.AsyncInvoke<void>( | 587 invoker_.AsyncInvoke<void>( |
| 574 RTC_FROM_HERE, worker_thread_, | 588 RTC_FROM_HERE, worker_thread_, |
| 575 Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_, transport_name, | 589 Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_, transport_name, |
| 576 network_route)); | 590 network_route)); |
| 577 } | 591 } |
| 578 | 592 |
| 579 void BaseChannel::SetTransportChannelReadyToSend(bool rtcp, bool ready) { | 593 void BaseChannel::SetTransportChannelReadyToSend(bool rtcp, bool ready) { |
| 580 RTC_DCHECK(network_thread_->IsCurrent()); | 594 RTC_DCHECK(network_thread_->IsCurrent()); |
| 581 if (rtcp) { | 595 if (rtcp) { |
| 582 rtcp_ready_to_send_ = ready; | 596 rtcp_ready_to_send_ = ready; |
| 583 } else { | 597 } else { |
| 584 rtp_ready_to_send_ = ready; | 598 rtp_ready_to_send_ = ready; |
| 585 } | 599 } |
| 586 | 600 |
| 587 bool ready_to_send = | 601 bool ready_to_send = |
| 588 (rtp_ready_to_send_ && | 602 (rtp_ready_to_send_ && |
| 589 // In the case of rtcp mux |rtcp_transport_| will be null. | 603 // In the case of rtcp mux |rtcp_dtls_transport_| will be null. |
| 590 (rtcp_ready_to_send_ || !rtcp_transport_)); | 604 (rtcp_ready_to_send_ || !rtcp_dtls_transport_)); |
| 591 | 605 |
| 592 invoker_.AsyncInvoke<void>( | 606 invoker_.AsyncInvoke<void>( |
| 593 RTC_FROM_HERE, worker_thread_, | 607 RTC_FROM_HERE, worker_thread_, |
| 594 Bind(&MediaChannel::OnReadyToSend, media_channel_, ready_to_send)); | 608 Bind(&MediaChannel::OnReadyToSend, media_channel_, ready_to_send)); |
| 595 } | 609 } |
| 596 | 610 |
| 597 bool BaseChannel::PacketIsRtcp(const rtc::PacketTransportInterface* transport, | 611 bool BaseChannel::PacketIsRtcp(const rtc::PacketTransportInterface* transport, |
| 598 const char* data, | 612 const char* data, |
| 599 size_t len) { | 613 size_t len) { |
| 600 return (transport == rtcp_transport_ || | 614 return (transport == rtcp_dtls_transport_ || |
| 601 rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len))); | 615 rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len))); |
| 602 } | 616 } |
| 603 | 617 |
| 604 bool BaseChannel::SendPacket(bool rtcp, | 618 bool BaseChannel::SendPacket(bool rtcp, |
| 605 rtc::CopyOnWriteBuffer* packet, | 619 rtc::CopyOnWriteBuffer* packet, |
| 606 const rtc::PacketOptions& options) { | 620 const rtc::PacketOptions& options) { |
| 607 // SendPacket gets called from MediaEngine, on a pacer or an encoder thread. | 621 // SendPacket gets called from MediaEngine, on a pacer or an encoder thread. |
| 608 // If the thread is not our network thread, we will post to our network | 622 // If the thread is not our network thread, we will post to our network |
| 609 // so that the real work happens on our network. This avoids us having to | 623 // so that the real work happens on our network. This avoids us having to |
| 610 // synchronize access to all the pieces of the send path, including | 624 // synchronize access to all the pieces of the send path, including |
| 611 // SRTP and the inner workings of the transport channels. | 625 // SRTP and the inner workings of the transport channels. |
| 612 // The only downside is that we can't return a proper failure code if | 626 // The only downside is that we can't return a proper failure code if |
| 613 // needed. Since UDP is unreliable anyway, this should be a non-issue. | 627 // needed. Since UDP is unreliable anyway, this should be a non-issue. |
| 614 if (!network_thread_->IsCurrent()) { | 628 if (!network_thread_->IsCurrent()) { |
| 615 // Avoid a copy by transferring the ownership of the packet data. | 629 // Avoid a copy by transferring the ownership of the packet data. |
| 616 int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET; | 630 int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET; |
| 617 SendPacketMessageData* data = new SendPacketMessageData; | 631 SendPacketMessageData* data = new SendPacketMessageData; |
| 618 data->packet = std::move(*packet); | 632 data->packet = std::move(*packet); |
| 619 data->options = options; | 633 data->options = options; |
| 620 network_thread_->Post(RTC_FROM_HERE, this, message_id, data); | 634 network_thread_->Post(RTC_FROM_HERE, this, message_id, data); |
| 621 return true; | 635 return true; |
| 622 } | 636 } |
| 623 TRACE_EVENT0("webrtc", "BaseChannel::SendPacket"); | 637 TRACE_EVENT0("webrtc", "BaseChannel::SendPacket"); |
| 624 | 638 |
| 625 // Now that we are on the correct thread, ensure we have a place to send this | 639 // Now that we are on the correct thread, ensure we have a place to send this |
| 626 // packet before doing anything. (We might get RTCP packets that we don't | 640 // packet before doing anything. (We might get RTCP packets that we don't |
| 627 // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP | 641 // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP |
| 628 // transport. | 642 // transport. |
| 629 TransportChannel* channel = | 643 DtlsTransportInternal* transport = (!rtcp || rtcp_mux_filter_.IsActive()) |
| 630 (!rtcp || rtcp_mux_filter_.IsActive()) ? rtp_transport_ : rtcp_transport_; | 644 ? rtp_dtls_transport_ |
| 631 if (!channel || !channel->writable()) { | 645 : rtcp_dtls_transport_; |
| 646 if (!transport || !transport->writable()) { |
| 632 return false; | 647 return false; |
| 633 } | 648 } |
| 634 | 649 |
| 635 // Protect ourselves against crazy data. | 650 // Protect ourselves against crazy data. |
| 636 if (!ValidPacket(rtcp, packet)) { | 651 if (!ValidPacket(rtcp, packet)) { |
| 637 LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " " | 652 LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " " |
| 638 << PacketType(rtcp) | 653 << PacketType(rtcp) |
| 639 << " packet: wrong size=" << packet->size(); | 654 << " packet: wrong size=" << packet->size(); |
| 640 return false; | 655 return false; |
| 641 } | 656 } |
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| 712 // However, there shouldn't be any RTP packets sent before SRTP is set up | 727 // However, there shouldn't be any RTP packets sent before SRTP is set up |
| 713 // (and SetSend(true) is called). | 728 // (and SetSend(true) is called). |
| 714 LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive" | 729 LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive" |
| 715 << " and crypto is required"; | 730 << " and crypto is required"; |
| 716 RTC_NOTREACHED(); | 731 RTC_NOTREACHED(); |
| 717 return false; | 732 return false; |
| 718 } | 733 } |
| 719 | 734 |
| 720 // Bon voyage. | 735 // Bon voyage. |
| 721 int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL; | 736 int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL; |
| 722 int ret = channel->SendPacket(packet->data<char>(), packet->size(), | 737 int ret = transport->SendPacket(packet->data<char>(), packet->size(), |
| 723 updated_options, flags); | 738 updated_options, flags); |
| 724 if (ret != static_cast<int>(packet->size())) { | 739 if (ret != static_cast<int>(packet->size())) { |
| 725 if (channel->GetError() == ENOTCONN) { | 740 if (transport->GetError() == ENOTCONN) { |
| 726 LOG(LS_WARNING) << "Got ENOTCONN from transport."; | 741 LOG(LS_WARNING) << "Got ENOTCONN from transport."; |
| 727 SetTransportChannelReadyToSend(rtcp, false); | 742 SetTransportChannelReadyToSend(rtcp, false); |
| 728 } | 743 } |
| 729 return false; | 744 return false; |
| 730 } | 745 } |
| 731 return true; | 746 return true; |
| 732 } | 747 } |
| 733 | 748 |
| 734 bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { | 749 bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
| 735 // Protect ourselves against crazy data. | 750 // Protect ourselves against crazy data. |
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| 790 } | 805 } |
| 791 } | 806 } |
| 792 | 807 |
| 793 packet->SetSize(len); | 808 packet->SetSize(len); |
| 794 } else if (srtp_required_) { | 809 } else if (srtp_required_) { |
| 795 // Our session description indicates that SRTP is required, but we got a | 810 // Our session description indicates that SRTP is required, but we got a |
| 796 // packet before our SRTP filter is active. This means either that | 811 // packet before our SRTP filter is active. This means either that |
| 797 // a) we got SRTP packets before we received the SDES keys, in which case | 812 // a) we got SRTP packets before we received the SDES keys, in which case |
| 798 // we can't decrypt it anyway, or | 813 // we can't decrypt it anyway, or |
| 799 // b) we got SRTP packets before DTLS completed on both the RTP and RTCP | 814 // b) we got SRTP packets before DTLS completed on both the RTP and RTCP |
| 800 // channels, so we haven't yet extracted keys, even if DTLS did complete | 815 // transports, so we haven't yet extracted keys, even if DTLS did |
| 801 // on the channel that the packets are being sent on. It's really good | 816 // complete on the transport that the packets are being sent on. It's |
| 802 // practice to wait for both RTP and RTCP to be good to go before sending | 817 // really good practice to wait for both RTP and RTCP to be good to go |
| 803 // media, to prevent weird failure modes, so it's fine for us to just eat | 818 // before sending media, to prevent weird failure modes, so it's fine |
| 804 // packets here. This is all sidestepped if RTCP mux is used anyway. | 819 // for us to just eat packets here. This is all sidestepped if RTCP mux |
| 820 // is used anyway. |
| 805 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp) | 821 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp) |
| 806 << " packet when SRTP is inactive and crypto is required"; | 822 << " packet when SRTP is inactive and crypto is required"; |
| 807 return; | 823 return; |
| 808 } | 824 } |
| 809 | 825 |
| 810 invoker_.AsyncInvoke<void>( | 826 invoker_.AsyncInvoke<void>( |
| 811 RTC_FROM_HERE, worker_thread_, | 827 RTC_FROM_HERE, worker_thread_, |
| 812 Bind(&BaseChannel::OnPacketReceived, this, rtcp, *packet, packet_time)); | 828 Bind(&BaseChannel::OnPacketReceived, this, rtcp, *packet, packet_time)); |
| 813 } | 829 } |
| 814 | 830 |
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| 868 RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); | 884 RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| 869 if (!enabled_) | 885 if (!enabled_) |
| 870 return; | 886 return; |
| 871 | 887 |
| 872 LOG(LS_INFO) << "Channel disabled"; | 888 LOG(LS_INFO) << "Channel disabled"; |
| 873 enabled_ = false; | 889 enabled_ = false; |
| 874 UpdateMediaSendRecvState_w(); | 890 UpdateMediaSendRecvState_w(); |
| 875 } | 891 } |
| 876 | 892 |
| 877 void BaseChannel::UpdateWritableState_n() { | 893 void BaseChannel::UpdateWritableState_n() { |
| 878 if (rtp_transport_ && rtp_transport_->writable() && | 894 if (rtp_dtls_transport_ && rtp_dtls_transport_->writable() && |
| 879 (!rtcp_transport_ || rtcp_transport_->writable())) { | 895 (!rtcp_dtls_transport_ || rtcp_dtls_transport_->writable())) { |
| 880 ChannelWritable_n(); | 896 ChannelWritable_n(); |
| 881 } else { | 897 } else { |
| 882 ChannelNotWritable_n(); | 898 ChannelNotWritable_n(); |
| 883 } | 899 } |
| 884 } | 900 } |
| 885 | 901 |
| 886 void BaseChannel::ChannelWritable_n() { | 902 void BaseChannel::ChannelWritable_n() { |
| 887 RTC_DCHECK(network_thread_->IsCurrent()); | 903 RTC_DCHECK(network_thread_->IsCurrent()); |
| 888 if (writable_) { | 904 if (writable_) { |
| 889 return; | 905 return; |
| (...skipping 20 matching lines...) Expand all Loading... |
| 910 invoker_.AsyncInvoke<void>( | 926 invoker_.AsyncInvoke<void>( |
| 911 RTC_FROM_HERE, signaling_thread(), | 927 RTC_FROM_HERE, signaling_thread(), |
| 912 Bind(&BaseChannel::SignalDtlsSrtpSetupFailure_s, this, rtcp)); | 928 Bind(&BaseChannel::SignalDtlsSrtpSetupFailure_s, this, rtcp)); |
| 913 } | 929 } |
| 914 | 930 |
| 915 void BaseChannel::SignalDtlsSrtpSetupFailure_s(bool rtcp) { | 931 void BaseChannel::SignalDtlsSrtpSetupFailure_s(bool rtcp) { |
| 916 RTC_DCHECK(signaling_thread() == rtc::Thread::Current()); | 932 RTC_DCHECK(signaling_thread() == rtc::Thread::Current()); |
| 917 SignalDtlsSrtpSetupFailure(this, rtcp); | 933 SignalDtlsSrtpSetupFailure(this, rtcp); |
| 918 } | 934 } |
| 919 | 935 |
| 920 bool BaseChannel::SetDtlsSrtpCryptoSuites_n(TransportChannel* tc, bool rtcp) { | 936 bool BaseChannel::SetDtlsSrtpCryptoSuites_n(DtlsTransportInternal* transport, |
| 937 bool rtcp) { |
| 921 std::vector<int> crypto_suites; | 938 std::vector<int> crypto_suites; |
| 922 // We always use the default SRTP crypto suites for RTCP, but we may use | 939 // We always use the default SRTP crypto suites for RTCP, but we may use |
| 923 // different crypto suites for RTP depending on the media type. | 940 // different crypto suites for RTP depending on the media type. |
| 924 if (!rtcp) { | 941 if (!rtcp) { |
| 925 GetSrtpCryptoSuites_n(&crypto_suites); | 942 GetSrtpCryptoSuites_n(&crypto_suites); |
| 926 } else { | 943 } else { |
| 927 GetDefaultSrtpCryptoSuites(crypto_options(), &crypto_suites); | 944 GetDefaultSrtpCryptoSuites(crypto_options(), &crypto_suites); |
| 928 } | 945 } |
| 929 return tc->SetSrtpCryptoSuites(crypto_suites); | 946 return transport->SetSrtpCryptoSuites(crypto_suites); |
| 930 } | 947 } |
| 931 | 948 |
| 932 bool BaseChannel::ShouldSetupDtlsSrtp_n() const { | 949 bool BaseChannel::ShouldSetupDtlsSrtp_n() const { |
| 933 // Since DTLS is applied to all channels, checking RTP should be enough. | 950 // Since DTLS is applied to all transports, checking RTP should be enough. |
| 934 return rtp_transport_ && rtp_transport_->IsDtlsActive(); | 951 return rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive(); |
| 935 } | 952 } |
| 936 | 953 |
| 937 // This function returns true if either DTLS-SRTP is not in use | 954 // This function returns true if either DTLS-SRTP is not in use |
| 938 // *or* DTLS-SRTP is successfully set up. | 955 // *or* DTLS-SRTP is successfully set up. |
| 939 bool BaseChannel::SetupDtlsSrtp_n(bool rtcp_channel) { | 956 bool BaseChannel::SetupDtlsSrtp_n(bool rtcp) { |
| 940 RTC_DCHECK(network_thread_->IsCurrent()); | 957 RTC_DCHECK(network_thread_->IsCurrent()); |
| 941 bool ret = false; | 958 bool ret = false; |
| 942 | 959 |
| 943 TransportChannel* channel = rtcp_channel ? rtcp_transport_ : rtp_transport_; | 960 DtlsTransportInternal* transport = |
| 961 rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
| 944 | 962 |
| 945 RTC_DCHECK(channel->IsDtlsActive()); | 963 RTC_DCHECK(transport->IsDtlsActive()); |
| 946 | 964 |
| 947 int selected_crypto_suite; | 965 int selected_crypto_suite; |
| 948 | 966 |
| 949 if (!channel->GetSrtpCryptoSuite(&selected_crypto_suite)) { | 967 if (!transport->GetSrtpCryptoSuite(&selected_crypto_suite)) { |
| 950 LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite"; | 968 LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite"; |
| 951 return false; | 969 return false; |
| 952 } | 970 } |
| 953 | 971 |
| 954 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " | 972 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " << content_name() << " " |
| 955 << content_name() << " " | 973 << PacketType(rtcp); |
| 956 << PacketType(rtcp_channel); | |
| 957 | 974 |
| 958 int key_len; | 975 int key_len; |
| 959 int salt_len; | 976 int salt_len; |
| 960 if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len, | 977 if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len, |
| 961 &salt_len)) { | 978 &salt_len)) { |
| 962 LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite" << selected_crypto_suite; | 979 LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite" << selected_crypto_suite; |
| 963 return false; | 980 return false; |
| 964 } | 981 } |
| 965 | 982 |
| 966 // OK, we're now doing DTLS (RFC 5764) | 983 // OK, we're now doing DTLS (RFC 5764) |
| 967 std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2); | 984 std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2); |
| 968 | 985 |
| 969 // RFC 5705 exporter using the RFC 5764 parameters | 986 // RFC 5705 exporter using the RFC 5764 parameters |
| 970 if (!channel->ExportKeyingMaterial( | 987 if (!transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0, false, |
| 971 kDtlsSrtpExporterLabel, | 988 &dtls_buffer[0], dtls_buffer.size())) { |
| 972 NULL, 0, false, | |
| 973 &dtls_buffer[0], dtls_buffer.size())) { | |
| 974 LOG(LS_WARNING) << "DTLS-SRTP key export failed"; | 989 LOG(LS_WARNING) << "DTLS-SRTP key export failed"; |
| 975 RTC_NOTREACHED(); // This should never happen | 990 RTC_NOTREACHED(); // This should never happen |
| 976 return false; | 991 return false; |
| 977 } | 992 } |
| 978 | 993 |
| 979 // Sync up the keys with the DTLS-SRTP interface | 994 // Sync up the keys with the DTLS-SRTP interface |
| 980 std::vector<unsigned char> client_write_key(key_len + salt_len); | 995 std::vector<unsigned char> client_write_key(key_len + salt_len); |
| 981 std::vector<unsigned char> server_write_key(key_len + salt_len); | 996 std::vector<unsigned char> server_write_key(key_len + salt_len); |
| 982 size_t offset = 0; | 997 size_t offset = 0; |
| 983 memcpy(&client_write_key[0], &dtls_buffer[offset], key_len); | 998 memcpy(&client_write_key[0], &dtls_buffer[offset], key_len); |
| 984 offset += key_len; | 999 offset += key_len; |
| 985 memcpy(&server_write_key[0], &dtls_buffer[offset], key_len); | 1000 memcpy(&server_write_key[0], &dtls_buffer[offset], key_len); |
| 986 offset += key_len; | 1001 offset += key_len; |
| 987 memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len); | 1002 memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len); |
| 988 offset += salt_len; | 1003 offset += salt_len; |
| 989 memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len); | 1004 memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len); |
| 990 | 1005 |
| 991 std::vector<unsigned char> *send_key, *recv_key; | 1006 std::vector<unsigned char> *send_key, *recv_key; |
| 992 rtc::SSLRole role; | 1007 rtc::SSLRole role; |
| 993 if (!channel->GetSslRole(&role)) { | 1008 if (!transport->GetSslRole(&role)) { |
| 994 LOG(LS_WARNING) << "GetSslRole failed"; | 1009 LOG(LS_WARNING) << "GetSslRole failed"; |
| 995 return false; | 1010 return false; |
| 996 } | 1011 } |
| 997 | 1012 |
| 998 if (role == rtc::SSL_SERVER) { | 1013 if (role == rtc::SSL_SERVER) { |
| 999 send_key = &server_write_key; | 1014 send_key = &server_write_key; |
| 1000 recv_key = &client_write_key; | 1015 recv_key = &client_write_key; |
| 1001 } else { | 1016 } else { |
| 1002 send_key = &client_write_key; | 1017 send_key = &client_write_key; |
| 1003 recv_key = &server_write_key; | 1018 recv_key = &server_write_key; |
| 1004 } | 1019 } |
| 1005 | 1020 |
| 1006 if (rtcp_channel) { | 1021 if (rtcp) { |
| 1007 ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0], | 1022 ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0], |
| 1008 static_cast<int>(send_key->size()), | 1023 static_cast<int>(send_key->size()), |
| 1009 selected_crypto_suite, &(*recv_key)[0], | 1024 selected_crypto_suite, &(*recv_key)[0], |
| 1010 static_cast<int>(recv_key->size())); | 1025 static_cast<int>(recv_key->size())); |
| 1011 } else { | 1026 } else { |
| 1012 ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0], | 1027 ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0], |
| 1013 static_cast<int>(send_key->size()), | 1028 static_cast<int>(send_key->size()), |
| 1014 selected_crypto_suite, &(*recv_key)[0], | 1029 selected_crypto_suite, &(*recv_key)[0], |
| 1015 static_cast<int>(recv_key->size())); | 1030 static_cast<int>(recv_key->size())); |
| 1016 } | 1031 } |
| (...skipping 14 matching lines...) Expand all Loading... |
| 1031 | 1046 |
| 1032 if (!ShouldSetupDtlsSrtp_n()) { | 1047 if (!ShouldSetupDtlsSrtp_n()) { |
| 1033 return; | 1048 return; |
| 1034 } | 1049 } |
| 1035 | 1050 |
| 1036 if (!SetupDtlsSrtp_n(false)) { | 1051 if (!SetupDtlsSrtp_n(false)) { |
| 1037 SignalDtlsSrtpSetupFailure_n(false); | 1052 SignalDtlsSrtpSetupFailure_n(false); |
| 1038 return; | 1053 return; |
| 1039 } | 1054 } |
| 1040 | 1055 |
| 1041 if (rtcp_transport_) { | 1056 if (rtcp_dtls_transport_) { |
| 1042 if (!SetupDtlsSrtp_n(true)) { | 1057 if (!SetupDtlsSrtp_n(true)) { |
| 1043 SignalDtlsSrtpSetupFailure_n(true); | 1058 SignalDtlsSrtpSetupFailure_n(true); |
| 1044 return; | 1059 return; |
| 1045 } | 1060 } |
| 1046 } | 1061 } |
| 1047 } | 1062 } |
| 1048 | 1063 |
| 1049 void BaseChannel::ChannelNotWritable_n() { | 1064 void BaseChannel::ChannelNotWritable_n() { |
| 1050 RTC_DCHECK(network_thread_->IsCurrent()); | 1065 RTC_DCHECK(network_thread_->IsCurrent()); |
| 1051 if (!writable_) | 1066 if (!writable_) |
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| 1083 return false; | 1098 return false; |
| 1084 } | 1099 } |
| 1085 | 1100 |
| 1086 if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) { | 1101 if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) { |
| 1087 return false; | 1102 return false; |
| 1088 } | 1103 } |
| 1089 | 1104 |
| 1090 return true; | 1105 return true; |
| 1091 } | 1106 } |
| 1092 | 1107 |
| 1093 // |dtls| will be set to true if DTLS is active for transport channel and | 1108 // |dtls| will be set to true if DTLS is active for transport and crypto is |
| 1094 // crypto is empty. | 1109 // empty. |
| 1095 bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos, | 1110 bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos, |
| 1096 bool* dtls, | 1111 bool* dtls, |
| 1097 std::string* error_desc) { | 1112 std::string* error_desc) { |
| 1098 *dtls = rtp_transport_->IsDtlsActive(); | 1113 *dtls = rtp_dtls_transport_->IsDtlsActive(); |
| 1099 if (*dtls && !cryptos.empty()) { | 1114 if (*dtls && !cryptos.empty()) { |
| 1100 SafeSetError("Cryptos must be empty when DTLS is active.", error_desc); | 1115 SafeSetError("Cryptos must be empty when DTLS is active.", error_desc); |
| 1101 return false; | 1116 return false; |
| 1102 } | 1117 } |
| 1103 return true; | 1118 return true; |
| 1104 } | 1119 } |
| 1105 | 1120 |
| 1106 bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos, | 1121 bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos, |
| 1107 ContentAction action, | 1122 ContentAction action, |
| 1108 ContentSource src, | 1123 ContentSource src, |
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| 1162 "contain 'a=rtcp-mux'.", | 1177 "contain 'a=rtcp-mux'.", |
| 1163 error_desc); | 1178 error_desc); |
| 1164 return false; | 1179 return false; |
| 1165 } | 1180 } |
| 1166 bool ret = false; | 1181 bool ret = false; |
| 1167 switch (action) { | 1182 switch (action) { |
| 1168 case CA_OFFER: | 1183 case CA_OFFER: |
| 1169 ret = rtcp_mux_filter_.SetOffer(enable, src); | 1184 ret = rtcp_mux_filter_.SetOffer(enable, src); |
| 1170 break; | 1185 break; |
| 1171 case CA_PRANSWER: | 1186 case CA_PRANSWER: |
| 1172 // This may activate RTCP muxing, but we don't yet destroy the channel | 1187 // This may activate RTCP muxing, but we don't yet destroy the transport |
| 1173 // because the final answer may deactivate it. | 1188 // because the final answer may deactivate it. |
| 1174 ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src); | 1189 ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src); |
| 1175 break; | 1190 break; |
| 1176 case CA_ANSWER: | 1191 case CA_ANSWER: |
| 1177 ret = rtcp_mux_filter_.SetAnswer(enable, src); | 1192 ret = rtcp_mux_filter_.SetAnswer(enable, src); |
| 1178 if (ret && rtcp_mux_filter_.IsActive()) { | 1193 if (ret && rtcp_mux_filter_.IsActive()) { |
| 1179 // We activated RTCP mux, close down the RTCP transport. | 1194 // We activated RTCP mux, close down the RTCP transport. |
| 1180 LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name() | 1195 LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name() |
| 1181 << " by destroying RTCP transport channel for " | 1196 << " by destroying RTCP transport for " |
| 1182 << transport_name(); | 1197 << transport_name(); |
| 1183 if (rtcp_transport()) { | 1198 if (rtcp_dtls_transport()) { |
| 1184 SetTransportChannel_n(true, nullptr); | 1199 SetTransport_n(true, nullptr); |
| 1185 SignalRtcpMuxFullyActive(rtp_transport()->transport_name()); | 1200 SignalRtcpMuxFullyActive(rtp_dtls_transport()->transport_name()); |
| 1186 } | 1201 } |
| 1187 UpdateWritableState_n(); | 1202 UpdateWritableState_n(); |
| 1188 SetTransportChannelReadyToSend(true, false); | 1203 SetTransportChannelReadyToSend(true, false); |
| 1189 } | 1204 } |
| 1190 break; | 1205 break; |
| 1191 case CA_UPDATE: | 1206 case CA_UPDATE: |
| 1192 // No RTCP mux info. | 1207 // No RTCP mux info. |
| 1193 ret = true; | 1208 ret = true; |
| 1194 break; | 1209 break; |
| 1195 default: | 1210 default: |
| 1196 break; | 1211 break; |
| 1197 } | 1212 } |
| 1198 if (!ret) { | 1213 if (!ret) { |
| 1199 SafeSetError("Failed to setup RTCP mux filter.", error_desc); | 1214 SafeSetError("Failed to setup RTCP mux filter.", error_desc); |
| 1200 return false; | 1215 return false; |
| 1201 } | 1216 } |
| 1202 // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or | 1217 // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or |
| 1203 // CA_ANSWER, but we only want to tear down the RTCP transport channel if we | 1218 // CA_ANSWER, but we only want to tear down the RTCP transport if we received |
| 1204 // received a final answer. | 1219 // a final answer. |
| 1205 if (rtcp_mux_filter_.IsActive()) { | 1220 if (rtcp_mux_filter_.IsActive()) { |
| 1206 // If the RTP transport is already writable, then so are we. | 1221 // If the RTP transport is already writable, then so are we. |
| 1207 if (rtp_transport_->writable()) { | 1222 if (rtp_dtls_transport_->writable()) { |
| 1208 ChannelWritable_n(); | 1223 ChannelWritable_n(); |
| 1209 } | 1224 } |
| 1210 } | 1225 } |
| 1211 | 1226 |
| 1212 return true; | 1227 return true; |
| 1213 } | 1228 } |
| 1214 | 1229 |
| 1215 bool BaseChannel::AddRecvStream_w(const StreamParams& sp) { | 1230 bool BaseChannel::AddRecvStream_w(const StreamParams& sp) { |
| 1216 RTC_DCHECK(worker_thread() == rtc::Thread::Current()); | 1231 RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| 1217 return media_channel()->AddRecvStream(sp); | 1232 return media_channel()->AddRecvStream(sp); |
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| 1457 | 1472 |
| 1458 VoiceChannel::~VoiceChannel() { | 1473 VoiceChannel::~VoiceChannel() { |
| 1459 TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); | 1474 TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); |
| 1460 StopAudioMonitor(); | 1475 StopAudioMonitor(); |
| 1461 StopMediaMonitor(); | 1476 StopMediaMonitor(); |
| 1462 // this can't be done in the base class, since it calls a virtual | 1477 // this can't be done in the base class, since it calls a virtual |
| 1463 DisableMedia_w(); | 1478 DisableMedia_w(); |
| 1464 Deinit(); | 1479 Deinit(); |
| 1465 } | 1480 } |
| 1466 | 1481 |
| 1467 bool VoiceChannel::Init_w(TransportChannel* rtp_transport, | 1482 bool VoiceChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| 1468 TransportChannel* rtcp_transport) { | 1483 DtlsTransportInternal* rtcp_dtls_transport) { |
| 1469 return BaseChannel::Init_w(rtp_transport, rtcp_transport); | 1484 return BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport); |
| 1470 } | 1485 } |
| 1471 | 1486 |
| 1472 bool VoiceChannel::SetAudioSend(uint32_t ssrc, | 1487 bool VoiceChannel::SetAudioSend(uint32_t ssrc, |
| 1473 bool enable, | 1488 bool enable, |
| 1474 const AudioOptions* options, | 1489 const AudioOptions* options, |
| 1475 AudioSource* source) { | 1490 AudioSource* source) { |
| 1476 return InvokeOnWorker(RTC_FROM_HERE, | 1491 return InvokeOnWorker(RTC_FROM_HERE, |
| 1477 Bind(&VoiceMediaChannel::SetAudioSend, media_channel(), | 1492 Bind(&VoiceMediaChannel::SetAudioSend, media_channel(), |
| 1478 ssrc, enable, options, source)); | 1493 ssrc, enable, options, source)); |
| 1479 } | 1494 } |
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| 1859 bool rtcp_mux_required, | 1874 bool rtcp_mux_required, |
| 1860 bool srtp_required) | 1875 bool srtp_required) |
| 1861 : BaseChannel(worker_thread, | 1876 : BaseChannel(worker_thread, |
| 1862 network_thread, | 1877 network_thread, |
| 1863 signaling_thread, | 1878 signaling_thread, |
| 1864 media_channel, | 1879 media_channel, |
| 1865 content_name, | 1880 content_name, |
| 1866 rtcp_mux_required, | 1881 rtcp_mux_required, |
| 1867 srtp_required) {} | 1882 srtp_required) {} |
| 1868 | 1883 |
| 1869 bool VideoChannel::Init_w(TransportChannel* rtp_transport, | 1884 bool VideoChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| 1870 TransportChannel* rtcp_transport) { | 1885 DtlsTransportInternal* rtcp_dtls_transport) { |
| 1871 return BaseChannel::Init_w(rtp_transport, rtcp_transport); | 1886 return BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport); |
| 1872 } | 1887 } |
| 1873 | 1888 |
| 1874 VideoChannel::~VideoChannel() { | 1889 VideoChannel::~VideoChannel() { |
| 1875 TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); | 1890 TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); |
| 1876 StopMediaMonitor(); | 1891 StopMediaMonitor(); |
| 1877 // this can't be done in the base class, since it calls a virtual | 1892 // this can't be done in the base class, since it calls a virtual |
| 1878 DisableMedia_w(); | 1893 DisableMedia_w(); |
| 1879 | 1894 |
| 1880 Deinit(); | 1895 Deinit(); |
| 1881 } | 1896 } |
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| 2129 | 2144 |
| 2130 RtpDataChannel::~RtpDataChannel() { | 2145 RtpDataChannel::~RtpDataChannel() { |
| 2131 TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel"); | 2146 TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel"); |
| 2132 StopMediaMonitor(); | 2147 StopMediaMonitor(); |
| 2133 // this can't be done in the base class, since it calls a virtual | 2148 // this can't be done in the base class, since it calls a virtual |
| 2134 DisableMedia_w(); | 2149 DisableMedia_w(); |
| 2135 | 2150 |
| 2136 Deinit(); | 2151 Deinit(); |
| 2137 } | 2152 } |
| 2138 | 2153 |
| 2139 bool RtpDataChannel::Init_w(TransportChannel* rtp_transport, | 2154 bool RtpDataChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| 2140 TransportChannel* rtcp_transport) { | 2155 DtlsTransportInternal* rtcp_dtls_transport) { |
| 2141 if (!BaseChannel::Init_w(rtp_transport, rtcp_transport)) { | 2156 if (!BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport)) { |
| 2142 return false; | 2157 return false; |
| 2143 } | 2158 } |
| 2144 media_channel()->SignalDataReceived.connect(this, | 2159 media_channel()->SignalDataReceived.connect(this, |
| 2145 &RtpDataChannel::OnDataReceived); | 2160 &RtpDataChannel::OnDataReceived); |
| 2146 media_channel()->SignalReadyToSend.connect( | 2161 media_channel()->SignalReadyToSend.connect( |
| 2147 this, &RtpDataChannel::OnDataChannelReadyToSend); | 2162 this, &RtpDataChannel::OnDataChannelReadyToSend); |
| 2148 return true; | 2163 return true; |
| 2149 } | 2164 } |
| 2150 | 2165 |
| 2151 bool RtpDataChannel::SendData(const SendDataParams& params, | 2166 bool RtpDataChannel::SendData(const SendDataParams& params, |
| (...skipping 225 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 2377 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, | 2392 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, |
| 2378 new DataChannelReadyToSendMessageData(writable)); | 2393 new DataChannelReadyToSendMessageData(writable)); |
| 2379 } | 2394 } |
| 2380 | 2395 |
| 2381 void RtpDataChannel::GetSrtpCryptoSuites_n( | 2396 void RtpDataChannel::GetSrtpCryptoSuites_n( |
| 2382 std::vector<int>* crypto_suites) const { | 2397 std::vector<int>* crypto_suites) const { |
| 2383 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); | 2398 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); |
| 2384 } | 2399 } |
| 2385 | 2400 |
| 2386 } // namespace cricket | 2401 } // namespace cricket |
| OLD | NEW |