NativeAPI 中的方法 PeerConnection::SetLocalDescription 崩溃
Method PeerConnection::SetLocalDescription in NativeAPI crashes
我正在尝试在用 C++ 编写的服务器和 Python 客户端之间建立数据通道。当服务器尝试设置在方法 "CreateAnswer"
中创建的本地会话描述时，服务器崩溃并出现 SIGSEGV 错误
服务端和客户端通过WebSocket交换SDP信息，应该打开没有视频和音频流的数据通道。这两个程序都在 docker-compose 下工作在不同的服务中。所以没有提供音频或视频设备。我使用来自 m76 分支的 WebRTC Native API。
崩溃处理程序：
static void OnAnswerCreated(WebRTCManagerImpl* impl_, webrtc::SessionDescriptionInterface* desc) {
LOG4CPLUS_INFO_FMT(impl_->logger_, "Answer created session_id %s", desc->session_id().c_str());
std::string offer_string;
desc->ToString(&offer_string);
LOG4CPLUS_DEBUG_FMT(impl_->logger_, "Offer string: %s", offer_string.c_str());
impl_->peer_connection_->SetLocalDescription(&impl_->set_session_description_observer_, desc);
impl_->signaling_->SendSessionDescription(*desc);
};
我与这家工厂建立联系：
webrtc::PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = nullptr;
dependencies.worker_thread = nullptr;
dependencies.signaling_thread = nullptr;
dependencies.call_factory = webrtc::CreateCallFactory();
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
dependencies.event_log_factory = absl::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
cricket::MediaEngineDependencies mediaDependencies;
mediaDependencies.task_queue_factory = dependencies.task_queue_factory.get();
mediaDependencies.adm = rtc::scoped_refptr<webrtc::FakeAudioDeviceModule>(new webrtc::FakeAudioDeviceModule);
mediaDependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
mediaDependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
mediaDependencies.audio_processing = webrtc::AudioProcessingBuilder().Create();
mediaDependencies.video_encoder_factory = webrtc::CreateBuiltinVideoEncoderFactory();
mediaDependencies.video_decoder_factory = webrtc::CreateBuiltinVideoDecoderFactory();
dependencies.media_engine = cricket::CreateMediaEngine(std::move(mediaDependencies));
return dependencies;
}
webrtc::PeerConnectionFactoryDependencies deps = CreatePeerConnectionFactoryDependencies();
deps.signaling_thread = signaling_thread_.get();
// deps.network_thread = network_thread.get();
// deps.worker_thread = worker_thread.get();
peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(std::move(deps));
调用堆栈：
<unknown> 0x0000000001e798f7
webrtc::PeerConnection::ValidateSessionDescription(webrtc::SessionDescriptionInterface const*, cricket::ContentSource) 0x00000000005e74dc
webrtc::PeerConnection::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x00000000005bb677
void webrtc::ReturnType<void>::Invoke<webrtc::PeerConnectionInterface, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>(webrtc::PeerConnectionInterface*, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x000000000059b814
webrtc::MethodCall2<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::OnMessage(rtc::Message*) 0x0000000000598f5f
webrtc::internal::SynchronousMethodCall::Invoke(rtc::Location const&, rtc::Thread*) 0x00000000007198fc
webrtc::MethodCall2<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::Marshal(rtc::Location const&, rtc::Thread*) 0x0000000000593706
webrtc::PeerConnectionProxyWithInternal<webrtc::PeerConnectionInterface>::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x000000000058c982
preprocessor::p2p::WebRTCManager::WebRTCManagerImpl::OnAnswerCreated webrtc_manager.cpp:226
std::__invoke_impl<void, void (*&)(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*), preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*&, webrtc::SessionDescriptionInterface*> invoke.h:60
std::__invoke<void (*&)(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*), preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*&, webrtc::SessionDescriptionInterface*> invoke.h:95
std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)>::__call<void, webrtc::SessionDescriptionInterface*&&, 0ul, 1ul>(std::tuple<webrtc::SessionDescriptionInterface*&&>&&, std::_Index_tuple<0ul, 1ul>) functional:467
std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)>::operator()<webrtc::SessionDescriptionInterface*, void>(webrtc::SessionDescriptionInterface*&&) functional:549
std::_Function_handler<void (webrtc::SessionDescriptionInterface*), std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)> >::_M_invoke(std::_Any_data const&, webrtc::SessionDescriptionInterface*&&) std_function.h:316
std::function<void (webrtc::SessionDescriptionInterface*)>::operator()(webrtc::SessionDescriptionInterface*) const std_function.h:706
preprocessor::p2p::CreateSessionDescriptionObserver::OnSuccess webrtc_manager.cpp:79
webrtc::WebRtcSessionDescriptionFactory::OnMessage(rtc::Message*) 0x0000000000b90785
rtc::MessageQueue::Dispatch(rtc::Message*) 0x00000000005712f8
rtc::Thread::ProcessMessages(int) 0x0000000000553398
rtc::Thread::Run() 0x0000000000552993
rtc::Thread::PreRun(void*) 0x0000000000552950
start_thread 0x00007ffff76536db
clone 0x00007ffff608a88f
WebRTC 日志：
(audio_processing_impl.cc:435): Capture analyzer activated: 0
Capture post processor activated: 0
Render pre processor activated: 0
(webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtc_video_engine.cc:479): WebRtcVideoEngine::WebRtcVideoEngine()
(webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init
(webrtc_voice_engine.cc:227): Supported send codecs in order of preference:
(webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtc_voice_engine.cc:230): ISAC/16000/1 (103)
(webrtc_voice_engine.cc:230): ISAC/32000/1 (104)
(webrtc_voice_engine.cc:230): G722/8000/1 (9)
(webrtc_voice_engine.cc:230): ILBC/8000/1 (102)
(webrtc_voice_engine.cc:230): PCMU/8000/1 (0)
(webrtc_voice_engine.cc:230): PCMA/8000/1 (8)
(webrtc_voice_engine.cc:230): CN/32000/1 (106)
(webrtc_voice_engine.cc:230): CN/16000/1 (105)
(webrtc_voice_engine.cc:230): CN/8000/1 (13)
(webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)
(webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)
(webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)
(webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)
(webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:
(webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtc_voice_engine.cc:236): ISAC/16000/1 (103)
(webrtc_voice_engine.cc:236): ISAC/32000/1 (104)
(webrtc_voice_engine.cc:236): G722/8000/1 (9)
(webrtc_voice_engine.cc:236): ILBC/8000/1 (102)
(webrtc_voice_engine.cc:236): PCMU/8000/1 (0)
(webrtc_voice_engine.cc:236): PCMA/8000/1 (8)
(webrtc_voice_engine.cc:236): CN/32000/1 (106)
(webrtc_voice_engine.cc:236): CN/16000/1 (105)
(webrtc_voice_engine.cc:236): CN/8000/1 (13)
(webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)
(webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)
(webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)
(webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)
(apm_helpers.cc:32): Setting AGC mode to 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, extended_filter_aec: 0, delay_agnostic_aec: 0, experimental_ns: 0, residual_echo_detector: 1, }
(render_delay_buffer.cc:341): Applying total delay of 5 blocks.
(matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms.
(matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms.
(matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms.
(matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms.
(matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms.
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(apm_helpers.cc:48): Echo control set to 1 with mode 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(apm_helpers.cc:62): NS set to 1
(webrtc_voice_engine.cc:447): Stereo swapping enabled? 0
(webrtc_voice_engine.cc:452): NetEq capacity is 200
(webrtc_voice_engine.cc:458): NetEq fast mode? 0
(webrtc_voice_engine.cc:464): NetEq minimum delay is 0
(webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0
(webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0
(webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0
(webrtc_voice_engine.cc:501): Experimental ns is enabled? 0
(webrtc_voice_engine.cc:511): Setting AGC to 1
(webrtc_voice_engine.cc:533): Typing detection is enabled? 1
(audio_processing_impl.cc:699): Highpass filter activated: 1
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(webrtc_sdp.cc:3255): Ignored line: a=sctpmap:5000 webrtc-datachannel 65535
(rtc_event_log_impl.cc:63): Creating legacy encoder for RTC event log.
(peer_connection_factory.cc:361): Using default network controller factory
(bitrate_prober.cc:69): Bandwidth probing enabled, set to inactive
(paced_sender.cc:421): ProcessThreadAttached 0xec072e20
(cpu_info.cc:53): Available number of cores: 8
(aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85
(remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.
(remote_estimator_proxy.cc:44): Maximum interval between transport feedback RTCP messages (ms): 250
(openssl_identity.cc:44): Making key pair
(peer_connection.cc:5531): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(openssl_identity.cc:92): Returning key pair
(openssl_certificate.cc:58): Making certificate for WebRTC
(openssl_certificate.cc:108): Returning certificate
(p2p_transport_channel.cc:519): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:528): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:535): Set ping most likely connection to 0
(p2p_transport_channel.cc:542): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:555): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:564): Set regather_on_failed_networks_interval to 300000
(p2p_transport_channel.cc:583): Set receiving_switching_delay to 1000
(jsep_transport_controller.cc:1214): Creating DtlsSrtpTransport.
(dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0
(dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport dc004830
(p2p_transport_channel.cc:465): Received remote ICE parameters: ufrag=YAvY, renomination disabled
(peer_connection.cc:4185): Session: 7301418690559709073 Old state: kStable New state: kHaveRemoteOffer
(peer_connection.cc:5531): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(peer_connection.cc:5559): Local and Remote descriptions must be applied to get the SSL Role of the session.
(paced_sender.cc:293): Elapsed time (12680 ms) longer than expected, limiting to 2000 ms
Signal: SIGSEGV (Segmentation fault)
我猜问题不在回调，而在连接初始化。但是我做错了什么？
我在我的代码中发现了错误：
peer_connection_->SetRemoteDescription(&set_session_description_observer_, desc.get());
我传递了原始指针然后释放了带内存的智能指针。
我正在尝试在用 C++ 编写的服务器和 Python 客户端之间建立数据通道。当服务器尝试设置在方法 "CreateAnswer"
中创建的本地会话描述时，服务器崩溃并出现 SIGSEGV 错误服务端和客户端通过WebSocket交换SDP信息，应该打开没有视频和音频流的数据通道。这两个程序都在 docker-compose 下工作在不同的服务中。所以没有提供音频或视频设备。我使用来自 m76 分支的 WebRTC Native API。
崩溃处理程序：
static void OnAnswerCreated(WebRTCManagerImpl* impl_, webrtc::SessionDescriptionInterface* desc) {
LOG4CPLUS_INFO_FMT(impl_->logger_, "Answer created session_id %s", desc->session_id().c_str());
std::string offer_string;
desc->ToString(&offer_string);
LOG4CPLUS_DEBUG_FMT(impl_->logger_, "Offer string: %s", offer_string.c_str());
impl_->peer_connection_->SetLocalDescription(&impl_->set_session_description_observer_, desc);
impl_->signaling_->SendSessionDescription(*desc);
};
我与这家工厂建立联系：
webrtc::PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = nullptr;
dependencies.worker_thread = nullptr;
dependencies.signaling_thread = nullptr;
dependencies.call_factory = webrtc::CreateCallFactory();
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
dependencies.event_log_factory = absl::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
cricket::MediaEngineDependencies mediaDependencies;
mediaDependencies.task_queue_factory = dependencies.task_queue_factory.get();
mediaDependencies.adm = rtc::scoped_refptr<webrtc::FakeAudioDeviceModule>(new webrtc::FakeAudioDeviceModule);
mediaDependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
mediaDependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
mediaDependencies.audio_processing = webrtc::AudioProcessingBuilder().Create();
mediaDependencies.video_encoder_factory = webrtc::CreateBuiltinVideoEncoderFactory();
mediaDependencies.video_decoder_factory = webrtc::CreateBuiltinVideoDecoderFactory();
dependencies.media_engine = cricket::CreateMediaEngine(std::move(mediaDependencies));
return dependencies;
}
webrtc::PeerConnectionFactoryDependencies deps = CreatePeerConnectionFactoryDependencies();
deps.signaling_thread = signaling_thread_.get();
// deps.network_thread = network_thread.get();
// deps.worker_thread = worker_thread.get();
peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(std::move(deps));
调用堆栈：
<unknown> 0x0000000001e798f7
webrtc::PeerConnection::ValidateSessionDescription(webrtc::SessionDescriptionInterface const*, cricket::ContentSource) 0x00000000005e74dc
webrtc::PeerConnection::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x00000000005bb677
void webrtc::ReturnType<void>::Invoke<webrtc::PeerConnectionInterface, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>(webrtc::PeerConnectionInterface*, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x000000000059b814
webrtc::MethodCall2<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::OnMessage(rtc::Message*) 0x0000000000598f5f
webrtc::internal::SynchronousMethodCall::Invoke(rtc::Location const&, rtc::Thread*) 0x00000000007198fc
webrtc::MethodCall2<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::Marshal(rtc::Location const&, rtc::Thread*) 0x0000000000593706
webrtc::PeerConnectionProxyWithInternal<webrtc::PeerConnectionInterface>::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) 0x000000000058c982
preprocessor::p2p::WebRTCManager::WebRTCManagerImpl::OnAnswerCreated webrtc_manager.cpp:226
std::__invoke_impl<void, void (*&)(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*), preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*&, webrtc::SessionDescriptionInterface*> invoke.h:60
std::__invoke<void (*&)(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*), preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*&, webrtc::SessionDescriptionInterface*> invoke.h:95
std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)>::__call<void, webrtc::SessionDescriptionInterface*&&, 0ul, 1ul>(std::tuple<webrtc::SessionDescriptionInterface*&&>&&, std::_Index_tuple<0ul, 1ul>) functional:467
std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)>::operator()<webrtc::SessionDescriptionInterface*, void>(webrtc::SessionDescriptionInterface*&&) functional:549
std::_Function_handler<void (webrtc::SessionDescriptionInterface*), std::_Bind<void (*(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, std::_Placeholder<1>))(preprocessor::p2p::WebRTCManager::WebRTCManagerImpl*, webrtc::SessionDescriptionInterface*)> >::_M_invoke(std::_Any_data const&, webrtc::SessionDescriptionInterface*&&) std_function.h:316
std::function<void (webrtc::SessionDescriptionInterface*)>::operator()(webrtc::SessionDescriptionInterface*) const std_function.h:706
preprocessor::p2p::CreateSessionDescriptionObserver::OnSuccess webrtc_manager.cpp:79
webrtc::WebRtcSessionDescriptionFactory::OnMessage(rtc::Message*) 0x0000000000b90785
rtc::MessageQueue::Dispatch(rtc::Message*) 0x00000000005712f8
rtc::Thread::ProcessMessages(int) 0x0000000000553398
rtc::Thread::Run() 0x0000000000552993
rtc::Thread::PreRun(void*) 0x0000000000552950
start_thread 0x00007ffff76536db
clone 0x00007ffff608a88f
WebRTC 日志：
(audio_processing_impl.cc:435): Capture analyzer activated: 0
Capture post processor activated: 0
Render pre processor activated: 0
(webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtc_video_engine.cc:479): WebRtcVideoEngine::WebRtcVideoEngine()
(webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init
(webrtc_voice_engine.cc:227): Supported send codecs in order of preference:
(webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtc_voice_engine.cc:230): ISAC/16000/1 (103)
(webrtc_voice_engine.cc:230): ISAC/32000/1 (104)
(webrtc_voice_engine.cc:230): G722/8000/1 (9)
(webrtc_voice_engine.cc:230): ILBC/8000/1 (102)
(webrtc_voice_engine.cc:230): PCMU/8000/1 (0)
(webrtc_voice_engine.cc:230): PCMA/8000/1 (8)
(webrtc_voice_engine.cc:230): CN/32000/1 (106)
(webrtc_voice_engine.cc:230): CN/16000/1 (105)
(webrtc_voice_engine.cc:230): CN/8000/1 (13)
(webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)
(webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)
(webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)
(webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)
(webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:
(webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtc_voice_engine.cc:236): ISAC/16000/1 (103)
(webrtc_voice_engine.cc:236): ISAC/32000/1 (104)
(webrtc_voice_engine.cc:236): G722/8000/1 (9)
(webrtc_voice_engine.cc:236): ILBC/8000/1 (102)
(webrtc_voice_engine.cc:236): PCMU/8000/1 (0)
(webrtc_voice_engine.cc:236): PCMA/8000/1 (8)
(webrtc_voice_engine.cc:236): CN/32000/1 (106)
(webrtc_voice_engine.cc:236): CN/16000/1 (105)
(webrtc_voice_engine.cc:236): CN/8000/1 (13)
(webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)
(webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)
(webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)
(webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)
(apm_helpers.cc:32): Setting AGC mode to 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, extended_filter_aec: 0, delay_agnostic_aec: 0, experimental_ns: 0, residual_echo_detector: 1, }
(render_delay_buffer.cc:341): Applying total delay of 5 blocks.
(matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms.
(matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms.
(matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms.
(matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms.
(matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms.
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(apm_helpers.cc:48): Echo control set to 1 with mode 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(audio_processing_impl.cc:699): Highpass filter activated: 0
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(apm_helpers.cc:62): NS set to 1
(webrtc_voice_engine.cc:447): Stereo swapping enabled? 0
(webrtc_voice_engine.cc:452): NetEq capacity is 200
(webrtc_voice_engine.cc:458): NetEq fast mode? 0
(webrtc_voice_engine.cc:464): NetEq minimum delay is 0
(webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0
(webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0
(webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0
(webrtc_voice_engine.cc:501): Experimental ns is enabled? 0
(webrtc_voice_engine.cc:511): Setting AGC to 1
(webrtc_voice_engine.cc:533): Typing detection is enabled? 1
(audio_processing_impl.cc:699): Highpass filter activated: 1
(audio_processing_impl.cc:717): Gain Controller 2 activated: 0
(audio_processing_impl.cc:719): Pre-amplifier activated: 0
(webrtc_sdp.cc:3255): Ignored line: a=sctpmap:5000 webrtc-datachannel 65535
(rtc_event_log_impl.cc:63): Creating legacy encoder for RTC event log.
(peer_connection_factory.cc:361): Using default network controller factory
(bitrate_prober.cc:69): Bandwidth probing enabled, set to inactive
(paced_sender.cc:421): ProcessThreadAttached 0xec072e20
(cpu_info.cc:53): Available number of cores: 8
(aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85
(remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.
(remote_estimator_proxy.cc:44): Maximum interval between transport feedback RTCP messages (ms): 250
(openssl_identity.cc:44): Making key pair
(peer_connection.cc:5531): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(openssl_identity.cc:92): Returning key pair
(openssl_certificate.cc:58): Making certificate for WebRTC
(openssl_certificate.cc:108): Returning certificate
(p2p_transport_channel.cc:519): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:528): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:535): Set ping most likely connection to 0
(p2p_transport_channel.cc:542): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:555): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:564): Set regather_on_failed_networks_interval to 300000
(p2p_transport_channel.cc:583): Set receiving_switching_delay to 1000
(jsep_transport_controller.cc:1214): Creating DtlsSrtpTransport.
(dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0
(dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport dc004830
(p2p_transport_channel.cc:465): Received remote ICE parameters: ufrag=YAvY, renomination disabled
(peer_connection.cc:4185): Session: 7301418690559709073 Old state: kStable New state: kHaveRemoteOffer
(peer_connection.cc:5531): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(peer_connection.cc:5559): Local and Remote descriptions must be applied to get the SSL Role of the session.
(paced_sender.cc:293): Elapsed time (12680 ms) longer than expected, limiting to 2000 ms
Signal: SIGSEGV (Segmentation fault)
我猜问题不在回调，而在连接初始化。但是我做错了什么？
我在我的代码中发现了错误：
peer_connection_->SetRemoteDescription(&set_session_description_observer_, desc.get());
我传递了原始指针然后释放了带内存的智能指针。