Method PeerConnection::SetLocalDescription in NativeAPI crashes - c++

Question

I'm trying to set up a data channel between a server written in C++ and a Python client. The server crashes with SIGSEGV error when it tries to set a local session description created in method "CreateAnswer"
The server and client exchange SDP information via WebSocket and should open the data channel without video and audio streams. Both programs are working under docker-compose in different services. So no audio or video devices are provided. I use WebRTC Native API from m76 branch.
Crashing handler:
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);
};
I create my connection with this factory:
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));
The call stack:
<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 logs:
(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)
I guess the problem is not in callback but in the connection initialization. But what am I doing wrong?

Answer 1

I've found the error in my code:
peer_connection_->SetRemoteDescription(&set_session_description_observer_, desc.get());
I passed the raw pointer then release the smart one with the memory.