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sourcey / libsourcey

C++14 evented IO libraries for high performance networking and media based applications
https://sourcey.com/libsourcey
GNU Lesser General Public License v2.1
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Output from webrtcrecorder #229

Open videodoctor opened 6 years ago

videodoctor commented 6 years ago

When I run webrtcrecorder on Ubuntu 16 (via remote ssh / terminal), I'm getting the following output:

ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM

If I don't have the node app running for the sample client, I'll also get an error about not finding the server, but that's to be expected I presume. Am I missing a dependency? I went through the circle.yml file to make sure I had all of the dependencies as listed there...

videodoctor commented 6 years ago

Hmm... I've compiled a debug release, and the webrtcrecorder app seems to be looking for local mic/cameras on the system when invoked? The description of the sample leads me to believe it's recording incoming webrtc streams, not local devices. Is there a sample that shows a stream recorder?

videodoctor commented 6 years ago

Ok, I've recompiled a debug version, and I can see what's going on a bit more clearly. I'm creating an SSH tunnel from my local computer to the remote box running the libsourcey webrtcrecorder/node client apps, and the tunnel is active over ports 4499 and 4500. The webrtcrecorderd app is logging a lot of data for the connected session, but it's reporting this after the handshake:

19:47:17 [debug] [signaler.cpp(89)] Peer message: demo|Wt9AUunVby2QjEyoAAAD 19:47:17 [debug] [peermanager.cpp(85)] Received candidate: candidate:1 2 TCP 2105524478 192.168.3.43 9 typ host tcptype active (webrtcsession.cc:1160): ProcessIceMessage: Not ready to use candidate.

videodoctor commented 6 years ago

Here's my full debug log:

ubuntu@ip-10-10-0-120:~/tmp/libsourcey/build/webrtc/samples/webrtcrecorder$ sudo ./webrtcrecorderd 22:25:03 [debug] [application.cpp(45)] Create (webrtcvoiceengine.cc:238): WebRtcVoiceEngine::WebRtcVoiceEngine (webrtcvoiceengine.cc:245): Supported send codecs in order of preference: (webrtcvoiceengine.cc:248): opus/48000/2 { minptime=10 useinbandfec=1 } (111) (webrtcvoiceengine.cc:248): ISAC/16000/1 (103) (webrtcvoiceengine.cc:248): ISAC/32000/1 (104) (webrtcvoiceengine.cc:248): G722/8000/1 (9) (webrtcvoiceengine.cc:248): ILBC/8000/1 (102) (webrtcvoiceengine.cc:248): PCMU/8000/1 (0) (webrtcvoiceengine.cc:248): PCMA/8000/1 (8) (webrtcvoiceengine.cc:248): CN/32000/1 (106) (webrtcvoiceengine.cc:248): CN/16000/1 (105) (webrtcvoiceengine.cc:248): CN/8000/1 (13) (webrtcvoiceengine.cc:248): telephone-event/48000/1 (110) (webrtcvoiceengine.cc:248): telephone-event/32000/1 (112) (webrtcvoiceengine.cc:248): telephone-event/16000/1 (113) (webrtcvoiceengine.cc:248): telephone-event/8000/1 (126) (webrtcvoiceengine.cc:251): Supported recv codecs in order of preference: (webrtcvoiceengine.cc:254): opus/48000/2 { minptime=10 useinbandfec=1 } (111) (webrtcvoiceengine.cc:254): ISAC/16000/1 (103) (webrtcvoiceengine.cc:254): ISAC/32000/1 (104) (webrtcvoiceengine.cc:254): G722/8000/1 (9) (webrtcvoiceengine.cc:254): ILBC/8000/1 (102) (webrtcvoiceengine.cc:254): PCMU/8000/1 (0) (webrtcvoiceengine.cc:254): PCMA/8000/1 (8) (webrtcvoiceengine.cc:254): CN/32000/1 (106) (webrtcvoiceengine.cc:254): CN/16000/1 (105) (webrtcvoiceengine.cc:254): CN/8000/1 (13) (webrtcvoiceengine.cc:254): telephone-event/48000/1 (110) (webrtcvoiceengine.cc:254): telephone-event/32000/1 (112) (webrtcvoiceengine.cc:254): telephone-event/16000/1 (113) (webrtcvoiceengine.cc:254): telephone-event/8000/1 (126) (webrtcvoiceengine.cc:262): VoiceEngine 4.1.0 (audio_device_impl.cc:84): Create (audio_device_buffer.cc:62): AudioDeviceBuffer::ctor (audio_device_impl.cc:128): AudioDeviceModuleImpl (audio_device_impl.cc:136): CheckPlatform (audio_device_impl.cc:150): current platform is Linux (audio_device_impl.cc:177): CreatePlatformSpecificObjects (audio_device_impl.cc:1844): PlatformAudioLayer (audio_device_impl.cc:258): attempting to use the Linux PulseAudio APIs... Home directory not accessible: Permission denied (webrtcvoiceengine.cc:672): webrtc: failed to connect context, error=-1 (audio_device_pulse_linux.cc:172): failed to initialize PulseAudio (webrtcvoiceengine.cc:672): webrtc: Close (webrtcvoiceengine.cc:672): webrtc: CloseSpeaker (webrtcvoiceengine.cc:672): webrtc: CloseMicrophone (audio_device_impl.cc:275): Linux PulseAudio is not supported => ALSA APIs will be utilized instead (audio_device_impl.cc:344): AttachAudioBuffer (audio_device_buffer.cc:178): SetRecordingSampleRate(0) (audio_device_buffer.cc:185): SetPlayoutSampleRate(0) (audio_device_buffer.cc:202): SetRecordingChannels(0) (audio_device_buffer.cc:209): SetPlayoutChannels(0) (audio_device_impl.cc:1452): RegisterEventObserver (audio_device_impl.cc:1465): RegisterAudioCallback (audio_device_buffer.cc:77): RegisterAudioCallback (audio_device_impl.cc:467): Init (audio_device_alsa_linux.cc:174): failed to open X display, typing detection will not work (audio_device_impl.cc:1196): SetPlayoutDevice(0) (webrtcvoiceengine.cc:672): webrtc: number of availiable audio output devices is 0 (audio_device_impl.cc:516): InitSpeaker (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=) (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: Init() failed to initialize the speaker (error=9005) (audio_device_impl.cc:1291): SetRecordingDevice(0) (webrtcvoiceengine.cc:672): webrtc: number of availiable audio input devices is 0 (audio_device_impl.cc:526): InitMicrophone (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=) (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: Init() failed to initialize the microphone (error=9004) (audio_device_impl.cc:1005): StereoPlayoutIsAvailable (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=) (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: InitSpeaker() failed (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: InitPlayout open () ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM (webrtcvoiceengine.cc:672): webrtc: unable to open playback device: No such file or directory (-2) (audio_device_impl.cc:1015): output: 0 (audio_device_impl.cc:1024): SetStereoPlayout(0) (audio_device_buffer.cc:209): SetPlayoutChannels(1) (audio_device_impl.cc:891): StereoRecordingIsAvailable (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=) (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: InitMicrophone() failed (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers (webrtcvoiceengine.cc:672): webrtc: InitRecording open () ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM (webrtcvoiceengine.cc:672): webrtc: unable to open record device: No such file or directory (audio_device_impl.cc:901): output: 0 (audio_device_impl.cc:910): SetStereoRecording(0) (audio_device_buffer.cc:202): SetRecordingChannels(1) (webrtcvoiceengine.cc:672): webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x7c0c7470) (webrtcvoiceengine.cc:672): webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x7c0c7470) (audio_processing_impl.cc:684): Level controller activated: 0 (audio_processing_impl.cc:691): Highpass filter activated: 1 (webrtcvoiceengine.cc:338): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, typing: true, agc_delta: 0, experimental_agc: false, extended_filter_aec: false, delay_agnostic_aec: false, experimental_ns: false, intelligibility_enhancer: false, level_control: false, residual_echo_detector: true, } (audio_device_impl.cc:1760): BuiltInAECIsAvailable (audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform (audio_device_impl.cc:1763): output: 0 (apm_helpers.cc:106): Echo control set to 1 with mode 0 (apm_helpers.cc:116): EC metrics set to 1 (audio_device_impl.cc:1776): BuiltInAGCIsAvailable (audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform (audio_device_impl.cc:1779): output: 0 (audio_device_impl.cc:1071): SetAGC(1) (apm_helpers.cc:70): AGC set to 1 with mode 0 (webrtcvoiceengine.cc:482): Adjusting AGC level from default -2dB to -2dB (audio_device_impl.cc:1792): BuiltInNSIsAvailable (audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform (audio_device_impl.cc:1795): output: 0 (apm_helpers.cc:141): NS set to 1 (webrtcvoiceengine.cc:513): Stereo swapping enabled? 0 (webrtcvoiceengine.cc:518): NetEq capacity is 50 (webrtcvoiceengine.cc:524): NetEq fast mode? 0 (webrtcvoiceengine.cc:531): Typing detection is enabled? 1 (apm_helpers.cc:166): VAD set to 1 for typing detection. (webrtcvoiceengine.cc:542): Delay agnostic aec is enabled? 0 (webrtcvoiceengine.cc:551): Extended filter aec is enabled? 0 (webrtcvoiceengine.cc:560): Experimental ns is enabled? 0 (webrtcvoiceengine.cc:566): Intelligibility Enhancer is enabled? 0 (webrtcvoiceengine.cc:576): Level control: 0 (audio_processing_impl.cc:684): Level controller activated: 0 (audio_processing_impl.cc:691): Highpass filter activated: 1 (audio_device_impl.cc:1441): Recording (audio_device_impl.cc:957): SetRecordingChannel(both) (audio_device_buffer.cc:216): SetRecordingChannel(2) (audio_device_buffer.cc:217): Not implemented (adm_helpers.cc:47): Unable to set recording channel to kChannelBoth. (audio_device_impl.cc:1291): SetRecordingDevice(0) (audio_device_impl.cc:526): InitMicrophone (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory (adm_helpers.cc:56): Unable to access microphone. (audio_device_impl.cc:891): StereoRecordingIsAvailable (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: InitMicrophone() failed (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM (webrtcvoiceengine.cc:672): webrtc: unable to open record device: No such file or directory (audio_device_impl.cc:901): output: 0 (audio_device_impl.cc:910): SetStereoRecording(0) (audio_device_buffer.cc:202): SetRecordingChannels(1) (adm_helpers.cc:80): Set recording device. (audio_device_impl.cc:1399): Playing (audio_device_impl.cc:1196): SetPlayoutDevice(0) (audio_device_impl.cc:516): InitSpeaker (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory (adm_helpers.cc:100): Unable to access speaker. (audio_device_impl.cc:1005): StereoPlayoutIsAvailable (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL (webrtcvoiceengine.cc:672): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory (webrtcvoiceengine.cc:672): webrtc: InitSpeaker() failed (webrtcvoiceengine.cc:672): webrtc: GetDevicesInfo - Could not find device name or numbers ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM (webrtcvoiceengine.cc:672): webrtc: unable to open playback device: No such file or directory (-2) (audio_device_impl.cc:1015): output: 0 (audio_device_impl.cc:1024): SetStereoPlayout(0) (audio_device_buffer.cc:209): SetPlayoutChannels(1) (adm_helpers.cc:124): Set playout device. (audio_processing_impl.cc:684): Level controller activated: 0 (audio_processing_impl.cc:691): Highpass filter activated: 0 (audio_device_impl.cc:1465): RegisterAudioCallback (audio_device_buffer.cc:77): RegisterAudioCallback (audio_device_impl.cc:1465): RegisterAudioCallback (audio_device_buffer.cc:77): RegisterAudioCallback (webrtcvideoengine2.cc:476): WebRtcVideoEngine2::WebRtcVideoEngine2() (webrtcvideoengine2.cc:484): WebRtcVideoEngine2::Init 22:25:03 [debug] [signaler.cpp(116)] Client state changed from Closed to Connecting 22:25:03 [debug] [application.cpp(104)] Wait for shutdown 22:25:03 [debug] [signaler.cpp(116)] Client state changed from Connecting to Connected 22:25:03 [debug] [client.cpp(364)] On handshake: sid=XhaewpspUfLy6_7cAAAB, pingInterval=25000, pingTimeout=60000 22:25:03 [debug] [client.cpp(387)] Peer connected:videorecorder|XhaewpspUfLy6_7cAAAB 22:25:03 [debug] [signaler.cpp(116)] Client state changed from Connected to Online 22:25:05 [debug] [signaler.cpp(89)] Peer message: demo|AE-ERY5yYkBY7IGgAAAA 22:25:05 [debug] [signaler.cpp(71)] Peer connected: AE-ERY5yYkBY7IGgAAAA (bitrate_prober.cc:63): Bandwidth probing enabled, set to inactive (delay_based_bwe.cc:173): Using Trendline filter for delay change estimation. (cpu_info.cc:50): Available number of cores: 2 (remote_bitrate_estimator_single_stream.cc:58): RemoteBitrateEstimatorSingleStream: Instantiating. (send_side_congestion_controller.cc:185): SignalNetworkState Down (paced_sender.cc:279): PacedSender paused. (delay_based_bwe.cc:361): BWE Setting start bitrate to: 300000 (paced_sender.cc:485): ProcessThreadAttached 0x0x7ffa7c1bd640 (webrtcvideoengine2.cc:621): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]} (opensslidentity.cc:41): Making key pair 22:25:05 [debug] [client.cpp(387)] Peer connected:demo|AE-ERY5yYkBY7IGgAAAA (opensslidentity.cc:89): Returning key pair (opensslidentity.cc:96): Making certificate for WebRTC (opensslidentity.cc:143): Returning certificate 22:25:06 [debug] [signaler.cpp(89)] Peer message: demo|AE-ERY5yYkBY7IGgAAAA 22:25:06 [debug] [signaler.cpp(89)] Peer message: demo|AE-ERY5yYkBY7IGgAAAA 22:26:09 [debug] [signaler.cpp(89)] Peer message: demo|AE-ERY5yYkBY7IGgAAAA 22:26:09 [debug] [client.cpp(394)] Peer disconnected:demo|AE-ERY5yYkBY7IGgAAAA 22:26:09 [debug] [signaler.cpp(104)] Peer disconnected 22:26:09 [debug] [signaler.cpp(108)] Closing peer connection: AE-ERY5yYkBY7IGgAAAA 22:26:09 [debug] [peer.cpp(106)] AE-ERY5yYkBY7IGgAAAA: Closing (webrtcsession.cc:848): Session:7793429856893616933 Old state:STATE_INIT New state:STATE_CLOSED 22:26:09 [debug] [peer.cpp(185)] AE-ERY5yYkBY7IGgAAAA: On ICE connection change: 6 22:26:09 [debug] [peer.cpp(191)] AE-ERY5yYkBY7IGgAAAA: On ICE gathering change: 2 22:26:09 [debug] [peer.cpp(165)] AE-ERY5yYkBY7IGgAAAA: On signaling state change: 5 22:26:09 [debug] [peermanager.cpp(110)] Deleting peer connection: AE-ERY5yYkBY7IGgAAAA (paced_sender.cc:485): ProcessThreadAttached 0x0 (paced_sender.cc:485): ProcessThreadAttached 0x0 22:26:10 [debug] [peer.cpp(51)] AE-ERY5yYkBY7IGgAAAA: Destroying (webrtcsession.cc:533): Session: 7793429856893616933 is destroyed. 22:26:14 [debug] [signaler.cpp(89)] Peer message: demo|8iiSz-xKqmbDEPrQAAAC 22:26:14 [debug] [signaler.cpp(71)] Peer connected: 8iiSz-xKqmbDEPrQAAAC (bitrate_prober.cc:63): Bandwidth probing enabled, set to inactive (delay_based_bwe.cc:173): Using Trendline filter for delay change estimation. (remote_bitrate_estimator_single_stream.cc:58): RemoteBitrateEstimatorSingleStream: Instantiating. (send_side_congestion_controller.cc:185): SignalNetworkState Down (paced_sender.cc:279): PacedSender paused. (delay_based_bwe.cc:361): BWE Setting start bitrate to: 300000 (paced_sender.cc:485): ProcessThreadAttached 0x0x7ffa7c1bcc80 (webrtcvideoengine2.cc:621): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]} (opensslidentity.cc:41): Making key pair 22:26:14 [debug] [client.cpp(387)] Peer connected:demo|8iiSz-xKqmbDEPrQAAAC (opensslidentity.cc:89): Returning key pair (opensslidentity.cc:96): Making certificate for WebRTC (opensslidentity.cc:143): Returning certificate 22:26:16 [debug] [signaler.cpp(89)] Peer message: demo|8iiSz-xKqmbDEPrQAAAC 22:26:16 [debug] [peer.cpp(130)] 8iiSz-xKqmbDEPrQAAAC: Receive offer: v=0 o=- 661132907157492103 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:wQ5X a=ice-pwd:xjtP77gg3+o2wTalgZZi1Oyf a=ice-options:trickle a=fingerprint:sha-256 53:68:A4:6D:51:22:D7:D6:AD:83:CD:06:D8:71:58:3A:08:45:A4:7D:A7:35:70:5B:16:1E:8D:DF:28:18:B8:6D a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:1120370049 cname:wsr+XpJJJiDlk07G a=ssrc:1120370049 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 36118ade-d8e1-4eef-b462-d916f46a1ae6 a=ssrc:1120370049 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:1120370049 label:36118ade-d8e1-4eef-b462-d916f46a1ae6 m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:wQ5X a=ice-pwd:xjtP77gg3+o2wTalgZZi1Oyf a=ice-options:trickle a=fingerprint:sha-256 53:68:A4:6D:51:22:D7:D6:AD:83:CD:06:D8:71:58:3A:08:45:A4:7D:A7:35:70:5B:16:1E:8D:DF:28:18:B8:6D a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:4 urn:3gpp:video-orientation a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 goog-remb a=rtcp-fb:96 transport-cc a=rtcp-fb:96 ccm fir a=rtcp-fb:96 nack a=rtcp-fb:96 nack pli a=rtpmap:97 rtx/90000 a=fmtp:97 apt=96 a=rtpmap:98 VP9/90000 a=rtcp-fb:98 goog-remb a=rtcp-fb:98 transport-cc a=rtcp-fb:98 ccm fir a=rtcp-fb:98 nack a=rtcp-fb:98 nack pli a=rtpmap:99 rtx/90000 a=fmtp:99 apt=98 a=rtpmap:100 H264/90000 a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:101 rtx/90000 a=fmtp:101 apt=100 a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc a=rtcp-fb:102 ccm fir a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f a=rtpmap:123 rtx/90000 a=fmtp:123 apt=102 a=rtpmap:127 H264/90000 a=rtcp-fb:127 goog-remb a=rtcp-fb:127 transport-cc a=rtcp-fb:127 ccm fir a=rtcp-fb:127 nack a=rtcp-fb:127 nack pli a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:122 rtx/90000 a=fmtp:122 apt=127 a=rtpmap:125 H264/90000 a=rtcp-fb:125 goog-remb a=rtcp-fb:125 transport-cc a=rtcp-fb:125 ccm fir a=rtcp-fb:125 nack a=rtcp-fb:125 nack pli a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:107 rtx/90000 a=fmtp:107 apt=125 a=rtpmap:108 red/90000 a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 a=rtpmap:124 ulpfec/90000 a=ssrc-group:FID 4265947015 2616070878 a=ssrc:4265947015 cname:wsr+XpJJJiDlk07G a=ssrc:4265947015 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:4265947015 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:4265947015 label:50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:2616070878 cname:wsr+XpJJJiDlk07G a=ssrc:2616070878 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:2616070878 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:2616070878 label:50db3d62-fb44-4ff5-bc56-fc0819bc33f6

(p2ptransportchannel.cc:400): Set ping most likely connection to 0 (p2ptransportchannel.cc:420): Set presume writable when fully relayed to 0 (webrtcvoiceengine.cc:1508): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, } (webrtcvoiceengine.cc:338): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, } (webrtcvoiceengine.cc:518): NetEq capacity is 50 (webrtcvoiceengine.cc:524): NetEq fast mode? 0 (webrtcvoiceengine.cc:542): Delay agnostic aec is enabled? 0 (webrtcvoiceengine.cc:551): Extended filter aec is enabled? 0 (webrtcvoiceengine.cc:560): Experimental ns is enabled? 0 (webrtcvoiceengine.cc:566): Intelligibility Enhancer is enabled? 0 (webrtcvoiceengine.cc:576): Level control: 0 (audio_processing_impl.cc:684): Level controller activated: 0 (audio_processing_impl.cc:691): Highpass filter activated: 1 (webrtcvoiceengine.cc:1527): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, } (channel.cc:182): Created channel for audio (channel.cc:334): Setting RTP Transport for audio on audio transport 0x7ffa700031b0 (p2ptransportchannel.cc:400): Set ping most likely connection to 0 (p2ptransportchannel.cc:420): Set presume writable when fully relayed to 0 (webrtcvideoengine2.cc:493): CreateChannel. Options: VideoOptions {} (webrtcvideoengine2.cc:621): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]} (channel.cc:182): Created channel for video (channel.cc:334): Setting RTP Transport for video on video transport 0x7ffa70004720 (transportcontroller.cc:638): Set remote transport description on audio (p2ptransportchannel.cc:334): Remote supports ICE renomination ? 0 (transportcontroller.cc:638): Set remote transport description on video (p2ptransportchannel.cc:334): Remote supports ICE renomination ? 0 (webrtcsession.cc:848): Session:8669743502869645713 Old state:STATE_INIT New state:STATE_RECEIVEDOFFER 22:26:16 [debug] [peer.cpp(165)] 8iiSz-xKqmbDEPrQAAAC: On signaling state change: 3 (channel.cc:1792): Setting remote voice description (webrtcvoiceengine.cc:1336): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], max_bandwidth_bps: -1, options: AudioOptions {}} (webrtcvoiceengine.cc:1725): Recreate all the receive streams because the send codec has changed. (webrtcvoiceengine.cc:2177): WebRtcVoiceMediaChannel::SetMaxSendBitrate. (webrtcvoiceengine.cc:1508): Setting voice channel options: AudioOptions {} (webrtcvoiceengine.cc:338): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, } (webrtcvoiceengine.cc:518): NetEq capacity is 50 (webrtcvoiceengine.cc:524): NetEq fast mode? 0 (webrtcvoiceengine.cc:542): Delay agnostic aec is enabled? 0 (webrtcvoiceengine.cc:551): Extended filter aec is enabled? 0 (webrtcvoiceengine.cc:560): Experimental ns is enabled? 0 (webrtcvoiceengine.cc:566): Intelligibility Enhancer is enabled? 0 (webrtcvoiceengine.cc:576): Level control: 0 (audio_processing_impl.cc:684): Level controller activated: 0 (audio_processing_impl.cc:691): Highpass filter activated: 1 (webrtcvoiceengine.cc:1527): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, } (webrtcvoiceengine.cc:1906): AddRecvStream: {id:36118ade-d8e1-4eef-b462-d916f46a1ae6;ssrcs:[1120370049];ssrc_groups:;cname:wsr+XpJJJiDlk07G;sync_label:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP} (neteq_impl.cc:109): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, background_noise_mode=2, playout_mode=0, enable_fast_accelerate=false, enable_muted_state= true (audio_receive_stream.cc:71): AudioReceiveStream: {rtp: {remote_ssrc: 1120370049, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP} (call.cc:1057): UpdateAggregateNetworkState: aggregate_state=down (send_side_congestion_controller.cc:185): SignalNetworkState Down (paced_sender.cc:279): PacedSender paused. (webrtcvoiceengine.cc:1268): Stopping playout for channel #0 (webrtcvoiceengine.cc:1268): Stopping playout for channel #0 (channel.cc:1396): Add remote ssrc: 1120370049 (channel.cc:1734): Changing voice state, recv=0 send=0 (channel.cc:2065): Setting remote video description (webrtcvideoengine2.cc:777): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:H264], VideoCodec[101:rtx], VideoCodec[102:H264], VideoCodec[123:rtx], VideoCodec[127:H264], VideoCodec[122:rtx], VideoCodec[125:H264], VideoCodec[107:rtx], VideoCodec[108:red], VideoCodec[109:rtx], VideoCodec[124:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}], max_bandwidth_bps: -1, } (webrtcvideoengine2.cc:621): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]} (webrtcmediaengine.cc:198): Unsupported RTP extension: {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8} (webrtcvideoengine2.cc:786): Using codec: VideoCodec[96:VP8] (webrtcvideoengine2.cc:834): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. (webrtcvideoengine2.cc:1210): AddRecvStream: {id:50db3d62-fb44-4ff5-bc56-fc0819bc33f6;ssrcs:[4265947015,2616070878];ssrc_groups:{semantics:FID;ssrcs:[4265947015,2616070878]};cname:wsr+XpJJJiDlk07G;sync_label:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP} (video_receive_stream.cc:198): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 4265947015, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 102, red_payload_type: 100, red_rtx_payload_type: 101}, rtx_ssrc: 2616070878, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP, pre_decode_callback: nullptr, target_delay_ms: 0} (call.cc:1057): UpdateAggregateNetworkState: aggregate_state=down (send_side_congestion_controller.cc:185): SignalNetworkState Down (paced_sender.cc:279): PacedSender paused. (channel.cc:1396): Add remote ssrc: 4265947015 (channel.cc:1987): Changing video state, send=0 (webrtcsession.cc:661): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. (webrtcvoiceengine.cc:2034): SetOutputVolume() to 1 for recv stream with ssrc 1120370049 (webrtcvideoengine2.cc:1326): SetSink: ssrc:4265947015 (ptr) 22:26:16 [debug] [peer.cpp(225)] 8iiSz-xKqmbDEPrQAAAC: On add stream 22:26:16 [debug] [peermanager.cpp(64)] Received offer: v=0 o=- 661132907157492103 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:wQ5X a=ice-pwd:xjtP77gg3+o2wTalgZZi1Oyf a=ice-options:trickle a=fingerprint:sha-256 53:68:A4:6D:51:22:D7:D6:AD:83:CD:06:D8:71:58:3A:08:45:A4:7D:A7:35:70:5B:16:1E:8D:DF:28:18:B8:6D a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:1120370049 cname:wsr+XpJJJiDlk07G a=ssrc:1120370049 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 36118ade-d8e1-4eef-b462-d916f46a1ae6 a=ssrc:1120370049 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:1120370049 label:36118ade-d8e1-4eef-b462-d916f46a1ae6 m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:wQ5X a=ice-pwd:xjtP77gg3+o2wTalgZZi1Oyf a=ice-options:trickle a=fingerprint:sha-256 53:68:A4:6D:51:22:D7:D6:AD:83:CD:06:D8:71:58:3A:08:45:A4:7D:A7:35:70:5B:16:1E:8D:DF:28:18:B8:6D a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:4 urn:3gpp:video-orientation a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 goog-remb a=rtcp-fb:96 transport-cc a=rtcp-fb:96 ccm fir a=rtcp-fb:96 nack a=rtcp-fb:96 nack pli a=rtpmap:97 rtx/90000 a=fmtp:97 apt=96 a=rtpmap:98 VP9/90000 a=rtcp-fb:98 goog-remb a=rtcp-fb:98 transport-cc a=rtcp-fb:98 ccm fir a=rtcp-fb:98 nack a=rtcp-fb:98 nack pli a=rtpmap:99 rtx/90000 a=fmtp:99 apt=98 a=rtpmap:100 H264/90000 a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:101 rtx/90000 a=fmtp:101 apt=100 a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc a=rtcp-fb:102 ccm fir a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f a=rtpmap:123 rtx/90000 a=fmtp:123 apt=102 a=rtpmap:127 H264/90000 a=rtcp-fb:127 goog-remb a=rtcp-fb:127 transport-cc a=rtcp-fb:127 ccm fir a=rtcp-fb:127 nack a=rtcp-fb:127 nack pli a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:122 rtx/90000 a=fmtp:122 apt=127 a=rtpmap:125 H264/90000 a=rtcp-fb:125 goog-remb a=rtcp-fb:125 transport-cc a=rtcp-fb:125 ccm fir a=rtcp-fb:125 nack a=rtcp-fb:125 nack pli a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:107 rtx/90000 a=fmtp:107 apt=125 a=rtpmap:108 red/90000 a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 a=rtpmap:124 ulpfec/90000 a=ssrc-group:FID 4265947015 2616070878 a=ssrc:4265947015 cname:wsr+XpJJJiDlk07G a=ssrc:4265947015 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:4265947015 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:4265947015 label:50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:2616070878 cname:wsr+XpJJJiDlk07G a=ssrc:2616070878 msid:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP 50db3d62-fb44-4ff5-bc56-fc0819bc33f6 a=ssrc:2616070878 mslabel:P3E84MQRZntpSVMGY5jq4GzpXIpjmlN6RmhP a=ssrc:2616070878 label:50db3d62-fb44-4ff5-bc56-fc0819bc33f6

22:26:16 [debug] [signaler.cpp(89)] Peer message: demo|8iiSz-xKqmbDEPrQAAAC 22:26:16 [debug] [peermanager.cpp(85)] Received candidate: candidate:3200290655 1 udp 2122260223 192.168.3.43 63340 typ host generation 0 ufrag wQ5X network-id 1 network-cost 10 (webrtcsession.cc:1160): ProcessIceMessage: Not ready to use candidate.

auscaster commented 6 years ago

Sorry for the delay here. I haven't seen this before, but WebRTC has just gone a big update, perhaps try updating your WebRTC and libsourcey builds?