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Created May 9, 2014 23:19
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node-webrtc #116
Raw
gistfile1.txt
Server running at http://0.0.0.0:9000/
/peer.html
/dist/wrtc.js
/peer.js
ws connected
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
ISAC/32000/1 (104)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: ISAC/48000/1 (105)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
WebRtcVoiceEngine::Init
webrtc: Thread with name:Trace started
webrtc: Thread with name:ProcessThread started
webrtc: CheckPlatform
webrtc: current platform is LINUX
webrtc: CreatePlatformSpecificObjects
webrtc: output: kPlatformDefaultAudio
webrtc: attempting to use the Linux PulseAudio APIs...
Error(webrtcvideoengine.cc:1552): webrtc: failed to connect context, error=-1
Error(webrtcvideoengine.cc:1552): webrtc: failed to initialize PulseAudio
webrtc: Close
webrtc: CloseSpeaker
webrtc: CloseMicrophone
Warning(webrtcvideoengine.cc:1552): webrtc: Linux PulseAudio is *not* supported => ALSA APIs will be utilized instead
webrtc: AttachAudioBuffer
webrtc: OS info: Linux
Warning(webrtcvideoengine.cc:1552): webrtc: failed to open X display, typing detection will not work
webrtc: number of availiable audio output devices is 0
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=)
webrtc: snd_mixer_attach(_outputMixerHandle, )
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
webrtc: Init() failed to initialize the speaker (error=9005)
webrtc: number of availiable audio input devices is 0
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=)
webrtc: snd_mixer_attach(_inputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
webrtc: Init() failed to initialize the microphone (error=9004)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=)
webrtc: snd_mixer_attach(_outputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitSpeaker() failed
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
webrtc: InitPlayout open ()
Error(webrtcvideoengine.cc:1552): webrtc: unable to open playback device: No such file or directory (-2)
webrtc: output: available=0
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=)
webrtc: snd_mixer_attach(_inputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitMicrophone() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: InitRecording open ()
Error(webrtcvideoengine.cc:1552): webrtc: unable to open record device: No such file or directory
webrtc: output: available=0
webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0xd856d110)
WebRtc VoiceEngine Version:
webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0xd856d110)
VoiceEngine 4.1.0
Build: May 8 2014 06:21:13 d
External recording and playout build
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, experimental_acm: false, }
ACM2 enabled? 0
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 1
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
ISAC/32000/1 (104)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.52.0
Build: May 8 2014 06:21:18 d
VideoEngine Init done
webrtc: (vie_base_impl.cc:68): virtual int webrtc::ViEBaseImpl::SetVoiceEngine(webrtc::VoiceEngine*): SetVoiceEngine
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, experimental_acm: false, }
ACM2 enabled? 0
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 1
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Setting microphone to (id=0, name=Default device) and speaker to (id=0, name=Default device)
Warning(webrtcvideoengine.cc:1552): webrtc: SetRecordingChannel() unable to set the recording channel (error=10028)
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: SetRecordingDevice() cannot access microphone (error=9004)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitMicrophone() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: unable to open record device: No such file or directory
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: SetPlayoutDevice() cannot access speaker (error=9005)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitSpeaker() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Set microphone to (id=0 name=Default device) and speaker to (id=0 name=Default device)
Error(webrtcvideoengine.cc:1552): webrtc: unable to open playback device: No such file or directory (-2)
Allowing SCTP data engine.
Generating identity.
{ type: 'offer',
sdp: 'v=0\r\no=- 6115600490265182459 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio data\r\na=msid-semantic: WMS\r\nm=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:8ut3T0oJvcPWY7Wt\r\na=ice-pwd:om52it2L0lRc9Snnsh07GU1t\r\na=ice-options:google-ice\r\na=fingerprint:sha-256 B8:F3:A7:AC:7D:69:45:C6:89:94:A6:94:F1:05:CC:2C:F5:5A:4E:DA:32:BF:18:A7:D0:88:B0:1A:15:66:10:D0\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=recvonly\r\na=rtcp-mux\r\na=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:FtoTsSkqSAmED7oqEJDE5PvEh0xcfin3kii2FDvv\r\na=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PBghO02bfJS1vS5wncbkjZwLj2z1SJO7yS0LGhAY\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 minptime=10\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=maxptime:60\r\nm=application 1 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:8ut3T0oJvcPWY7Wt\r\na=ice-pwd:om52it2L0lRc9Snnsh07GU1t\r\na=ice-options:google-ice\r\na=fingerprint:sha-256 B8:F3:A7:AC:7D:69:45:C6:89:94:A6:94:F1:05:CC:2C:F5:5A:4E:DA:32:BF:18:A7:D0:88:B0:1A:15:66:10:D0\r\na=setup:actpass\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n' }
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Ignored line: a=sctpmap:5000 webrtc-datachannel 1024
ParseMediaDescription: Got SCTP Port Number 5000
Created channel for audio
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
Created channel for data
Session:5095903880647653194 Old state:STATE_INIT New state:STATE_RECEIVEDINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting remote voice description
WebRtcVoiceMediaChanne::SetSendBandwidth.
WebRtcVoiceMediaChannel::SetSendBandwidthInternal.
The send codec has not been set up yet. The send bandwidth setting will be applied later.
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
Changing voice state, recv=0 send=0
Setting SCTP remote data description
Changing data state, recv=0 send=0
Local and Remote descriptions must be applied to get SSL Role of the session.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
Local and Remote descriptions must be applied to get SSL Role of the session.
Video is not available in the offer.
signaling state change: { type: 'signalingstatechange' }
{ type: 'answer',
sdp: 'v=0\r\no=- 5095903880647653194 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio data\r\na=msid-semantic: WMS\r\nm=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:GM51ciqMwHJ9Quut\r\na=ice-pwd:nsBEyb8WDMoOunyr68iA8Xc+\r\na=fingerprint:sha-1 F1:97:19:CE:3E:EB:EB:F4:8B:F6:12:01:1F:32:91:12:BA:D6:13:8E\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 minptime=10\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=maxptime:60\r\nm=application 1 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:GM51ciqMwHJ9Quut\r\na=ice-pwd:nsBEyb8WDMoOunyr68iA8Xc+\r\na=fingerprint:sha-1 F1:97:19:CE:3E:EB:EB:F4:8B:F6:12:01:1F:32:91:12:BA:D6:13:8E\r\na=setup:active\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n' }
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Ignored line: a=sctpmap:5000 webrtc-datachannel 1024
ParseMediaDescription: Got SCTP Port Number 5000
Jingle:Channel[audio|1|__]: DTLS setup complete.
Jingle:Channel[audio|2|__]: DTLS setup complete.
Jingle:Channel[data|1|__]: DTLS setup complete.
Destroying NSS identity
Enabling BUNDLE, bundling onto transport: audio
Channel enabled
Changing voice state, recv=0 send=0
Channel enabled
Changing data state, recv=0 send=0
Session:5095903880647653194 Old state:STATE_RECEIVEDINITIATE New state:STATE_SENTACCEPT Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting local voice description
Destroying NSS identity
Setting receive voice codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
ISAC/32000/1 (104)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
telephone-event/8000/1 (126)
Changing voice state, recv=0 send=0
Setting local data description
Changing data state, recv=1 send=0
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[294083896:2:udp:2122260223:192.168.233.1:60239:local::0:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[324927998:2:udp:2122194687:172.16.100.155:60240:local::0:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[3879585599:2:udp:2122129151:192.168.226.1:60241:local::0:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[2886211718:2:udp:2122063615:172.16.100.119:60242:local::0:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[3755163245:2:udp:1685987071:203.44.30.66:60240:stun:172.16.100.155:60240:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[1619339029:2:udp:1685855999:203.44.30.66:60242:stun:172.16.100.119:60242:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t] for content: audio
Transport: audio, allocating candidates
Transport: audio, allocating candidates
Session:5095903880647653194 Old state:STATE_SENTACCEPT New state:STATE_INPROGRESS Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
awaiting data channels
signaling state change: { type: 'signalingstatechange' }
ice connection state change: { type: 'iceconnectionstatechange' }
ice gathering state change: { type: 'icegatheringstatechange' }
ice gathering state change: { type: 'icegatheringstatechange' }
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[eth0:50.116.7.0/24]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0::Net[eth0:50.116.7.0/24]]: Added port to allocator
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.233.1:60239|C--W|9114475305677766143|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (1 total)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.155:60240|C--W|9114475305677635071|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (2 total)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.226.1:60241|C--W|9114475305677503998|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (3 total)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.119:60242|C--W|9114193830700793342|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (4 total)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60240|C--W|7241259335668088318|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (5 total)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60242|C--W|7240696385714667006|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (6 total)
AllocationSequence: UDPPort will be handling the STUN candidate generation.
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Relay
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[eth0:50.116.7.0/24]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:local:Net[eth0:50.116.7.0/24]]: Added port to allocator
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=SslTcp
All candidates gathered for audio:1:0
Transport: audio, component 1 allocation complete
Transport: audio allocation complete
Candidate gathering is complete.
ice gathering state change: { type: 'icegatheringstatechange' }
Jingle:Channel[audio|1|R_]: New best connection: Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60240|CRWS|7241259335668088318|2291]
BeginSSL: with peer
BeginSSL: as client
ContinueSSL
Would have blocked
Timeout is 1000 ms
Jingle:Channel[audio|1|__]: DtlsTransportChannelWrapper: Started DTLS handshake
NSSStreamAdapter::OnEvent SE_READ
ContinueSSL
NSSStreamAdapter::AuthCertificateHook
Checking against specified digest
Accepted peer certificate
Client cert requested
Would have blocked
Timeout is 1000 ms
NSSStreamAdapter::OnEvent SE_READ
ContinueSSL
Would have blocked
Timeout is 1000 ms
NSSStreamAdapter::OnEvent SE_READ
ContinueSSL
Handshake complete
Jingle:Channel[audio|1|__]: DTLS handshake complete.
Channel socket writable (audio, 1) for the first time
Using Cand[3462590651:1:udp:2122129151:50.116.7.95:41562:local::0:GM51ciqMwHJ9Quut:nsBEyb8WDMoOunyr68iA8Xc+]->Cand[3755163245:1:udp:1685987071:203.44.30.66:60240:stun:172.16.100.155:60240:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t]
Installing keys from DTLS-SRTP on audio RTP
WARNING: no real random source present!
ice connection state change: { type: 'iceconnectionstatechange' }
SRTP activated with negotiated parameters: send cipher_suite AES_CM_128_HMAC_SHA1_32 recv cipher_suite AES_CM_128_HMAC_SHA1_32
Changing voice state, recv=0 send=1
Channel socket writable (data, 1) for the first time
Using Cand[3462590651:1:udp:2122129151:50.116.7.95:41562:local::0:GM51ciqMwHJ9Quut:nsBEyb8WDMoOunyr68iA8Xc+]->Cand[3755163245:1:udp:1685987071:203.44.30.66:60240:stun:172.16.100.155:60240:8ut3T0oJvcPWY7Wt:om52it2L0lRc9Snnsh07GU1t]
Changing data state, recv=1 send=1
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=2
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=100, seqnum=5000, SSRC=240712049
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=2
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=404, seqnum=5000, SSRC=3793681724
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=2
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=412, seqnum=5000, SSRC=553333326
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Sent OPEN_ACK message on channel 1
ondatachannel reliable open
onopen
complete
onmessage: { '0': 97,
'1': 99,
'2': 107,
'3': 0,
slice: [Function: slice],
byteLength: 4 }
onmessage: Hello bridge!
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=2
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=16, seqnum=5000, SSRC=1140448212
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=7
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=28, seqnum=5000, SSRC=1013618060
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=7
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=28, seqnum=5000, SSRC=1571816545
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
/peer.html
/dist/wrtc.js
/peer.js
ws connected
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Warning(srtpfilter.cc:569): Failed to unprotect SRTP packet, err=2
Error(channel.cc:602): Failed to unprotect audio RTP packet: size=48, seqnum=5000, SSRC=33273324
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
NSSStreamAdapter::OnEvent SE_READ
-- onStreamReadable
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
Error(webrtcsdp.cc:356): Failed to parse: "undefined". Reason: Expect line: candidate:<candidate-str>
Error(webrtcsession.cc:881): ProcessIceMessage: Candidate is NULL
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
ISAC/32000/1 (104)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: ISAC/48000/1 (105)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:501): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
WebRtcVoiceEngine::Init
webrtc: Thread with name:ProcessThread started
webrtc: CheckPlatform
webrtc: current platform is LINUX
webrtc: CreatePlatformSpecificObjects
webrtc: output: kPlatformDefaultAudio
webrtc: attempting to use the Linux PulseAudio APIs...
Error(webrtcvideoengine.cc:1552): webrtc: failed to connect context, error=-1
Error(webrtcvideoengine.cc:1552): webrtc: failed to initialize PulseAudio
webrtc: Close
webrtc: CloseSpeaker
webrtc: CloseMicrophone
Warning(webrtcvideoengine.cc:1552): webrtc: Linux PulseAudio is *not* supported => ALSA APIs will be utilized instead
webrtc: AttachAudioBuffer
webrtc: OS info: Linux
Warning(webrtcvideoengine.cc:1552): webrtc: failed to open X display, typing detection will not work
webrtc: number of availiable audio output devices is 0
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=)
webrtc: snd_mixer_attach(_outputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
webrtc: Init() failed to initialize the speaker (error=9005)
webrtc: number of availiable audio input devices is 0
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=)
webrtc: snd_mixer_attach(_inputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
webrtc: Init() failed to initialize the microphone (error=9004)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenSpeaker(name=)
webrtc: snd_mixer_attach(_outputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitSpeaker() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: InitPlayout open ()
Error(webrtcvideoengine.cc:1552): webrtc: unable to open playback device: No such file or directory (-2)
webrtc: output: available=0
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: AudioMixerManagerLinuxALSA::OpenMicrophone(name=)
webrtc: snd_mixer_attach(_inputMixerHandle, )
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitMicrophone() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
webrtc: InitRecording open ()
Error(webrtcvideoengine.cc:1552): webrtc: unable to open record device: No such file or directory
webrtc: output: available=0
webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0xcc0d45e0)
WebRtc VoiceEngine Version:
webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0xcc0d45e0)
VoiceEngine 4.1.0
Build: May 8 2014 06:21:13 d
External recording and playout build
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, experimental_acm: false, }
ACM2 enabled? 0
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 1
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
ISAC/32000/1 (104)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.52.0
Build: May 8 2014 06:21:18 d
VideoEngine Init done
webrtc: (vie_base_impl.cc:68): virtual int webrtc::ViEBaseImpl::SetVoiceEngine(webrtc::VoiceEngine*): SetVoiceEngine
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, experimental_acm: false, }
ACM2 enabled? 0
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 1
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Setting microphone to (id=0, name=Default device) and speaker to (id=0, name=Default device)
Warning(webrtcvideoengine.cc:1552): webrtc: SetRecordingChannel() unable to set the recording channel (error=10028)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: SetRecordingDevice() cannot access microphone (error=9004)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitMicrophone() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: unable to open record device: No such file or directory
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: SetPlayoutDevice() cannot access speaker (error=9005)
ALSA lib control.c:951:(snd_ctl_open_noupdate) Invalid CTL
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Error(webrtcvideoengine.cc:1552): webrtc: snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
Warning(webrtcvideoengine.cc:1552): webrtc: InitSpeaker() failed
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM
Error(webrtcvideoengine.cc:1552): webrtc: GetDevicesInfo - Could not find device name or numbers
Set microphone to (id=0 name=Default device) and speaker to (id=0 name=Default device)
Error(webrtcvideoengine.cc:1552): webrtc: unable to open playback device: No such file or directory (-2)
Allowing SCTP data engine.
Generating identity.
{ type: 'offer',
sdp: 'v=0\r\no=- 7297097371266589758 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio data\r\na=msid-semantic: WMS\r\nm=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:OCNF5FluaUUUuXIC\r\na=ice-pwd:004phmMoi+bb0WfNDrBSIQpB\r\na=ice-options:google-ice\r\na=fingerprint:sha-256 B8:F3:A7:AC:7D:69:45:C6:89:94:A6:94:F1:05:CC:2C:F5:5A:4E:DA:32:BF:18:A7:D0:88:B0:1A:15:66:10:D0\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=recvonly\r\na=rtcp-mux\r\na=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:b0+TW0IBmuZqJYd1IWJdjPrhL89AD+EAbHqAuZZ8\r\na=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:01tDgUDVjHuCPoG7FGSxGAzwDC4jfj1fWg2X4z3g\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 minptime=10\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=maxptime:60\r\nm=application 1 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:OCNF5FluaUUUuXIC\r\na=ice-pwd:004phmMoi+bb0WfNDrBSIQpB\r\na=ice-options:google-ice\r\na=fingerprint:sha-256 B8:F3:A7:AC:7D:69:45:C6:89:94:A6:94:F1:05:CC:2C:F5:5A:4E:DA:32:BF:18:A7:D0:88:B0:1A:15:66:10:D0\r\na=setup:actpass\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n' }
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Ignored line: a=sctpmap:5000 webrtc-datachannel 1024
ParseMediaDescription: Got SCTP Port Number 5000
Created channel for audio
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
Created channel for data
Session:4889178110677594274 Old state:STATE_INIT New state:STATE_RECEIVEDINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting remote voice description
WebRtcVoiceMediaChanne::SetSendBandwidth.
WebRtcVoiceMediaChannel::SetSendBandwidthInternal.
The send codec has not been set up yet. The send bandwidth setting will be applied later.
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
Changing voice state, recv=0 send=0
Setting SCTP remote data description
Changing data state, recv=0 send=0
Local and Remote descriptions must be applied to get SSL Role of the session.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
ProcessIceMessage: Local/Remote description not set on the Transport, save the candidate for later use.
Local and Remote descriptions must be applied to get SSL Role of the session.
Video is not available in the offer.
signaling state change: { type: 'signalingstatechange' }
{ type: 'answer',
sdp: 'v=0\r\no=- 4889178110677594274 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio data\r\na=msid-semantic: WMS\r\nm=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:+9kHSZ1doKMr9LGv\r\na=ice-pwd:KwTfIz2qNRHkqkRMEhjBxkFK\r\na=fingerprint:sha-1 3B:FD:74:46:74:A2:87:DA:51:AD:CF:A6:6D:A8:38:D4:A9:AE:E8:8C\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 minptime=10\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=maxptime:60\r\nm=application 1 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:+9kHSZ1doKMr9LGv\r\na=ice-pwd:KwTfIz2qNRHkqkRMEhjBxkFK\r\na=fingerprint:sha-1 3B:FD:74:46:74:A2:87:DA:51:AD:CF:A6:6D:A8:38:D4:A9:AE:E8:8C\r\na=setup:active\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n' }
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Ignored line: a=sctpmap:5000 webrtc-datachannel 1024
ParseMediaDescription: Got SCTP Port Number 5000
Jingle:Channel[audio|1|__]: DTLS setup complete.
Jingle:Channel[audio|2|__]: DTLS setup complete.
Jingle:Channel[data|1|__]: DTLS setup complete.
Destroying NSS identity
Enabling BUNDLE, bundling onto transport: audio
Channel enabled
Changing voice state, recv=0 send=0
Channel enabled
Changing data state, recv=0 send=0
Session:4889178110677594274 Old state:STATE_RECEIVEDINITIATE New state:STATE_SENTACCEPT Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting local voice description
Destroying NSS identity
Setting receive voice codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
ISAC/32000/1 (104)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
telephone-event/8000/1 (126)
Changing voice state, recv=0 send=0
Setting local data description
Changing data state, recv=1 send=0
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[294083896:2:udp:2122260223:192.168.233.1:60239:local::0:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[324927998:2:udp:2122194687:172.16.100.155:60240:local::0:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[3879585599:2:udp:2122129151:192.168.226.1:60241:local::0:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[2886211718:2:udp:2122063615:172.16.100.119:60242:local::0:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[3755163245:2:udp:1685987071:203.44.30.66:60240:stun:172.16.100.155:60240:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Warning(webrtcsession.cc:1433): Candidate has unknown component: Cand[1619339029:2:udp:1685855999:203.44.30.66:60242:stun:172.16.100.119:60242:OCNF5FluaUUUuXIC:004phmMoi+bb0WfNDrBSIQpB] for content: audio
Transport: audio, allocating candidates
Transport: audio, allocating candidates
signaling state change: { type: 'signalingstatechange' }
ice connection state change: { type: 'iceconnectionstatechange' }
ice gathering state change: { type: 'icegatheringstatechange' }
ice gathering state change: { type: 'icegatheringstatechange' }
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[eth0:50.116.7.0/24]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0::Net[eth0:50.116.7.0/24]]: Added port to allocator
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.233.1:60239|C--W|9114475305677766143|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (1 total)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.155:60240|C--W|9114475305677635071|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (2 total)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.226.1:60241|C--W|9114475305677503998|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (3 total)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.119:60242|C--W|9114193830700793342|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (4 total)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60240|C--W|7241259335668088318|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (5 total)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60242|C--W|7240696385714667006|-]: Connection created
Jingle:Channel[audio|1|__]: Created connection with origin=2, (6 total)
AllocationSequence: UDPPort will be handling the STUN candidate generation.
Session:4889178110677594274 Old state:STATE_SENTACCEPT New state:STATE_INPROGRESS Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
awaiting data channels
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Relay
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[eth0:50.116.7.0/24]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:local:Net[eth0:50.116.7.0/24]]: Added port to allocator
Jingle:Net[eth0:50.116.7.0/24]: Allocation Phase=SslTcp
All candidates gathered for audio:1:0
Transport: audio, component 1 allocation complete
Transport: audio allocation complete
Candidate gathering is complete.
ice gathering state change: { type: 'icegatheringstatechange' }
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.233.1:60239|C--I|9114475305677766143|-]: Timed out after 15287 ms without a response, rtt=3000
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.155:60240|C--I|9114475305677635071|-]: Timed out after 15239 ms without a response, rtt=3000
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.226.1:60241|C--I|9114475305677503998|-]: Timed out after 15191 ms without a response, rtt=3000
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.119:60242|C--I|9114193830700793342|-]: Timed out after 15142 ms without a response, rtt=3000
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.233.1:60239|C-xI|9114475305677766143|-]: Connection deleted
Jingle:Channel[audio|1|RW]: Removed connection (5 remaining)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.155:60240|C-xI|9114475305677635071|-]: Connection deleted
Jingle:Channel[audio|1|RW]: Removed connection (4 remaining)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:192.168.226.1:60241|C-xI|9114475305677503998|-]: Connection deleted
Jingle:Channel[audio|1|RW]: Removed connection (3 remaining)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.99:local:udp:172.16.100.119:60242|C-xI|9114193830700793342|-]: Connection deleted
Jingle:Channel[audio|1|RW]: Removed connection (2 remaining)
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60240|CRWI|7241259335668088318|252]: Timing-out STUN ping nWL9HFgrDiFg after 5000 ms
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60240|CRWI|7241259335668088318|252]: Unwritable after 5 ping failures and 5288 ms without a response, ms since last received ping=6147 ms since last received data=5584 rtt=504
Channel socket not writable (audio, 1)
Changing voice state, recv=0 send=1
Channel socket not writable (data, 1)
Changing data state, recv=1 send=1
ice connection state change: { type: 'iceconnectionstatechange' }
Jingle:Conn[audio:ahorSc4a:1:0:local:udp:50.116.7.95:41562->:1:0.78:stun:udp:203.44.30.66:60240|CRwI|7241259335668088318|252]: Timed out after 15048 ms without a response, rtt=504
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.233.1:60239|C--I|9114475305677766143|-]: Timed out after 15040 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.233.1:60239|C-xI|9114475305677766143|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (5 remaining)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.155:60240|C--I|9114475305677635071|-]: Timed out after 15041 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.155:60240|C-xI|9114475305677635071|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (4 remaining)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.226.1:60241|C--I|9114475305677503998|-]: Timed out after 15041 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:192.168.226.1:60241|C-xI|9114475305677503998|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (3 remaining)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.119:60242|C--I|9114193830700793342|-]: Timed out after 15040 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.99:local:udp:172.16.100.119:60242|C-xI|9114193830700793342|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (2 remaining)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60240|C--I|7241259335668088318|-]: Timed out after 15040 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60240|C-xI|7241259335668088318|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (1 remaining)
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60242|C--I|7240696385714667006|-]: Timed out after 15040 ms without a response, rtt=3000
Jingle:Conn[audio:qWNXBVoj:1:0:local:udp:50.116.7.95:52388->:1:0.78:stun:udp:203.44.30.66:60242|C-xI|7240696385714667006|-]: Connection deleted
Jingle:Channel[audio|1|__]: Removed connection (0 remaining)
@Globik
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Globik commented Apr 2, 2016

At which port is your websocket here? At 9000 or 9001?

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