[2017-05-19 18:24:09] [devel] endpoint constructor
[2017-05-19 18:24:09] [devel] client constructor
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Date: Fri, 19 May 2017 10:24:11 GMT
Connection: close

[2017-05-19 18:24:11] [connect] Successful connection
[2017-05-19 18:24:11] [connect] WebSocket Connection 192.168.1.26:8080 v-2 "WebSocket++/0.7.0" /socket.io/?EIO=4&transport=websocket&t=1495189451 101
Connected.
On handshake,sid:YPlDGl8VpY2KQcltAAAF,ping interval:25000,ping timeout60000
Received Message type (Connect)
encoded payload length:410
Received Message type (ACK)
(conferenceclient.cc:138): Find streams in the conference.
(conferenceclient.cc:606): OnStreamAdded: mixed stream.

url: rtsp://admin:agv10086@192.168.1.65:554, transport::udp, buffer_size: 2097152
[udp @ 06dae420] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 06dae4e0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
Input #0, rtsp, from 'rtsp://admin:agv10086@192.168.1.65:554':
  Metadata:
    title           : H3C IPC Realtime stream
  Duration: N/A, start: 0.240000, bitrate: N/A
    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080, 25 fps, 25 tbr, 90k tbn, 50 tbc
(peerconnectiondependencyfactory.cc:156): PeerConnectionDependencyFactory::CreatePeerConnectionFactory()
(peerconnectiondependencyfactory.cc:101): CreatePeerConnectionOnCurrentThread
(webrtcvoiceengine.cc:514): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtcvoiceengine.cc:521): Supported send codecs in order of preference:
(webrtcvoiceengine.cc:276): Adding supported codec: opus/48000/2 (111)
(webrtcvoiceengine.cc:276): Adding supported codec: ISAC/16000/1 (103)
(webrtcvoiceengine.cc:276): Adding supported codec: ISAC/32000/1 (104)
(webrtcvoiceengine.cc:276): Adding supported codec: G722/8000/1 (9)
(webrtcvoiceengine.cc:276): Adding supported codec: ILBC/8000/1 (102)
(webrtcvoiceengine.cc:276): Adding supported codec: PCMU/8000/1 (0)
(webrtcvoiceengine.cc:276): Adding supported codec: PCMA/8000/1 (8)
(webrtcvoiceengine.cc:276): Adding supported codec: CN/32000/1 (106)
(webrtcvoiceengine.cc:276): Adding supported codec: CN/16000/1 (105)
(webrtcvoiceengine.cc:276): Adding supported codec: CN/8000/1 (13)
(webrtcvoiceengine.cc:276): Adding supported codec: telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:524): opus/48000/2 (111)
(webrtcvoiceengine.cc:524): ISAC/16000/1 (103)
(webrtcvoiceengine.cc:524): ISAC/32000/1 (104)
(webrtcvoiceengine.cc:524): G722/8000/1 (9)
(webrtcvoiceengine.cc:524): ILBC/8000/1 (102)
(webrtcvoiceengine.cc:524): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:524): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:524): CN/32000/1 (106)
(webrtcvoiceengine.cc:524): CN/16000/1 (105)
(webrtcvoiceengine.cc:524): CN/8000/1 (13)
(webrtcvoiceengine.cc:524): telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:527): Supported recv codecs in order of preference:
(webrtcvoiceengine.cc:530): opus/48000/2 (111)
(webrtcvoiceengine.cc:530): isac/16000/1 (103)
(webrtcvoiceengine.cc:530): isac/32000/1 (104)
(webrtcvoiceengine.cc:530): G722/8000/1 (9)
(webrtcvoiceengine.cc:530): iLBC/8000/1 (102)
(webrtcvoiceengine.cc:530): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:530): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:530): cn/32000/1 (106)
(webrtcvoiceengine.cc:530): cn/16000/1 (105)
(webrtcvoiceengine.cc:530): cn/8000/1 (13)
(webrtcvoiceengine.cc:530): telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:538): VoiceEngine 4.1.0
(audio_device_impl.cc:84): webrtc::AudioDeviceModule::Create
(audio_device_buffer.cc:63): AudioDeviceBuffer::ctor
(audio_device_impl.cc:131): webrtc::AudioDeviceModuleImpl::AudioDeviceModuleImpl
(audio_device_impl.cc:139): webrtc::AudioDeviceModuleImpl::CheckPlatform
(audio_device_impl.cc:147): current platform is Win32
(audio_device_impl.cc:180): webrtc::AudioDeviceModuleImpl::CreatePlatformSpecificObjects
(audio_device_impl.cc:1870): webrtc::AudioDeviceModuleImpl::PlatformAudioLayer
(audio_device_impl.cc:215): attempting to use the Windows Core Audio APIs...
(audio_device_impl.cc:220): Windows Core Audio APIs will be utilized
(audio_device_impl.cc:360): webrtc::AudioDeviceModuleImpl::AttachAudioBuffer
(audio_device_buffer.cc:119): SetRecordingSampleRate(0)
(audio_device_buffer.cc:126): SetPlayoutSampleRate(0)
(audio_device_buffer.cc:141): SetRecordingChannels(0)
(audio_device_buffer.cc:150): SetPlayoutChannels(0)
(audio_device_impl.cc:1471): webrtc::AudioDeviceModuleImpl::RegisterEventObserver
(audio_device_impl.cc:1484): webrtc::AudioDeviceModuleImpl::RegisterAudioCallback
(audio_device_buffer.cc:91): webrtc::AudioDeviceBuffer::RegisterAudioCallback
(audio_device_impl.cc:488): webrtc::AudioDeviceModuleImpl::Init
(audio_device_impl.cc:1227): webrtc::AudioDeviceModuleImpl::SetPlayoutDevice
(audio_device_impl.cc:537): webrtc::AudioDeviceModuleImpl::InitSpeaker
(audio_device_impl.cc:1322): webrtc::AudioDeviceModuleImpl::SetRecordingDevice
(audio_device_impl.cc:547): webrtc::AudioDeviceModuleImpl::InitMicrophone
(audio_device_impl.cc:1026): webrtc::AudioDeviceModuleImpl::StereoPlayoutIsAvailable
(audio_device_impl.cc:1036): output: 1
(audio_device_impl.cc:1045): webrtc::AudioDeviceModuleImpl::SetStereoPlayout(1)
(audio_device_buffer.cc:150): SetPlayoutChannels(2)
(audio_device_impl.cc:912): webrtc::AudioDeviceModuleImpl::StereoRecordingIsAvailable
(audio_device_impl.cc:922): output: 1
(audio_device_impl.cc:931): webrtc::AudioDeviceModuleImpl::SetStereoRecording(1)
(audio_device_buffer.cc:141): SetRecordingChannels(2)
(audio_device_impl.cc:1092): webrtc::AudioDeviceModuleImpl::SetAGC(1)
(webrtcvoiceengine.cc:603): 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, }
(audio_device_impl.cc:1786): webrtc::AudioDeviceModuleImpl::BuiltInAECIsAvailable
(audio_device_generic.cc:51): webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable: Not supported on this platform
(audio_device_impl.cc:1789): output: 0
(webrtcvoiceengine.cc:678): Echo control set to 1 with mode 2
(audio_device_impl.cc:1802): webrtc::AudioDeviceModuleImpl::BuiltInAGCIsAvailable
(audio_device_generic.cc:61): webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable: Not supported on this platform
(audio_device_impl.cc:1805): output: 0
(audio_device_impl.cc:1092): webrtc::AudioDeviceModuleImpl::SetAGC(1)
(webrtcvoiceengine.cc:712): Auto gain set to 1 with mode 2
(audio_device_impl.cc:1818): webrtc::AudioDeviceModuleImpl::BuiltInNSIsAvailable
(audio_device_generic.cc:71): webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable: Not supported on this platform
(audio_device_impl.cc:1821): output: 0
(webrtcvoiceengine.cc:767): Noise suppression set to 1 with mode 5
(webrtcvoiceengine.cc:773): High pass filter enabled? 1
(webrtcvoiceengine.cc:781): Stereo swapping enabled? 0
(webrtcvoiceengine.cc:790): NetEq capacity is 50
(webrtcvoiceengine.cc:798): NetEq fast mode? 0
(webrtcvoiceengine.cc:806): Typing detection is enabled? 1
(webrtcvoiceengine.cc:815): Adjust agc delta is 0
(webrtcvoiceengine.cc:1006): Adjusting AGC level from default -2dB to -2dB
(webrtcvoiceengine.cc:826): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:835): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:844): Experimental ns is enabled? 0
(webrtcvoiceengine.cc:850): Intelligibility Enhancer is enabled? 0
(webrtcvoiceengine.cc:860): Level control: 0
(webrtcvoiceengine.cc:896): Setting microphone to (id=-1) and speaker to (id=-1)
(audio_device_impl.cc:1460): webrtc::AudioDeviceModuleImpl::Recording
(audio_device_impl.cc:978): webrtc::AudioDeviceModuleImpl::SetRecordingChannel(both)
(audio_device_impl.cc:1322): webrtc::AudioDeviceModuleImpl::SetRecordingDevice
(audio_device_impl.cc:547): webrtc::AudioDeviceModuleImpl::InitMicrophone
(audio_device_impl.cc:912): webrtc::AudioDeviceModuleImpl::StereoRecordingIsAvailable
(audio_device_impl.cc:922): output: 1
(audio_device_impl.cc:931): webrtc::AudioDeviceModuleImpl::SetStereoRecording(1)
(audio_device_buffer.cc:141): SetRecordingChannels(2)
(audio_device_impl.cc:1420): webrtc::AudioDeviceModuleImpl::Playing
(audio_device_impl.cc:1227): webrtc::AudioDeviceModuleImpl::SetPlayoutDevice
(audio_device_impl.cc:537): webrtc::AudioDeviceModuleImpl::InitSpeaker
(audio_device_impl.cc:1026): webrtc::AudioDeviceModuleImpl::StereoPlayoutIsAvailable
(audio_device_impl.cc:1036): output: 1
(audio_device_impl.cc:1045): webrtc::AudioDeviceModuleImpl::SetStereoPlayout(1)
(audio_device_buffer.cc:150): SetPlayoutChannels(2)
(webrtcvoiceengine.cc:915): Set microphone to (id=-1) and speaker to (id=-1)
(webrtcvideoengine2.cc:554): WebRtcVideoEngine2::WebRtcVideoEngine2()
(webrtcvideoengine2.cc:636): Supported codecs: {VideoCodec[100:VP8:640:400:30], VideoCodec[96:rtx:0:0:0], VideoCodec[101:VP9:640:400:30], VideoCodec[97:rtx:0:0:0], VideoCodec[116:red:0:0:0], VideoCodec[98:rtx:0:0:0], VideoCodec[117:ulpfec:0:0:0]}
(webrtcvideoengine2.cc:672): Supported codecs (incl. external codecs): {VideoCodec[100:VP8:640:400:30], VideoCodec[96:rtx:0:0:0], VideoCodec[101:VP9:640:400:30], VideoCodec[97:rtx:0:0:0], VideoCodec[116:red:0:0:0], VideoCodec[98:rtx:0:0:0], VideoCodec[117:ulpfec:0:0:0], VideoCodec[120:H264:1920:1080:60], VideoCodec[99:rtx:0:0:0]}
(webrtcvideoengine2.cc:674): Codecs supported by the external encoder factory: H264, VP8
(webrtcvideoengine2.cc:563): WebRtcVideoEngine2::Init
(messagequeue.cc:520): Message took 89ms to dispatch. Posted from: webrtc::CreatePeerConnectionFactory@c:\cppsdk\src\webrtc\api\peerconnectionfactory.cc:75
(peerconnectiondependencyfactory.cc:141): CreatePeerConnectionOnCurrentThread finished.
(customizedframescapturer.cc:154): Yuv Frame Generator started
Executing GetNextFrameSize
(opensslidentity.cc:43): Making key pair
Executing GenerateNextFrame
key frame
Receive video frame packet with size 46070


#
# Fatal error in c:\cppsdk\src\talk\woogeen\sdk\base\customizedframescapturer.cc, line 210
# last system error: 0
# Check failed: false
#
#
(opensslidentity.cc:84): Returning key pair
(opensslidentity.cc:91): Making certificate for WebRTC
(opensslidentity.cc:139): Returning certificate
(peerconnectionchannel.cc:19): PeerConnectionChannel video codec: 2
(peerconnectionchannel.cc:32): Initialize PeerConnection.
(webrtcsessiondescriptionfactory.cc:328): Using certificate supplied to the constructor.
(conferencepeerconnectionchannel.cc:309): Publish a local stream.
(conferencepeerconnectionchannel.cc:713): Failed to get resolution info. Retry after 1 second(s).
(conferencepeerconnectionchannel.cc:713): Failed to get resolution info. Retry after 1 second(s).
(conferencepeerconnectionchannel.cc:713): Failed to get resolution info. Retry after 1 second(s).
(conferencepeerconnectionchannel.cc:713): Failed to get resolution info. Retry after 1 second(s).
(conferencepeerconnectionchannel.cc:713): Failed to get resolution info. Retry after 1 second(s).
(conferencepeerconnectionchannel.cc:708): Failed to get resolution info.
encoded payload length:100
Received Message type (ACK)
(conferencesocketsignalingchannel.cc:374): Received ack from server.
(conferencepeerconnectionchannel.cc:657): Setting stream ID 754748542793095200
(conferencepeerconnectionchannel.cc:111): Create offer.
(conferencepeerconnectionchannel.cc:124): Post create offer
(mediasession.cc:383): Duplicate id found. Reassigning from 101 to 127
(conferencepeerconnectionchannel.cc:221): Create sdp success.
(conferencepeerconnectionchannel.cc:232): Post set local desc
(sdputils.cc:206): RTP map for OPUS is not found. SDP: v=0
o=- 7347636247221083791 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE video
a=msid-semantic: WMS MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1
m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 120 96 97 98 99
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:S9h4
a=ice-pwd:0qqSiajKhYOaXz3ERSOBvOKy
a=fingerprint:sha-256 90:96:38:E0:B0:09:1A:0B:8E:FD:BB:F3:07:34:01:CA:4A:5D:E1:32:BA:1B:F5:6E:64:86:46:43:3B:83:A5:1F
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:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=sendonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm fir
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 transport-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:120 H264/90000
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=101
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=116
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=120
a=ssrc-group:FID 780220367 3434847234
a=ssrc:780220367 cname:/fW5hikIIzRBshqf
a=ssrc:780220367 msid:MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1 VideoTrack-62e86b6a-6045-4536-9f95-914e0f624349
a=ssrc:780220367 mslabel:MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1
a=ssrc:780220367 label:VideoTrack-62e86b6a-6045-4536-9f95-914e0f624349
a=ssrc:3434847234 cname:/fW5hikIIzRBshqf
a=ssrc:3434847234 msid:MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1 VideoTrack-62e86b6a-6045-4536-9f95-914e0f624349
a=ssrc:3434847234 mslabel:MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1
a=ssrc:3434847234 label:VideoTrack-62e86b6a-6045-4536-9f95-914e0f624349

(sdputils.cc:210): RTP map found for H264: a=rtpmap:120 H264/90000
(sdputils.cc:242): New m-line: m=video 9 UDP/TLS/RTP/SAVPF 120 100 101 116 117 96 97 98 99
(webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
(systeminfo.cc:82): Available number of cores: 8
(bitrate_prober.cc:46): Bandwidth probing enabled, set to inactive
(remote_bitrate_estimator_single_stream.cc:61): RemoteBitrateEstimatorSingleStream: Instantiating.
(bitrate_prober.cc:74): Probe cluster (bitrate:packets): (900000:6)
(bitrate_prober.cc:74): Probe cluster (bitrate:packets): (1800000:5)
(congestion_controller.cc:384): Bitrate estimate state changed, BWE: 300000 bps.
(bitrate_allocator.cc:77): Current BWE 300000
(webrtcvideoengine2.cc:572): CreateChannel. Options: VideoOptions {}
(channel.cc:178): Created channel for video
(channel.cc:300): Create RTCP TransportChannel for video on video transport
(p2ptransportchannel.cc:383): Set ping most likely connection to 0
(p2ptransportchannel.cc:403): Set presume writable when fully relayed to 0
(p2ptransportchannel.cc:383): Set ping most likely connection to 0
(p2ptransportchannel.cc:403): Set presume writable when fully relayed to 0
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=down
(congestion_controller.cc:316): SignalNetworkState Down
(paced_sender.cc:274): PacedSender paused.
(congestion_controller.cc:384): Bitrate estimate state changed, BWE: 0 bps.
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=down
(webrtcsession.cc:818): Session:7347636247221083791 Old state:STATE_INIT New state:STATE_SENTOFFER
(congestion_controller.cc:316): SignalNetworkState Down
(conferencepeerconnectionchannel.cc:148): Signaling state changed: 1
(paced_sender.cc:274): PacedSender paused.
(channel.cc:1980): Setting local video description
(webrtcvideoengine2.cc:1027): SetRecvParameters: {codecs: [VideoCodec[120:H264:1920:1920:30], VideoCodec[100:VP8:1920:1920:30], VideoCodec[101:VP9:1920:1920:30], VideoCodec[116:red:1920:1920:30], VideoCodec[117:ulpfec:1920:1920:30], VideoCodec[96:rtx:1920:1920:30], VideoCodec[97:rtx:1920:1920:30], VideoCodec[98:rtx:1920:1920:30], VideoCodec[99:rtx:1920:1920:30]], 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.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}]}
(webrtcvideoengine2.cc:1036): Changing recv codecs from {VideoCodec[100:VP8:640:400:30], VideoCodec[101:VP9:640:400:30], VideoCodec[120:H264:1920:1080:60]} to {VideoCodec[120:H264:1920:1920:30], VideoCodec[100:VP8:1920:1920:30], VideoCodec[101:VP9:1920:1920:30]}
(webrtcvideoengine2.cc:1142): AddSendStream: {id:VideoTrack-62e86b6a-6045-4536-9f95-914e0f624349;ssrcs:[780220367,3434847234];ssrc_groups:{semantics:FID;ssrcs:[780220367,3434847234]};cname:/fW5hikIIzRBshqf;sync_label:MediaStream-8ba9ed27-dea7-4873-8c75-539f3047abb1}
(webrtcvideoengine2.cc:1168): SetLocalSsrc on all the receive streams because we added a send stream.
(channel.cc:1316): Add send stream ssrc: 780220367
(channel.cc:1947): Changing video state, send=0
(webrtcvideoengine2.cc:1102): SetVideoSend (ssrc= 780220367, enable = 1, options: VideoOptions {is_screencast : false, }, source = (source))
(basicportallocator.cc:250): Pruning turn ports disabled
(basicportallocator.cc:250): Pruning turn ports disabled
(conferencepeerconnectionchannel.cc:242): Set local sdp success.
(messagequeue.cc:520): Message took 65ms to dispatch. Posted from: woogeen::conference::ConferencePeerConnectionChannel::OnCreateSessionDescriptionSuccess@c:\cppsdk\src\talk\woogeen\sdk\conference\conferencepeerconnectionchannel.cc:233
(conferencepeerconnectionchannel.cc:247): Local SDP for stream: 754748542793095200
encoded payload length:2243
(conferencepeerconnectionchannel.cc:189): Ice gathering state changed: 1
(network.cc:848): Connect failed with 10051
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Udp
(port.cc:203): Jingle:Port[09457BC8::1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port created with network cost 50
(basicportallocator.cc:643): Adding allocated port for video
(basicportallocator.cc:663): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Added port to allocator
(basicportallocator.cc:680): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Gathered candidate: Cand[:2622663921:1:udp:2122260223:192.168.1.177:51550:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(basicportallocator.cc:707): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port ready.
(physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:480): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:723): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:786): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port completed gathering candidates.
(basicportallocator.cc:836): Signaling candidate because protocol was enabled: Cand[:2622663921:1:udp:2122260223:192.168.1.177:51550:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(conferencepeerconnectionchannel.cc:194): On ice candidate
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Udp
(port.cc:203): Jingle:Port[09457EF0::1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port created with network cost 50
(basicportallocator.cc:643): Adding allocated port for video
(basicportallocator.cc:663): Jingle:Port[09457EF0:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Added port to allocator
(basicportallocator.cc:680): Jingle:Port[09457EF0:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Gathered candidate: Cand[:2622663921:2:udp:2122260222:192.168.1.177:51551:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(basicportallocator.cc:707): Jingle:Port[09457EF0:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port ready.
(physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:480): Jingle:Port[09457EF0:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:723): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:786): Jingle:Port[09457EF0:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port completed gathering candidates.
(basicportallocator.cc:836): Signaling candidate because protocol was enabled: Cand[:2622663921:2:udp:2122260222:192.168.1.177:51551:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(conferencepeerconnectionchannel.cc:194): On ice candidate
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Relay
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Relay
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Tcp
(port.cc:203): Jingle:Port[09458218::1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port created with network cost 50
(basicportallocator.cc:643): Adding allocated port for video
(basicportallocator.cc:663): Jingle:Port[09458218:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Added port to allocator
(basicportallocator.cc:680): Jingle:Port[09458218:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Gathered candidate: Cand[:3536932865:1:tcp:1518280447:192.168.1.177:61820:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(basicportallocator.cc:707): Jingle:Port[09458218:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port ready.
(physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:480): Jingle:Port[09458218:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:723): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:786): Jingle:Port[09458218:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port completed gathering candidates.
(basicportallocator.cc:836): Signaling candidate because protocol was enabled: Cand[:3536932865:1:tcp:1518280447:192.168.1.177:61820:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(conferencepeerconnectionchannel.cc:194): On ice candidate
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=Tcp
(port.cc:203): Jingle:Port[10DA3D20::1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port created with network cost 50
(basicportallocator.cc:643): Adding allocated port for video
(basicportallocator.cc:663): Jingle:Port[10DA3D20:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Added port to allocator
(basicportallocator.cc:680): Jingle:Port[10DA3D20:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Gathered candidate: Cand[:3536932865:2:tcp:1518280446:192.168.1.177:61821:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(basicportallocator.cc:707): Jingle:Port[10DA3D20:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port ready.
(physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:480): Jingle:Port[10DA3D20:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:723): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:786): Jingle:Port[10DA3D20:video:2:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Port completed gathering candidates.
(basicportallocator.cc:836): Signaling candidate because protocol was enabled: Cand[:3536932865:2:tcp:1518280446:192.168.1.177:61821:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]
(conferencepeerconnectionchannel.cc:194): On ice candidate
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=SslTcp
(basicportallocator.cc:910): All candidates gathered for video:1:0
(p2ptransportchannel.cc:527): P2PTransportChannel: video, component 1 gathering complete
(basicportallocator.cc:1077): Jingle:Net[Realtek:192.168.1.177/32:Unknown]: Allocation Phase=SslTcp
(basicportallocator.cc:910): All candidates gathered for video:2:0
(p2ptransportchannel.cc:527): P2PTransportChannel: video, component 2 gathering complete
(conferencepeerconnectionchannel.cc:189): Ice gathering state changed: 2
Received Message type (Event)
(conferencepeerconnectionchannel.cc:637): On signaling message: answer
(webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
(conferencepeerconnectionchannel.cc:288): Post set remote desc
(webrtcsdp.cc:2606): Ignored line: c=IN IP4 192.168.1.26
(webrtcvideoengine2.cc:1384): Call stats: 32967344, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
(webrtcsession.cc:1079): BUNDLE already enabled for video on video.
(dtlstransportchannel.cc:301): Jingle:Channel[video|1|__]: DTLS setup complete.
(dtlstransportchannel.cc:301): Jingle:Channel[video|2|__]: DTLS setup complete.
(channel.cc:890): Channel enabled
(channel.cc:1947): Changing video state, send=0
(webrtcsession.cc:818): Session:7347636247221083791 Old state:STATE_SENTOFFER New state:STATE_INPROGRESS
(conferencepeerconnectionchannel.cc:148): Signaling state changed: 0
encoded payload length:269
encoded payload length:269
encoded payload length:285
encoded payload length:285
(channel.cc:2025): Setting remote video description
(channel.cc:1217): Enabling rtcp-mux for video by destroying RTCP transport channel for video
(opensslstreamadapter.cc:861): Cleanup
(webrtcvideoengine2.cc:824): SetSendParameters: {codecs: [VideoCodec[120:H264:1920:1920:30], VideoCodec[100:VP8:1920:1920:30], VideoCodec[101:VP9:1920:1920:30], VideoCodec[116:red:1920:1920:30], VideoCodec[117:ulpfec:1920:1920:30]], extensions: [], max_bandwidth_bps: -1, }
(webrtcvideoengine2.cc:833): Using codec: VideoCodec[120:H264:1920:1920:30]
(webrtcvideoengine2.cc:1876): RTX SSRCs configured but there's no configured RTX payload type. Ignoring.
(webrtcvideoengine2.cc:1890): RecreateWebRtcStream (send) because of SetCodec.
(video_send_stream.cc:496): VideoSendStream: {encoder_settings: {payload_name: H264, payload_type: 120, encoder: (VideoEncoder)}, rtp: {ssrcs: [780220367], rtcp_mode: RtcpMode::kCompound, max_packet_size: 1200, extensions: [], nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: -1}, rtx: {ssrcs: [], payload_type: -1}, c_name: /fW5hikIIzRBshqf}, pre_encode_callback: nullptr, post_encode_callback: nullptr, local_renderer: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
(video_send_stream.cc:830): Transmitting payload type without picture ID usingNACK+FEC is a waste of bandwidth since FEC packets also have to be retransmitted. Disabling FEC.
(video_send_stream.cc:752): (Re)configureVideoEncoder: {streams: [{width: 176, height: 144, max_framerate: 30, min_bitrate_bps:30000, target_bitrate_bps:600000, max_bitrate_bps:600000, max_qp: 56, temporal_layer_thresholds_bps: []}], content_type: kRealtimeVideo, encoder_specific_settings: (ptr), min_transmit_bitrate_bps: 0}
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=down
(congestion_controller.cc:316): SignalNetworkState Down
(paced_sender.cc:274): PacedSender paused.
(webrtcvideoengine2.cc:872): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
(webrtcvideoengine2.cc:1232): AddRecvStream: {id:v0;ssrcs:[55543,55555];ssrc_groups:;cname:o/i14u9pJrxRKAsu;sync_label:F7gWndyaSk}
(video_decoder.cc:32): Unable to create an H.264 decoder fallback. Decoding of this stream will be broken.
(video_receive_stream.cc:225): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 120, payload_name: H264, decoder_specific: { h264_extra_settings: nullptr}}, {decoder: (VideoDecoder), payload_type: 100, payload_name: VP8, decoder_specific: { h264_extra_settings: nullptr}}, {decoder: (VideoDecoder), payload_type: 101, payload_name: VP9, decoder_specific: { h264_extra_settings: nullptr}}], rtp: {remote_ssrc: 55543, local_ssrc: 780220367, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: off, nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: 98}, rtx: {}, extensions: [{uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: F7gWndyaSk, pre_decode_callback: nullptr, pre_render_callback: nullptr, target_delay_ms: 0}
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=down
(congestion_controller.cc:316): SignalNetworkState Down
(paced_sender.cc:274): PacedSender paused.
(channel.cc:1391): Add remote ssrc: 55543
(video_send_stream.cc:603): VideoSendStream::Stop
(channel.cc:1947): Changing video state, send=0
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=down
(congestion_controller.cc:316): SignalNetworkState Down
(paced_sender.cc:274): PacedSender paused.
(port.cc:312): Received STUN ping  id=f9906c02a92a28c13b984f3c from unknown address 192.168.1.26:38143
(port.cc:877): Jingle:Conn[10DB36C8:video:/ofqh2vo:1:0:local:udp:192.168.1.177:51550->36GoaoAe:1:1845494271:prflx:udp:192.168.1.26:38143|C--W|0|0|7926337543144881663|-]: Connection created
(p2ptransportchannel.cc:643): Adding connection from peer reflexive candidate: Cand[:3091674687:1:udp:1845494271:192.168.1.26:38143:prflx::0:/0fy:obasPuoiCe0KdGC3q5Ohor:0:0:0]
(port.cc:607): Jingle:Port[09457BC8:video:1:0:local:Net[Realtek:192.168.1.177/32:Unknown]]: Sent STUN ping response, to=192.168.1.26:38143, id=f9906c02a92a28c13b984f3c
(p2ptransportchannel.cc:1348): Jingle:Channel[video|1|__]: Transport channel state changed from 0 to 2
(transportcontroller.cc:622): video TransportChannel 1 state changed. Check if state is complete.
(p2ptransportchannel.cc:1034): Jingle:Channel[video|1|R_]: Have a pingable connection for the first time; starting to ping.
(webrtcsession.cc:1349): Changing IceConnectionState 0 => 1
(p2ptransportchannel.cc:1865): Selecting connection for triggered check: Conn[10DB36C8:video:/ofqh2vo:1:0:local:udp:192.168.1.177:51550->Ej1SIEOa:1:2013266431:local:udp:192.168.1.26:38143|CR-W|0|0|8646913483524161023|-]
(conferencepeerconnectionchannel.cc:172): Ice connection state changed: 1
(port.cc:1412): Jingle:Conn[10DB36C8:video:/ofqh2vo:1:0:local:udp:192.168.1.177:51550->Ej1SIEOa:1:2013266431:local:udp:192.168.1.26:38143|CR-W|0|0|8646913483524161023|-]: Sent STUN ping, id=644d39315653513838334547, use_candidate=1, nomination=0
(peerconnectionchannel.cc:193): Set remote sdp success.
(port.cc:1359): Jingle:Conn[10DB36C8:video:/ofqh2vo:1:0:local:udp:192.168.1.177:51550->Ej1SIEOa:1:2013266431:local:udp:192.168.1.26:38143|CR-I|0|0|8646913483524161023|-]: Received STUN ping response, id=644d39315653513838334547, code=0, rtt=1, pings_since_last_response=644d39315653513838334547
(dtlstransportchannel.cc:472): Jingle:Channel[video|1|__]: Packet received before DTLS started.
(dtlstransportchannel.cc:479): Jingle:Channel[video|1|__]: Caching DTLS ClientHello packet until DTLS is started.
(p2ptransportchannel.cc:232): Switching selected connection due to sorting
(p2ptransportchannel.cc:1319): Jingle:Channel[video|1|R_]: New selected connection: Conn[10DB36C8:video:/ofqh2vo:1:0:local:udp:192.168.1.177:51550->Ej1SIEOa:1:2013266431:local:udp:192.168.1.26:38143|CRWS|0|0|8646913483524161023|2250]
(opensslstreamadapter.cc:745): BeginSSL with peer.
(openssladapter.cc:851): SSL_accept:before/accept initialization
(openssladapter.cc:861): SSL_accept:error in SSLv3 read client hello B
(dtlstransportchannel.cc:586): Jingle:Channel[video|1|__]: DtlsTransportChannelWrapper: Started DTLS handshake
(srtpfilter.cc:459): SRTP reset to init state
(dtlstransportchannel.cc:593): Jingle:Channel[video|1|__]: Handling cached DTLS ClientHello packet.
(openssladapter.cc:851): SSL_accept:SSLv3 read client hello B
(openssladapter.cc:851): SSL_accept:SSLv3 write server hello A
(openssladapter.cc:851): SSL_accept:SSLv3 write certificate A
(openssladapter.cc:851): SSL_accept:SSLv3 write key exchange A
(openssladapter.cc:851): SSL_accept:SSLv3 write certificate request A
(openssladapter.cc:851): SSL_accept:SSLv3 write server done A
(openssladapter.cc:851): SSL_accept:SSLv3 flush data
(openssladapter.cc:861): SSL_accept:error in SSLv3 read client certificate A
(opensslstreamadapter.cc:1075): Accepted peer certificate.
(opensslstreamadapter.cc:1075): Accepted peer certificate.
(openssladapter.cc:851): SSL_accept:SSLv3 read client certificate A
(openssladapter.cc:851): SSL_accept:SSLv3 read client key exchange A
(openssladapter.cc:851): SSL_accept:SSLv3 read certificate verify A
(openssladapter.cc:851): SSL_accept:SSLv3 read finished A
(openssladapter.cc:851): SSL_accept:SSLv3 write session ticket A
(openssladapter.cc:851): SSL_accept:SSLv3 write change cipher spec A
(openssladapter.cc:851): SSL_accept:SSLv3 write finished A
(openssladapter.cc:851): SSL_accept:SSLv3 flush data
(dtlstransportchannel.cc:544): Jingle:Channel[video|1|__]: DTLS handshake complete.
(transportcontroller.cc:554): video TransportChannel 1 writability changed to 1.
(channel.cc:920): Channel writable (video) for the first time
(webrtcsession.cc:1386): Changing to ICE completed state because all transports are complete.
(channel.cc:928): Using Cand[:2622663921:1:udp:2122260223:192.168.1.177:51550:local::0:S9h4:0qqSiajKhYOaXz3ERSOBvOKy:3:50:0]->Cand[:1:1:udp:2013266431:192.168.1.26:38143:local::0:/0fy:obasPuoiCe0KdGC3q5Ohor:0:0:0]
(webrtcsession.cc:1349): Changing IceConnectionState 1 => 2
(channel.cc:987): Installing keys from DTLS-SRTP on video RTP
(conferencepeerconnectionchannel.cc:172): Ice connection state changed: 2
(webrtcsession.cc:1349): Changing IceConnectionState 2 => 3
(conferencepeerconnectionchannel.cc:172): Ice connection state changed: 3
(srtpfilter.cc:144): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1
(call.cc:716): UpdateAggregateNetworkState: aggregate_state=up
(congestion_controller.cc:316): SignalNetworkState Up
(paced_sender.cc:280): PacedSender resumed.
(congestion_controller.cc:384): Bitrate estimate state changed, BWE: 300000 bps.
(video_send_stream.cc:590): VideoSendStream::Start
(channel.cc:1947): Changing video state, send=1
(video_send_stream.cc:960): OnBitrateUpdated: Encoder state changed, target bitrate 300000 bps.
(bitrate_allocator.cc:171): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps
(video_send_stream.cc:701): webrtc::internal::VideoSendStream::EncoderProcess: Encoder started.
Received Message type (Event)
(conferenceclient.cc:631): OnStreamAdded: camera stream << 754748542793095200
Received Message type (Event)
(conferencepeerconnectionchannel.cc:637): On signaling message: ready
(conferencepeerconnectionchannel.cc:644): Received ready.
Received Message type (Event)
(network.cc:848): Connect failed with 10051
(video_send_stream.cc:719): webrtc::internal::VideoSendStream::EncoderProcess: Encoder timed out.
(bitrate_allocator.cc:171): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051
(network.cc:848): Connect failed with 10051