How to exchange SDP when relay media stream/track from remote
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raulf tang
Sep 5, 2016, 8:33:49 AM9/5/16
to discuss-webrtc
Hi All,
I try to use a server to relay media stream between clients, current just between two clients and multiple clients in future.
The architecture looks like this:
Client A <------------Peerconnection1-------------->Server<---------------------------Peerconnection2----------------->Client B
Server will createOffer/Answer/Candidate to Client A and Client B individually, and addStream to another peerconnection when receiving stream from one peerconnection。
If I let server to create local track, it woks well, Client A and Client B can see server's audio/video. But if I just relay the stream, it didn't work, no audio/video and server report DTLS timeout and connection 65 error(can't connect to 8.8.8.8).
Also I tried to relay offer/answer, server doesn't create offer/answer, and just take Client A's offer and send to Client B and receive Client B's answer then send to Client A. this will report sha-256 error, I understand the error, but don't know how to replace the "fingerprint:sha-256" with server's own. If server don't create offer/answer, there is a method to get the "fingerprint:sha-256"? How?
If Anybody experienced the process, please give me a help.
Thanks in advance.
raulf tang
Sep 5, 2016, 9:07:51 AM9/5/16
to discuss-webrtc
[000:000] [9987] (opensslidentity.cc:43): Making key pair
[000:001] [9987] (opensslidentity.cc:84): Returning key pair
[000:001] [9987] (opensslidentity.cc:91): Making certificate for WebRTC
[000:002] [9987] (opensslidentity.cc:139): Returning certificate
[004:858] [9987] (opensslidentity.cc:43): Making key pair
[004:859] [9987] (opensslidentity.cc:84): Returning key pair
[004:859] [9987] (opensslidentity.cc:91): Making certificate for WebRTC
[004:860] [9987] (opensslidentity.cc:139): Returning certificate
[004:860] [10499] (mediasession.cc:353): Duplicate id found. Reassigning from 101 to 127
[004:861] [10499] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:861] [10499] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:862] [10243] (systeminfo.cc:82): Available number of cores: 4
[004:863] [10243] (remote_bitrate_estimator_single_stream.cc:59): RemoteBitrateEstimatorSingleStream: Instantiating.
[004:863] [10243] (delay_based_bwe.cc:84): RemoteBitrateEstimatorAbsSendTime: Instantiating.
[004:863] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 300000 bps.
[004:863] [10243] (bitrate_allocator.cc:77): Current BWE 300000
[004:863] [10243] (webrtcvoiceengine.cc:1584): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:863] [10243] (webrtcvoiceengine.cc:594): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:863] [10243] (webrtcvoiceengine.cc:799): NetEq capacity is 50
[004:863] [10243] (webrtcvoiceengine.cc:807): NetEq fast mode? 0
[004:863] [10243] (webrtcvoiceengine.cc:835): Delay agnostic aec is enabled? 0
[004:863] [10243] (webrtcvoiceengine.cc:844): Extended filter aec is enabled? 0
[004:863] [10243] (webrtcvoiceengine.cc:853): Experimental ns is enabled? 0
[004:863] [10243] (webrtcvoiceengine.cc:859): Intelligibility Enhancer is enabled? 0
[004:863] [10243] (webrtcvoiceengine.cc:869): Level control: 0
[004:863] [10243] (webrtcvoiceengine.cc:1596): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:863] [10243] (channel.cc:192): Created channel for audio
[004:863] [9987] (channel.cc:313): Create RTCP TransportChannel for audio on audio transport
[004:863] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:863] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:864] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:864] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:864] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:864] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:864] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 0 bps.
[004:864] [10243] (webrtcvideoengine2.cc:558): CreateChannel. Options: VideoOptions {}
[004:864] [10243] (channel.cc:192): Created channel for video
[004:864] [9987] (channel.cc:313): Create RTCP TransportChannel for video on video transport
[004:864] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:864] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:864] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:864] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:865] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:865] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:865] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:865] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:865] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:865] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:865] [10499] (webrtcsession.cc:793): Session:6775006599987471599 Old state:STATE_INIT New state:STATE_SENTOFFER
[004:865] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:179 -[AudioVideoInterfaceImpl peerConnection:didChangeSignalingState:]): Signaling state changed: 1
[004:865] [10243] (channel.cc:1706): Setting local voice description
[004:866] [10243] (webrtcvoiceengine.cc:1454): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:32000:1], AudioCodec[104:ISAC:32000:56000:1], AudioCodec[9:G722:8000:0:1], AudioCodec[102:ILBC: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[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}
[004:866] [10243] (webrtcvoiceengine.cc:1606): Setting receive voice codecs.
[004:866] [10243] (webrtcvoiceengine.cc:1644): opus/48000/2 (111)
[004:866] [10243] (webrtcvoiceengine.cc:1644): ISAC/16000/1 (103)
[004:866] [10243] (webrtcvoiceengine.cc:1644): ISAC/32000/1 (104)
[004:866] [10243] (webrtcvoiceengine.cc:1644): G722/8000/1 (9)
[004:866] [10243] (webrtcvoiceengine.cc:1644): ILBC/8000/1 (102)
[004:866] [10243] (webrtcvoiceengine.cc:1644): PCMU/8000/1 (0)
[004:866] [10243] (webrtcvoiceengine.cc:1644): PCMA/8000/1 (8)
[004:866] [10243] (webrtcvoiceengine.cc:1644): CN/32000/1 (106)
[004:866] [10243] (webrtcvoiceengine.cc:1644): CN/16000/1 (105)
[004:866] [10243] (webrtcvoiceengine.cc:1644): CN/8000/1 (13)
[004:866] [10243] (webrtcvoiceengine.cc:1644): telephone-event/8000/1 (126)
[004:866] [10243] (channel.cc:1693): Changing voice state, recv=0 send=0
[004:866] [10243] (channel.cc:1989): Setting local video description
[004:866] [10243] (webrtcvideoengine2.cc:1012): SetRecvParameters: {codecs: [VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60], VideoCodec[116:red:1920:1080:60], VideoCodec[117:ulpfec:1920:1080:60], VideoCodec[96:rtx:1920:1080:60], VideoCodec[97:rtx:1920:1080:60], VideoCodec[98:rtx:1920:1080:60]], 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}]}
[004:867] [10243] (webrtcvideoengine2.cc:1021): Changing recv codecs from {VideoCodec[100:VP8:640:400:30], VideoCodec[101:VP9:640:400:30]} to {VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60]}
[004:867] [10243] (channel.cc:1956): Changing video state, send=0
[004:867] [9987] (basicportallocator.cc:250): Pruning turn ports disabled
[004:867] [9987] (basicportallocator.cc:250): Pruning turn ports disabled
[004:867] [9987] (basicportallocator.cc:250): Pruning turn ports disabled
[004:867] [9987] (basicportallocator.cc:250): Pruning turn ports disabled
[004:868] [9987] (network.cc:842): Connect failed with 65
[004:869] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Udp
[004:869] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:216 -[AudioVideoInterfaceImpl peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1
[004:869] [9987] (port.cc:201): Jingle:Port[0x104809000::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:869] [9987] (basicportallocator.cc:1152): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[004:869] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:869] [9987] (basicportallocator.cc:663): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:870] [9987] (basicportallocator.cc:680): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:3713327443:1:udp:2122260223:10.10.130.88:57893:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:870] [9987] (basicportallocator.cc:707): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:870] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:870] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:870] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[004:870] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:3713327443:1:udp:2122260223:10.10.130.88:57893:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:870] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Udp
[004:870] [9987] (port.cc:201): Jingle:Port[0x103829c00::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:870] [9987] (basicportallocator.cc:1152): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[004:870] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:871] [9987] (basicportallocator.cc:663): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:871] [9987] (basicportallocator.cc:680): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:3713327443:2:udp:2122260222:10.10.130.88:53751:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:871] [9987] (basicportallocator.cc:707): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:871] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:871] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:871] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[004:871] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:3713327443:2:udp:2122260222:10.10.130.88:53751:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:871] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Udp
[004:871] [9987] (port.cc:201): Jingle:Port[0x104800a00::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:871] [9987] (basicportallocator.cc:1152): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[004:871] [9987] (basicportallocator.cc:643): Adding allocated port for video
[004:871] [9987] (basicportallocator.cc:663): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:871] [9987] (basicportallocator.cc:680): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:3713327443:1:udp:2122260223:10.10.130.88:55876:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:871] [9987] (basicportallocator.cc:707): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:872] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:872] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:872] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[004:872] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:3713327443:1:udp:2122260223:10.10.130.88:55876:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:872] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Udp
[004:872] [9987] (port.cc:201): Jingle:Port[0x104801000::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:872] [9987] (basicportallocator.cc:1152): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[004:872] [9987] (basicportallocator.cc:643): Adding allocated port for video
[004:872] [9987] (basicportallocator.cc:663): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:872] [9987] (basicportallocator.cc:680): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:3713327443:2:udp:2122260222:10.10.130.88:57607:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:872] [9987] (basicportallocator.cc:707): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:872] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:872] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:872] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[004:873] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:3713327443:2:udp:2122260222:10.10.130.88:57607:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:875] [9987] (basicportallocator.cc:680): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:725915783:1:udp:1686052607:124.42.103.133:38561:stun:10.10.130.88:57893:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:875] [9987] (basicportallocator.cc:773): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:875] [9987] (basicportallocator.cc:680): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:725915783:2:udp:1686052606:124.42.103.133:56691:stun:10.10.130.88:53751:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:875] [9987] (basicportallocator.cc:773): Jingle:Port[0x103829c00:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:876] [9987] (basicportallocator.cc:680): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:725915783:1:udp:1686052607:124.42.103.133:57536:stun:10.10.130.88:55876:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:876] [9987] (basicportallocator.cc:773): Jingle:Port[0x104800a00:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:877] [9987] (basicportallocator.cc:680): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:725915783:2:udp:1686052606:124.42.103.133:56195:stun:10.10.130.88:57607:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:877] [9987] (basicportallocator.cc:773): Jingle:Port[0x104801000:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:922] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Relay
[004:922] [9987] (port.cc:201): Jingle:Port[0x103827200::1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:922] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:922] [9987] (basicportallocator.cc:663): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:922] [9987] (turnport.cc:303): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Trying to connect to TURN server via udp @ 120.92.22.201:3478
[004:922] [9987] (turnport.cc:1077): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=415a35492f67464c2b6e7576
[004:922] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Relay
[004:922] [9987] (port.cc:201): Jingle:Port[0x102027e00::1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:922] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:922] [9987] (basicportallocator.cc:663): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:939] [9987] (turnport.cc:303): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Trying to connect to TURN server via udp @ 120.92.22.201:3478
[004:939] [9987] (turnport.cc:1077): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=704d4a4d695a652b6b595a70
[004:940] [26119] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:940] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Relay
[004:940] [9987] (port.cc:201): Jingle:Port[0x10480b600::1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:940] [9987] (basicportallocator.cc:643): Adding allocated port for video
[004:940] [9987] (basicportallocator.cc:663): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:940] [9987] (turnport.cc:303): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Trying to connect to TURN server via udp @ 120.92.22.201:3478
[004:940] [9987] (turnport.cc:1077): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=666e4759562f5a7a37384937
[004:940] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Relay
[004:941] [9987] (port.cc:201): Jingle:Port[0x10382ae00::1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:941] [9987] (basicportallocator.cc:643): Adding allocated port for video
[004:941] [9987] (basicportallocator.cc:663): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:941] [9987] (turnport.cc:303): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Trying to connect to TURN server via udp @ 120.92.22.201:3478
[004:941] [26119] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:941] [9987] (turnport.cc:1077): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=66326f4e655a4b6e37693636
[004:941] [9987] (turnport.cc:1124): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Received TURN allocate error response, id=415a35492f67464c2b6e7576, code=401, rtt=19
[004:941] [9987] (turnport.cc:1077): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=776d6f7934777841687a5378
[004:942] [10243] (remote_bitrate_estimator_single_stream.cc:59): RemoteBitrateEstimatorSingleStream: Instantiating.
[004:942] [10243] (delay_based_bwe.cc:84): RemoteBitrateEstimatorAbsSendTime: Instantiating.
[004:942] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 300000 bps.
[004:942] [10243] (bitrate_allocator.cc:77): Current BWE 300000
[004:942] [10243] (webrtcvoiceengine.cc:1584): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:942] [10243] (webrtcvoiceengine.cc:594): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:942] [10243] (webrtcvoiceengine.cc:799): NetEq capacity is 50
[004:942] [10243] (webrtcvoiceengine.cc:807): NetEq fast mode? 0
[004:942] [10243] (webrtcvoiceengine.cc:835): Delay agnostic aec is enabled? 0
[004:942] [10243] (webrtcvoiceengine.cc:844): Extended filter aec is enabled? 0
[004:942] [10243] (webrtcvoiceengine.cc:853): Experimental ns is enabled? 0
[004:942] [10243] (webrtcvoiceengine.cc:859): Intelligibility Enhancer is enabled? 0
[004:943] [9987] (turnport.cc:1124): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Received TURN allocate error response, id=704d4a4d695a652b6b595a70, code=401, rtt=4
[004:967] [10243] (webrtcvoiceengine.cc:869): Level control: 0
[004:967] [10243] (webrtcvoiceengine.cc:1596): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:967] [10243] (channel.cc:192): Created channel for audio
[004:968] [9987] (turnport.cc:1077): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=506d716247484c336f75574d
[004:968] [9987] (turnport.cc:1083): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate requested successfully, id=776d6f7934777841687a5378, code=0, rtt=27
[004:968] [9987] (basicportallocator.cc:680): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:64523198:1:udp:41885439:120.92.22.201:55548:relay:124.42.103.133:38561:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:968] [9987] (basicportallocator.cc:707): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:968] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:968] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:968] [9987] (basicportallocator.cc:773): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:968] [9987] (turnport.cc:905): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Scheduled refresh in 540000ms.
[004:968] [9987] (turnport.cc:1124): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Received TURN allocate error response, id=666e4759562f5a7a37384937, code=401, rtt=28
[004:968] [9987] (turnport.cc:1077): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=66634450754a645766477253
[004:969] [9987] (turnport.cc:1124): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Received TURN allocate error response, id=66326f4e655a4b6e37693636, code=401, rtt=28
[004:969] [9987] (turnport.cc:1077): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate request sent, id=61344e4c4c3467664e6a6135
[004:969] [9987] (channel.cc:313): Create RTCP TransportChannel for audio on audio transport
[004:969] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:969] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:969] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:969] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:970] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:970] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:970] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 0 bps.
[004:970] [10243] (webrtcvideoengine2.cc:558): CreateChannel. Options: VideoOptions {}
[004:970] [10243] (channel.cc:192): Created channel for video
[004:970] [9987] (channel.cc:313): Create RTCP TransportChannel for video on video transport
[004:970] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:970] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:970] [9987] (p2ptransportchannel.cc:371): Set ping most likely connection to 0
[004:970] [9987] (p2ptransportchannel.cc:391): Set presume writable when fully relayed to 0
[004:970] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:970] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:970] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:970] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:970] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:970] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:971] [10499] (webrtcsession.cc:793): Session:1958704365979769431 Old state:STATE_INIT New state:STATE_RECEIVEDOFFER
[004:971] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:179 -[AudioVideoInterfaceImpl peerConnection:didChangeSignalingState:]): Signaling state changed: 3
[004:971] [10243] (channel.cc:1751): Setting remote voice description
[004:971] [10243] (webrtcvoiceengine.cc:1422): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:32000:1], AudioCodec[104:ISAC:32000:56000:1], AudioCodec[9:G722:8000:0:1], AudioCodec[102:ILBC: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[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], max_bandwidth_bps: -1, options: AudioOptions {}}
[004:971] [10243] (webrtcvoiceengine.cc:1781): Recreate all the receive streams because the send codec has changed.
[004:972] [10243] (webrtcvoiceengine.cc:2424): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[004:972] [10243] (webrtcvoiceengine.cc:1584): Setting voice channel options: AudioOptions {}
[004:972] [10243] (webrtcvoiceengine.cc:594): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:972] [10243] (webrtcvoiceengine.cc:799): NetEq capacity is 50
[004:972] [10243] (webrtcvoiceengine.cc:807): NetEq fast mode? 0
[004:972] [10243] (webrtcvoiceengine.cc:835): Delay agnostic aec is enabled? 0
[004:972] [10243] (webrtcvoiceengine.cc:844): Extended filter aec is enabled? 0
[004:972] [10243] (webrtcvoiceengine.cc:853): Experimental ns is enabled? 0
[004:972] [10243] (webrtcvoiceengine.cc:859): Intelligibility Enhancer is enabled? 0
[004:972] [10243] (webrtcvoiceengine.cc:869): Level control: 0
[004:972] [10243] (webrtcvoiceengine.cc:1596): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[004:972] [10243] (webrtcvoiceengine.cc:2105): AddRecvStream: {id:dd86ec49-9562-44d5-bd79-10c09eb013a6;ssrcs:[216853334];ssrc_groups:;cname:Zpqokf9ax1tcVSKz;sync_label:BlinkMS}
[004:973] [10243] (neteq_impl.cc:109): NetEq config: sample_rate_hz=16000, enable_audio_classifier=false, 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
[004:973] [9987] (turnport.cc:1083): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate requested successfully, id=506d716247484c336f75574d, code=0, rtt=6
[004:973] [9987] (basicportallocator.cc:680): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:64523198:2:udp:41885438:120.92.22.201:52033:relay:124.42.103.133:56691:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:973] [9987] (basicportallocator.cc:707): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:973] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:973] [10243] (audio_receive_stream.cc:89): AudioReceiveStream: {rtp: {remote_ssrc: 216853334, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: BlinkMS}
[004:973] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:974] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:974] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:974] [9987] (basicportallocator.cc:773): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:975] [10243] (webrtcvoiceengine.cc:2618): Stopping playout for channel #0
[004:975] [9987] (turnport.cc:905): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Scheduled refresh in 540000ms.
[004:976] [10243] (audio_device_impl.cc:1401): StopPlayout
[004:976] [9987] (turnport.cc:1083): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate requested successfully, id=66634450754a645766477253, code=0, rtt=8
[004:977] [10243] (audio_device_impl.cc:1404): output: 0
[004:977] [10243] (channel.cc:1400): Add remote ssrc: 216853334
[004:977] [10243] (channel.cc:1693): Changing voice state, recv=0 send=0
[004:977] [9987] (basicportallocator.cc:680): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:64523198:1:udp:41885439:120.92.22.201:53450:relay:124.42.103.133:57536:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:977] [9987] (basicportallocator.cc:707): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:977] [10243] (channel.cc:2034): Setting remote video description
[004:978] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:978] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:978] [9987] (basicportallocator.cc:773): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:978] [9987] (turnport.cc:905): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Scheduled refresh in 540000ms.
[004:978] [9987] (turnport.cc:1083): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN allocate requested successfully, id=61344e4c4c3467664e6a6135, code=0, rtt=9
[004:978] [9987] (basicportallocator.cc:680): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:64523198:2:udp:41885438:120.92.22.201:57447:relay:124.42.103.133:56195:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:978] [9987] (basicportallocator.cc:707): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:978] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:978] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:978] [9987] (basicportallocator.cc:773): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:978] [9987] (turnport.cc:905): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Scheduled refresh in 540000ms.
[004:978] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Tcp
[004:979] [9987] (port.cc:201): Jingle:Port[0x10334d590::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:979] [10243] (webrtcvideoengine2.cc:809): SetSendParameters: {codecs: [VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60], VideoCodec[107:H264:1920:1080:60], VideoCodec[116:red:1920:1080:60], VideoCodec[117:ulpfec:1920:1080:60], VideoCodec[96:rtx:1920:1080:60], VideoCodec[97:rtx:1920:1080:60], VideoCodec[99:rtx:1920:1080:60], VideoCodec[98:rtx:1920:1080:60]], 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}], max_bandwidth_bps: -1, }
[004:979] [10243] (webrtcvideoengine2.cc:818): Using codec: VideoCodec[100:VP8:1920:1080:60]
[004:979] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:979] [10243] (webrtcvideoengine2.cc:857): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
[004:979] [9987] (basicportallocator.cc:663): Jingle:Port[0x10334d590:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:979] [10243] (webrtcvideoengine2.cc:1217): AddRecvStream: {id:5c4b60f7-7913-42fe-9cd5-382b9f87e431;ssrcs:[1029649540,2978780783];ssrc_groups:{semantics:FID;ssrcs:[1029649540,2978780783]};cname:Zpqokf9ax1tcVSKz;sync_label:BlinkMS}
[004:979] [9987] (basicportallocator.cc:680): Jingle:Port[0x10334d590:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:2480348579:1:tcp:1518280447:10.10.130.88:53578:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:979] [9987] (basicportallocator.cc:707): Jingle:Port[0x10334d590:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:979] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:979] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x10334d590:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:979] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[004:979] [9987] (basicportallocator.cc:773): Jingle:Port[0x10334d590:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[004:979] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:2480348579:1:tcp:1518280447:10.10.130.88:53578:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:979] [10243] (video_receive_stream.cc:177): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 100, payload_name: VP8}, {decoder: (VideoDecoder), payload_type: 101, payload_name: VP9}], rtp: {remote_ssrc: 1029649540, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: 98}, rtx: {100 -> {ssrc: 2978780783, payload_type: 96}101 -> {ssrc: 2978780783, payload_type: 97}}, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: BlinkMS, pre_decode_callback: nullptr, pre_render_callback: nullptr, target_delay_ms: 0}
[004:980] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:980] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:980] [10243] (channel.cc:1400): Add remote ssrc: 1029649540
[004:980] [10243] (channel.cc:1956): Changing video state, send=0
[004:980] [10243] (webrtcvoiceengine.cc:2293): SetOutputVolume() to 1 for recv stream with ssrc 216853334
[004:981] [10243] (webrtcvideoengine2.cc:1327): SetSink: ssrc:1029649540 (ptr)
[004:981] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:186 -[AudioVideoInterfaceImpl peerConnection:didAddStream:]): Received 1 video tracks and 1 audio tracks
[004:981] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:206 -[AudioVideoInterfaceImpl peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.
[004:983] [10499] (webrtcsession.cc:628): Local and Remote descriptions must be applied to get SSL Role of the session.
[004:983] [10499] (webrtcsession.cc:628): Local and Remote descriptions must be applied to get SSL Role of the session.
[004:984] [10499] (webrtcsession.cc:1095): ProcessIceMessage: Not ready to use candidate.
[004:984] [10499] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:985] [10499] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[004:985] [10499] (webrtcsession.cc:1045): BUNDLE already enabled for audio on audio.
[004:985] [9987] (channel.cc:313): Create RTCP TransportChannel for video on audio transport
[004:986] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:986] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:986] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:986] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:986] [10499] (webrtcsession.cc:1054): Enabled BUNDLE for video on audio.
[004:986] [9987] (dtlstransportchannel.cc:301): Jingle:Channel[audio|1|__]: DTLS setup complete.
[004:986] [9987] (dtlstransportchannel.cc:301): Jingle:Channel[audio|2|__]: DTLS setup complete.
[004:986] [10243] (channel.cc:898): Channel enabled
[004:986] [10243] (channel.cc:1693): Changing voice state, recv=0 send=0
[004:986] [10243] (channel.cc:898): Channel enabled
[004:986] [10243] (channel.cc:1956): Changing video state, send=0
[004:986] [10499] (webrtcsession.cc:793): Session:1958704365979769431 Old state:STATE_RECEIVEDOFFER New state:STATE_INPROGRESS
[004:986] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:179 -[AudioVideoInterfaceImpl peerConnection:didChangeSignalingState:]): Signaling state changed: 0
[004:986] [10243] (channel.cc:1706): Setting local voice description
[004:987] [9987] (channel.cc:1225): Enabling rtcp-mux for audio by destroying RTCP transport channel for audio
[004:987] [10243] (webrtcvoiceengine.cc:1454): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:32000:1], AudioCodec[104:ISAC:32000:56000:1], AudioCodec[9:G722:8000:0:1], AudioCodec[102:ILBC: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[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}
[004:987] [10243] (webrtcvoiceengine.cc:1606): Setting receive voice codecs.
[004:987] [10243] (webrtcvoiceengine.cc:1644): opus/48000/2 (111)
[004:987] [10243] (webrtcvoiceengine.cc:1644): ISAC/16000/1 (103)
[004:987] [10243] (webrtcvoiceengine.cc:1644): ISAC/32000/1 (104)
[004:987] [10243] (webrtcvoiceengine.cc:1644): G722/8000/1 (9)
[004:987] [10243] (webrtcvoiceengine.cc:1644): ILBC/8000/1 (102)
[004:987] [10243] (webrtcvoiceengine.cc:1644): PCMU/8000/1 (0)
[004:987] [10243] (webrtcvoiceengine.cc:1644): PCMA/8000/1 (8)
[004:987] [10243] (webrtcvoiceengine.cc:1644): CN/32000/1 (106)
[004:987] [10243] (webrtcvoiceengine.cc:1644): CN/16000/1 (105)
[004:987] [10243] (webrtcvoiceengine.cc:1644): CN/8000/1 (13)
[004:987] [10243] (webrtcvoiceengine.cc:1644): telephone-event/8000/1 (126)
[004:987] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:987] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:988] [10243] (audio_receive_stream.cc:146): ~AudioReceiveStream: {rtp: {remote_ssrc: 216853334, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: BlinkMS}
[004:988] [10243] (audio_receive_stream.cc:89): AudioReceiveStream: {rtp: {remote_ssrc: 216853334, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: BlinkMS}
[004:988] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[004:988] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[004:988] [10243] (webrtcvoiceengine.cc:2612): Starting playout for channel #0
[004:988] [10243] (audio_device_impl.cc:1415): Playing
[004:988] [10243] (audio_device_impl.cc:1337): InitPlayout
[004:988] [10243] (audio_device_buffer.cc:94): InitPlayout
[004:989] [10243] (audio_device_buffer.cc:120): SetPlayoutSampleRate(48000)
[004:990] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Tcp
[004:990] [9987] (port.cc:201): Jingle:Port[0x103214fe0::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[004:990] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[004:990] [9987] (basicportallocator.cc:663): Jingle:Port[0x103214fe0:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[004:990] [9987] (basicportallocator.cc:680): Jingle:Port[0x103214fe0:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:2480348579:2:tcp:1518280446:10.10.130.88:53579:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[004:990] [9987] (basicportallocator.cc:707): Jingle:Port[0x103214fe0:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[004:990] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[004:990] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x103214fe0:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[004:993] [10243] (audio_device_impl.cc:1341): output: 0
[005:051] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[005:052] [10243] (audio_device_impl.cc:1387): StartPlayout
[005:076] [9987] (basicportallocator.cc:773): Jingle:Port[0x103214fe0:audio:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[005:076] [10243] (audio_device_impl.cc:1390): output: 0
[005:078] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:2480348579:2:tcp:1518280446:10.10.130.88:53579:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[005:078] [10243] (channel.cc:1693): Changing voice state, recv=1 send=0
[005:079] [9987] (messagequeue.cc:516): Message took 89ms to dispatch. Posted from: OnMessage@../../webrtc/p2p/client/basicportallocator.cc:1091
[005:079] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:080] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Tcp
[005:080] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:080] [9987] (port.cc:201): Jingle:Port[0x103329000::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[005:080] [10243] (channel.cc:1989): Setting local video description
[005:082] [9987] (basicportallocator.cc:643): Adding allocated port for video
[005:082] [9987] (basicportallocator.cc:663): Jingle:Port[0x103329000:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[005:083] [9987] (basicportallocator.cc:680): Jingle:Port[0x103329000:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:2480348579:1:tcp:1518280447:10.10.130.88:53580:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[005:083] [9987] (basicportallocator.cc:707): Jingle:Port[0x103329000:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[005:083] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[005:084] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x103329000:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[005:084] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[005:084] [9987] (basicportallocator.cc:773): Jingle:Port[0x103329000:video:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[005:084] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:2480348579:1:tcp:1518280447:10.10.130.88:53580:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[005:084] [9987] (channel.cc:1225): Enabling rtcp-mux for video by destroying RTCP transport channel for audio
[005:084] [9987] (opensslstreamadapter.cc:874): Cleanup
[005:085] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Tcp
[005:085] [9987] (port.cc:201): Jingle:Port[0x101e2b6d0::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[005:085] [10243] (webrtcvideoengine2.cc:1012): SetRecvParameters: {codecs: [VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60], VideoCodec[116:red:1920:1080:60], VideoCodec[117:ulpfec:1920:1080:60], VideoCodec[96:rtx:1920:1080:60], VideoCodec[97:rtx:1920:1080:60], VideoCodec[98:rtx:1920:1080:60]], 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}]}
[005:085] [10243] (webrtcvideoengine2.cc:1021): Changing recv codecs from {VideoCodec[100:VP8:640:400:30], VideoCodec[101:VP9:640:400:30]} to {VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60]}
[005:086] [9987] (basicportallocator.cc:643): Adding allocated port for video
[005:086] [9987] (basicportallocator.cc:663): Jingle:Port[0x101e2b6d0:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[005:086] [10243] (webrtcvideoengine2.cc:2418): RecreateWebRtcStream (recv) because of SetRecvParameters
[005:086] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:086] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:086] [9987] (basicportallocator.cc:680): Jingle:Port[0x101e2b6d0:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:2480348579:2:tcp:1518280446:10.10.130.88:53581:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[005:086] [9987] (basicportallocator.cc:707): Jingle:Port[0x101e2b6d0:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[005:086] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[005:086] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x101e2b6d0:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[005:086] [10243] (video_receive_stream.cc:208): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 100, payload_name: VP8}, {decoder: (VideoDecoder), payload_type: 101, payload_name: VP9}], rtp: {remote_ssrc: 1029649540, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: 98}, rtx: {100 -> {ssrc: 2978780783, payload_type: 96}101 -> {ssrc: 2978780783, payload_type: 97}}, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: BlinkMS, pre_decode_callback: nullptr, pre_render_callback: nullptr, target_delay_ms: 0}
[005:086] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[005:087] [9987] (basicportallocator.cc:773): Jingle:Port[0x101e2b6d0:video:2:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[005:087] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:2480348579:2:tcp:1518280446:10.10.130.88:53581:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]
[005:087] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=SslTcp
[005:087] [9987] (basicportallocator.cc:897): All candidates gathered for audio:1:0
[005:087] [9987] (p2ptransportchannel.cc:509): P2PTransportChannel: audio, component 1 gathering complete
[005:088] [10243] (video_receive_stream.cc:177): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 100, payload_name: VP8}, {decoder: (VideoDecoder), payload_type: 101, payload_name: VP9}], rtp: {remote_ssrc: 1029649540, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: 98}, rtx: {100 -> {ssrc: 2978780783, payload_type: 96}101 -> {ssrc: 2978780783, payload_type: 97}}, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {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: BlinkMS, pre_decode_callback: nullptr, pre_render_callback: nullptr, target_delay_ms: 0}
[005:088] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:088] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:088] [10243] (channel.cc:1956): Changing video state, send=0
[005:089] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:089] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:089] [10499] (webrtcsession.cc:1315): Changing IceConnectionState 0 => 1
[005:089] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:211 -[AudioVideoInterfaceImpl peerConnection:didChangeIceConnectionState:]): ICE state changed: 1
[005:089] [9987] (basicportallocator.cc:250): Pruning turn ports disabled
[005:089] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=Udp
[005:089] [9987] (port.cc:201): Jingle:Port[0x104882e00::1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port created with network cost 50
[005:090] [9987] (basicportallocator.cc:1152): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[005:090] [9987] (basicportallocator.cc:643): Adding allocated port for audio
[005:090] [10499] (messagequeue.cc:516): Message took 106ms to dispatch. Posted from: PostCreateSessionDescriptionSucceeded@../../webrtc/api/webrtcsessiondescriptionfactory.cc:463
[005:090] [9987] (basicportallocator.cc:663): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Added port to allocator
[005:090] [9987] (basicportallocator.cc:680): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:3713327443:1:udp:2122260223:10.10.130.88:50550:local::0:4BvE:aSlkU44anHI1Cu/Vs9j2kZ6/:1:50:0]
[005:090] [9987] (basicportallocator.cc:707): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port ready.
[005:090] [9987] (physicalsocketserver.cc:595): Socket::OPT_DSCP not supported.
[005:090] [9987] (p2ptransportchannel.cc:462): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: SetOption(5, 0) failed: 0
[005:090] [9987] (port.cc:851): Jingle:Conn[0x104883400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->Ed+EVk3g:1:2122260223:local:udp:10.10.140.234:59655|C--W|9115038255631187454|-]: Connection created
[005:090] [9987] (p2ptransportchannel.cc:809): Jingle:Channel[audio|1|__]: Created connection with origin=2, (1 total)
[005:090] [9987] (p2ptransportchannel.cc:220): Switching selected connection due to sorting
[005:090] [9987] (p2ptransportchannel.cc:1293): Jingle:Channel[audio|1|__]: New selected connection: Conn[0x104883400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->Ed+EVk3g:1:2122260223:local:udp:10.10.140.234:59655|C--W|9115038255631187454|-]
[005:090] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|__]: Transport channel state changed from 0 to 2
[005:090] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:090] [10243] (call.cc:639): Transport audio is disconnected
[005:090] [10243] (call.cc:639): Transport audio is disconnected
[005:090] [9987] (p2ptransportchannel.cc:1009): Jingle:Channel[audio|1|__]: Have a pingable connection for the first time; starting to ping.
[005:090] [9987] (basicportallocator.cc:722): Not yet signaling candidate because protocol is not yet enabled.
[005:091] [9987] (basicportallocator.cc:823): Signaling candidate because protocol was enabled: Cand[:3713327443:1:udp:2122260223:10.10.130.88:50550:local::0:4BvE:aSlkU44anHI1Cu/Vs9j2kZ6/:1:50:0]
[005:091] [9987] (port.cc:1373): Jingle:Conn[0x104883400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->Ed+EVk3g:1:2122260223:local:udp:10.10.140.234:59655|C--W|9115038255631187454|-]: Sent STUN ping, id=51304952765674756f776a6b, use_candidate=0
[005:091] [10499] (webrtcsession.cc:1521): Candidate has an unknown component: Cand[:3530593514:2:udp:2122260222:10.10.140.234:64961:local::0:ejX2::1:10:0] for content: audio
[005:094] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:216 -[AudioVideoInterfaceImpl peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1
[005:095] [9987] (basicportallocator.cc:680): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Gathered candidate: Cand[:725915783:1:udp:1686052607:124.42.103.133:36338:stun:10.10.130.88:50550:4BvE:aSlkU44anHI1Cu/Vs9j2kZ6/:1:50:0]
[005:095] [9987] (basicportallocator.cc:773): Jingle:Port[0x104882e00:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Port completed gathering candidates.
[005:095] [9987] (port.cc:851): Jingle:Conn[0x104883c00:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->8Ol9KTvY:1:1686052607:stun:udp:124.42.103.133:54147|C--W|7241540810645061118|-]: Connection created
[005:095] [9987] (p2ptransportchannel.cc:809): Jingle:Channel[audio|1|__]: Created connection with origin=2, (2 total)
[005:095] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|__]: Transport channel state changed from 2 to 1
[005:095] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:096] [9987] (port.cc:1319): Jingle:Conn[0x104883400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->Ed+EVk3g:1:2122260223:local:udp:10.10.140.234:59655|CRWS|9115038255631187454|2251]: Received STUN ping response, id=51304952765674756f776a6b, code=0, rtt=5, use_candidate=0, pings_since_last_response=
[005:096] [9987] (opensslstreamadapter.cc:756): BeginSSL: with peer
[005:096] [10499] (webrtcsession.cc:1521): Candidate has an unknown component: Cand[:1396216414:2:udp:1686052606:124.42.103.133:56133:stun:10.10.140.234:64961:ejX2::1:10:0] for content: audio
[005:096] [9987] (openssladapter.cc:851): SSL_connect:before connect initialization
[005:096] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write client hello A
[005:096] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 flush data
[005:096] [9987] (openssladapter.cc:861): SSL_connect:error in DTLS1 read hello verify request A
[005:096] [9987] (dtlstransportchannel.cc:586): Jingle:Channel[audio|1|__]: DtlsTransportChannelWrapper: Started DTLS handshake
[005:096] [9987] (srtpfilter.cc:439): SRTP reset to init state
[005:096] [9987] (srtpfilter.cc:439): SRTP reset to init state
[005:096] [9987] (basicportallocator.cc:897): All candidates gathered for audio:1:0
[005:096] [9987] (p2ptransportchannel.cc:509): P2PTransportChannel: audio, component 1 gathering complete
[005:097] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:216 -[AudioVideoInterfaceImpl peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2
[005:097] [9987] (port.cc:851): Jingle:Conn[0x10202b400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->pKjpK0Zq:1:41885439:relay:udp:120.92.22.201:64935|C--W|179896594928123390|-]: Connection created
[005:097] [9987] (p2ptransportchannel.cc:809): Jingle:Channel[audio|1|RW]: Created connection with origin=2, (3 total)
[005:098] [10499] (webrtcsession.cc:1521): Candidate has an unknown component: Cand[:64523198:2:udp:41885438:120.92.22.201:53721:relay:124.42.103.133:56133:ejX2::1:10:0] for content: audio
[005:098] [10499] (webrtcsession.cc:1521): Candidate has an unknown component: Cand[:2633197594:2:tcp:1518280446:10.10.140.234:50149:local::0:ejX2::1:10:0] for content: audio
[005:122] [9987] (port.cc:310): Received STUN ping id=506c6b464e31492b30473769 from unknown address 10.10.140.187:64832
[005:122] [9987] (port.cc:851): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->HN7fOTc4:1:1853824767:prflx:udp:10.10.140.187:64832|C--W|7962116751024340479|-]: Connection created
[005:122] [9987] (p2ptransportchannel.cc:625): Adding connection from peer reflexive candidate: Cand[:1137033019:1:udp:1853824767:10.10.140.187:64832:prflx::0:dUyp::1:10:0]
[005:122] [9987] (port.cc:605): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Sent STUN ping response, to=10.10.140.187:64832, id=506c6b464e31492b30473769
[005:166] [9987] (p2ptransportchannel.cc:220): Switching selected connection due to sorting
[005:166] [9987] (p2ptransportchannel.cc:1293): Jingle:Channel[audio|1|__]: New selected connection: Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->HN7fOTc4:1:1853824767:prflx:udp:10.10.140.187:64832|CR-W|7962116751024340479|-]
[005:166] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|__]: Transport channel state changed from 0 to 2
[005:166] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:166] [10243] (call.cc:639): Transport audio is disconnected
[005:166] [9987] (dtlstransportchannel.cc:474): Jingle:Channel[audio|1|__]: Packet received before we know if we are doing DTLS or not.
[005:166] [9987] (dtlstransportchannel.cc:479): Jingle:Channel[audio|1|__]: Caching DTLS ClientHello packet until DTLS is started.
[005:166] [36355] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[005:168] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server hello A
[005:168] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server certificate A
[005:168] [36355] (webrtcsdp.cc:2606): Ignored line: c=IN IP4 0.0.0.0
[005:169] [9987] (opensslstreamadapter.cc:1088): Accepted peer certificate.
[005:169] [9987] (opensslstreamadapter.cc:1088): Accepted peer certificate.
[005:170] [9987] (openssladapter.cc:851): SSL_connect:unknown state
[005:172] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server key exchange A
[005:172] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server certificate request A
[005:172] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server done A
[005:172] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write client certificate A
[005:173] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write client key exchange A
[005:174] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write certificate verify A
[005:174] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write change cipher spec
[005:174] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 write finished A
[005:174] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 flush data
[005:204] [9987] (openssladapter.cc:861): SSL_connect:error in SSLv3 read server session ticket A
[005:204] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=SslTcp
[005:204] [9987] (basicportallocator.cc:897): All candidates gathered for audio:2:0
[005:204] [9987] (p2ptransportchannel.cc:509): P2PTransportChannel: audio, component 2 gathering complete
[005:205] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=SslTcp
[005:205] [9987] (basicportallocator.cc:897): All candidates gathered for video:1:0
[005:205] [9987] (p2ptransportchannel.cc:509): P2PTransportChannel: video, component 1 gathering complete
[005:205] [9987] (basicportallocator.cc:1061): Jingle:Net[en5:10.10.130.0/23:Unknown]: Allocation Phase=SslTcp
[005:205] [9987] (basicportallocator.cc:897): All candidates gathered for video:2:0
[005:205] [9987] (p2ptransportchannel.cc:509): P2PTransportChannel: video, component 2 gathering complete
[005:206] [9987] (port.cc:976): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->HN7fOTc4:1:1853824767:prflx:udp:10.10.140.187:64832|CR-W|7962116751024340479|-]: Received STUN ping, id=58786b594f69506232534f31
[005:207] [9987] (port.cc:605): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Sent STUN ping response, to=10.10.140.187:64832, id=58786b594f69506232534f31
[005:207] [9987] (openssladapter.cc:861): SSL_connect:error in SSLv3 read server session ticket A
[005:207] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read server session ticket A
[005:207] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read change cipher spec
[005:207] [9987] (openssladapter.cc:851): SSL_connect:SSLv3 read finished A
[005:207] [9987] (dtlstransportchannel.cc:544): Jingle:Channel[audio|1|__]: DTLS handshake complete.
[005:207] [9987] (transportcontroller.cc:554): audio TransportChannel 1 writability changed to 1.
[005:207] [9987] (channel.cc:928): Channel writable (audio) for the first time
[005:207] [9987] (channel.cc:936): Using Cand[:3713327443:1:udp:2122260223:10.10.130.88:50550:local::0:4BvE:aSlkU44anHI1Cu/Vs9j2kZ6/:1:50:0]->Cand[:3530593514:1:udp:2122260223:10.10.140.234:59655:local::0:ejX2:+mbZXyIJmJfDBT/75Z7cbk8w:1:10:0]
[005:208] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=up
[005:209] [9987] (channel.cc:995): Installing keys from DTLS-SRTP on audio RTP
[005:209] [10243] (congestion_controller.cc:291): SignalNetworkState Up
[005:210] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 300000 bps.
[005:210] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=up
[005:210] [10243] (congestion_controller.cc:291): SignalNetworkState Up
[005:212] [9987] (srtpfilter.cc:150): SRTP activated with negotiated parameters: send cipher_suite 2 recv cipher_suite 2
[005:212] [9987] (channel.cc:928): Channel writable (video) for the first time
[005:212] [10243] (channel.cc:1693): Changing voice state, recv=1 send=0
[005:212] [9987] (channel.cc:936): Using Cand[:3713327443:1:udp:2122260223:10.10.130.88:50550:local::0:4BvE:aSlkU44anHI1Cu/Vs9j2kZ6/:1:50:0]->Cand[:3530593514:1:udp:2122260223:10.10.140.234:59655:local::0:ejX2:+mbZXyIJmJfDBT/75Z7cbk8w:1:10:0]
[005:212] [9987] (channel.cc:995): Installing keys from DTLS-SRTP on video RTP
[005:212] [9987] (srtpfilter.cc:150): SRTP activated with negotiated parameters: send cipher_suite 2 recv cipher_suite 2
[005:212] [10243] (channel.cc:1956): Changing video state, send=0
[005:213] [10499] (webrtcsession.cc:1045): BUNDLE already enabled for audio on audio.
[005:213] [9987] (srtpfilter.cc:439): SRTP reset to init state
[005:213] [9987] (channel.cc:313): Create RTCP TransportChannel for video on audio transport
[005:213] [9987] (turnport.cc:1252): Jingle:Port[0x10382ae00:video:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN refresh request sent, id=684e6c4435627768526b7930
[005:214] [9987] (turnport.cc:1252): Jingle:Port[0x10480b600:video:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN refresh request sent, id=3275414c627142627a627255
[005:214] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:214] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:214] [10499] (webrtcsession.cc:1054): Enabled BUNDLE for video on audio.
[005:214] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:214] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:214] [9987] (dtlstransportchannel.cc:301): Jingle:Channel[audio|1|__]: DTLS setup complete.
[005:215] [9987] (dtlstransportchannel.cc:301): Jingle:Channel[audio|2|__]: DTLS setup complete.
[005:215] [9987] (p2ptransportchannel.cc:1009): Jingle:Channel[audio|1|R_]: Have a pingable connection for the first time; starting to ping.
[005:215] [9987] (p2ptransportchannel.cc:1808): Selecting connection for triggered check: Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->HN7fOTc4:1:1853824767:prflx:udp:10.10.140.187:64832|CR-W|7962116751024340479|-]
[005:215] [9987] (port.cc:1373): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->HN7fOTc4:1:1853824767:prflx:udp:10.10.140.187:64832|CR-W|7962116751024340479|-]: Sent STUN ping, id=304b3578764a6b5671574559, use_candidate=1
[005:215] [10243] (channel.cc:898): Channel enabled
[005:215] [10243] (channel.cc:1693): Changing voice state, recv=1 send=0
[005:215] [10243] (channel.cc:898): Channel enabled
[005:215] [10243] (channel.cc:1956): Changing video state, send=0
[005:215] [10499] (webrtcsession.cc:793): Session:6775006599987471599 Old state:STATE_SENTOFFER New state:STATE_INPROGRESS
[005:215] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:179 -[AudioVideoInterfaceImpl peerConnection:didChangeSignalingState:]): Signaling state changed: 0
[005:215] [10243] (channel.cc:1751): Setting remote voice description
[005:215] [9987] (channel.cc:1225): Enabling rtcp-mux for audio by destroying RTCP transport channel for audio
[005:215] [10243] (webrtcvoiceengine.cc:1422): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:32000:1], AudioCodec[9:G722:8000:0:1], AudioCodec[102:ILBC: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[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], max_bandwidth_bps: -1, options: AudioOptions {}}
[005:216] [10243] (webrtcvoiceengine.cc:1781): Recreate all the receive streams because the send codec has changed.
[005:216] [10243] (webrtcvoiceengine.cc:2424): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[005:216] [10243] (webrtcvoiceengine.cc:1584): Setting voice channel options: AudioOptions {}
[005:216] [10243] (webrtcvoiceengine.cc:594): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[005:216] [10243] (webrtcvoiceengine.cc:799): NetEq capacity is 50
[005:216] [10243] (webrtcvoiceengine.cc:807): NetEq fast mode? 0
[005:216] [10243] (webrtcvoiceengine.cc:835): Delay agnostic aec is enabled? 0
[005:216] [10243] (webrtcvoiceengine.cc:844): Extended filter aec is enabled? 0
[005:216] [10243] (webrtcvoiceengine.cc:853): Experimental ns is enabled? 0
[005:216] [10243] (webrtcvoiceengine.cc:859): Intelligibility Enhancer is enabled? 0
[005:216] [10243] (webrtcvoiceengine.cc:869): Level control: 0
[005:216] [10243] (webrtcvoiceengine.cc:1596): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
[005:216] [10243] (webrtcvoiceengine.cc:2105): AddRecvStream: {id:5e5f049a-ab36-4503-950a-96bd2fee5c1a;ssrcs:[3493221509];ssrc_groups:;cname:3MeBV7IyThHcsgCj;sync_label:BlinkMS}
[005:216] [10243] (neteq_impl.cc:109): NetEq config: sample_rate_hz=16000, enable_audio_classifier=false, 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
[005:216] [10243] (audio_receive_stream.cc:89): AudioReceiveStream: {rtp: {remote_ssrc: 3493221509, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), voe_channel_id: 1, sync_group: BlinkMS}
[005:217] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:217] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:217] [10243] (webrtcvoiceengine.cc:2612): Starting playout for channel #1
[005:251] [10243] (audio_device_impl.cc:1415): Playing
[005:251] [10243] (channel.cc:1400): Add remote ssrc: 3493221509
[005:251] [10243] (channel.cc:1693): Changing voice state, recv=1 send=0
[005:251] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:251] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:252] [10243] (channel.cc:2034): Setting remote video description
[005:252] [9987] (channel.cc:1225): Enabling rtcp-mux for video by destroying RTCP transport channel for audio
[005:252] [9987] (opensslstreamadapter.cc:874): Cleanup
[005:252] [9987] (turnport.cc:1252): Jingle:Port[0x102027e00:audio:2:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN refresh request sent, id=445058616e7a462b56705841
[005:252] [10243] (webrtcvideoengine2.cc:809): SetSendParameters: {codecs: [VideoCodec[100:VP8:1920:1080:60], VideoCodec[101:VP9:1920:1080:60], VideoCodec[116:red:1920:1080:60], VideoCodec[117:ulpfec:1920:1080:60], VideoCodec[96:rtx:1920:1080:60], VideoCodec[97:rtx:1920:1080:60], VideoCodec[98:rtx:1920:1080:60]], 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}], max_bandwidth_bps: -1, }
[005:253] [10243] (webrtcvideoengine2.cc:818): Using codec: VideoCodec[100:VP8:1920:1080:60]
[005:253] [10243] (webrtcvideoengine2.cc:857): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
[005:253] [10243] (webrtcvideoengine2.cc:1217): AddRecvStream: {id:e867afe7-2c01-4545-8986-7e87c76006f7;ssrcs:[432006656,2152666235];ssrc_groups:{semantics:FID;ssrcs:[432006656,2152666235]};cname:3MeBV7IyThHcsgCj;sync_label:BlinkMS}
[005:253] [10243] (video_receive_stream.cc:177): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 100, payload_name: VP8}, {decoder: (VideoDecoder), payload_type: 101, payload_name: VP9}], rtp: {remote_ssrc: 432006656, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, fec: {ulpfec_payload_type: 117, red_payload_type: 116, red_rtx_payload_type: 98}, rtx: {100 -> {ssrc: 2152666235, payload_type: 96}101 -> {ssrc: 2152666235, payload_type: 97}}, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {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: BlinkMS, pre_decode_callback: nullptr, pre_render_callback: nullptr, target_delay_ms: 0}
[005:253] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:253] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:254] [10243] (channel.cc:1400): Add remote ssrc: 432006656
[005:254] [10243] (channel.cc:1956): Changing video state, send=0
[005:254] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=down
[005:254] [10499] (webrtcsession.cc:1315): Changing IceConnectionState 0 => 1
[005:254] [10243] (congestion_controller.cc:291): SignalNetworkState Down
[005:254] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:211 -[AudioVideoInterfaceImpl peerConnection:didChangeIceConnectionState:]): ICE state changed: 1
[005:254] [10243] (webrtcvoiceengine.cc:2293): SetOutputVolume() to 1 for recv stream with ssrc 3493221509
[005:254] [10243] (webrtcvideoengine2.cc:1327): SetSink: ssrc:432006656 (ptr)
[005:254] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:186 -[AudioVideoInterfaceImpl peerConnection:didAddStream:]): Received 1 video tracks and 1 audio tracks
[005:255] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:206 -[AudioVideoInterfaceImpl peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.
[005:255] [10499] (messagequeue.cc:516): Message took 84ms to dispatch. Posted from: SetRemoteDescription@../../webrtc/api/peerconnectionproxy.h:64
[005:255] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:216 -[AudioVideoInterfaceImpl peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2
[005:255] [10499] (webrtcsession.cc:1347): Changing to ICE connected state because all transports are writable.
[005:255] [10499] (webrtcsession.cc:1315): Changing IceConnectionState 1 => 2
[005:255] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:211 -[AudioVideoInterfaceImpl peerConnection:didChangeIceConnectionState:]): ICE state changed: 2
[005:256] [9987] (turnport.cc:1320): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN create permission request sent, id=664c2f50365567495262716a
[005:256] [9987] (port.cc:851): Jingle:Conn[0x103875a00:audio:LznwORH2:1:0:relay:udp:120.92.22.201:55548->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|C--W|179896594928123390|-]: Connection created
[005:256] [9987] (p2ptransportchannel.cc:809): Jingle:Channel[audio|1|R_]: Created connection with origin=2, (2 total)
[005:256] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|R_]: Transport channel state changed from 2 to 1
[005:256] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:260] [9987] (turnport.cc:1326): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: TURN permission requested successfully, id=664c2f50365567495262716a, code=0, rtt=5
[005:260] [9987] (turnport.cc:1480): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Create permission for 10.10.140.187:64832 succeeded
[005:260] [9987] (turnport.cc:1493): Jingle:Port[0x103827200:audio:1:0:relay:Net[en5:10.10.130.0/23:Unknown]]: Scheduled create-permission-request in 240000ms.
[005:264] [9987] (port.cc:1373): Jingle:Conn[0x103875a00:audio:LznwORH2:1:0:relay:udp:120.92.22.201:55548->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|C--W|179896594928123390|-]: Sent STUN ping, id=6f344f3053464c762b426e32, use_candidate=1
[005:307] [9987] (port.cc:1373): Jingle:Conn[0x104883c00:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->8Ol9KTvY:1:1686052607:stun:udp:124.42.103.133:54147|C--W|7241540810645061118|-]: Sent STUN ping, id=704530336f71313341455379, use_candidate=0
[005:312] [9987] (port.cc:1373): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|CR-I|9115038255631187454|-]: Sent STUN ping, id=5753704d6676695877527541, use_candidate=1
[005:327] [9987] (dtlstransportchannel.cc:472): Jingle:Channel[audio|1|__]: Packet received before DTLS started.
[005:327] [9987] (dtlstransportchannel.cc:479): Jingle:Channel[audio|1|__]: Caching DTLS ClientHello packet until DTLS is started.
[005:327] [9987] (port.cc:976): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|CR-I|9115038255631187454|-]: Received STUN ping, id=65786d4a7a76693475685546
[005:328] [9987] (port.cc:605): Jingle:Port[0x104809000:audio:1:0:local:Net[en5:10.10.130.0/23:Unknown]]: Sent STUN ping response, to=10.10.140.187:64832, id=65786d4a7a76693475685546
[005:328] [9987] (port.cc:1319): Jingle:Conn[0x104094000:audio:wQnc7u1n:1:0:local:udp:10.10.130.88:57893->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|CRWS|9115038255631187454|2278]: Received STUN ping response, id=304b3578764a6b5671574559, code=0, rtt=113, use_candidate=0, pings_since_last_response=
[005:328] [9987] (dtlstransportchannel.cc:472): Jingle:Channel[audio|1|__]: Packet received before DTLS started.
[005:334] [9987] (dtlstransportchannel.cc:479): Jingle:Channel[audio|1|__]: Caching DTLS ClientHello packet until DTLS is started.
[005:334] [9987] (port.cc:1076): Jingle:Conn[0x103875a00:audio:LznwORH2:1:0:relay:udp:120.92.22.201:55548->ynyiJ4g2:1:2122260223:local:udp:10.10.140.187:64832|C--I|179896594928123390|-]: Connection pruned
[005:334] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|R_]: Transport channel state changed from 1 to 2
[005:334] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:334] [9987] (opensslstreamadapter.cc:756): BeginSSL: with peer
[005:334] [9987] (openssladapter.cc:851): SSL_accept:before accept initialization
[005:334] [9987] (openssladapter.cc:861): SSL_accept:error in SSLv3 read client hello A
[005:334] [9987] (dtlstransportchannel.cc:586): Jingle:Channel[audio|1|__]: DtlsTransportChannelWrapper: Started DTLS handshake
[005:334] [9987] (srtpfilter.cc:439): SRTP reset to init state
[005:334] [9987] (srtpfilter.cc:439): SRTP reset to init state
[005:334] [9987] (dtlstransportchannel.cc:593): Jingle:Channel[audio|1|__]: Handling cached DTLS ClientHello packet.
[005:334] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read client hello A
[005:334] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write server hello A
[005:334] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write certificate A
[005:335] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write key exchange A
[005:335] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write certificate request A
[005:335] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write server done A
[005:335] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 flush data
[005:335] [9987] (openssladapter.cc:861): SSL_accept:error in SSLv3 read client certificate A
[005:360] [9987] (port.cc:1373): Jingle:Conn[0x10202b400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->pKjpK0Zq:1:41885439:relay:udp:120.92.22.201:64935|C--W|179896594928123390|-]: Sent STUN ping, id=46705a49426e6e703535526c, use_candidate=0
[005:373] [9987] (port.cc:1319): Jingle:Conn[0x10202b400:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->pKjpK0Zq:1:41885439:relay:udp:120.92.22.201:64935|CRWS|179896594928123390|2253]: Received STUN ping response, id=46705a49426e6e703535526c, code=0, rtt=13, use_candidate=0, pings_since_last_response=
[005:373] [9987] (port.cc:1076): Jingle:Conn[0x10202b400:audio:yS55DQtX:1:0:prflx:udp:124.42.103.133:36850->pKjpK0Zq:1:41885439:relay:udp:120.92.22.201:64935|CRWS|179896594391252478|2253]: Connection pruned
[005:373] [9987] (port.cc:1076): Jingle:Conn[0x104883c00:audio:GcpVe4gs:1:0:local:udp:10.10.130.88:50550->8Ol9KTvY:1:1686052607:stun:udp:124.42.103.133:54147|C--I|7241540810645061118|-]: Connection pruned
[005:373] [9987] (p2ptransportchannel.cc:1321): Jingle:Channel[audio|1|RW]: Transport channel state changed from 1 to 2
[005:373] [9987] (transportcontroller.cc:622): audio TransportChannel 1 state changed. Check if state is complete.
[005:379] [9987] (opensslstreamadapter.cc:1088): Accepted peer certificate.
[005:379] [9987] (opensslstreamadapter.cc:1088): Accepted peer certificate.
[005:379] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read client certificate A
[005:380] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read client key exchange A
[005:381] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read certificate verify A
[005:390] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read change cipher spec
[005:390] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 read finished A
[005:390] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write session ticket A
[005:390] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write change cipher spec
[005:390] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 write finished A
[005:391] [9987] (openssladapter.cc:851): SSL_accept:SSLv3 flush data
[005:391] [9987] (dtlstransportchannel.cc:544): Jingle:Channel[audio|1|__]: DTLS handshake complete.
[005:391] [9987] (transportcontroller.cc:554): audio TransportChannel 1 writability changed to 1.
[005:391] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=up
[005:391] [9987] (channel.cc:928): Channel writable (audio) for the first time
[005:391] [10243] (congestion_controller.cc:291): SignalNetworkState Up
[005:391] [10499] (webrtcsession.cc:1352): Changing to ICE completed state because all transports are complete.
[005:391] [9987] (channel.cc:936): Using Cand[:3713327443:1:udp:2122260223:10.10.130.88:57893:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]->Cand[:830568635:1:udp:2122260223:10.10.140.187:64832:local::0:dUyp:072oOM1jZ4Ut6qiVdF/DlGc3:1:10:0]
[005:391] [10243] (congestion_controller.cc:367): Bitrate estimate state changed, BWE: 300000 bps.
[005:391] [10499] (webrtcsession.cc:1315): Changing IceConnectionState 1 => 2
[005:391] [9987] (channel.cc:995): Installing keys from DTLS-SRTP on audio RTP
[005:391] [10243] (call.cc:694): UpdateAggregateNetworkState: aggregate_state=up
[005:391] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:211 -[AudioVideoInterfaceImpl peerConnection:didChangeIceConnectionState:]): ICE state changed: 2
[005:391] [10243] (congestion_controller.cc:291): SignalNetworkState Up
[005:391] [10499] (webrtcsession.cc:1315): Changing IceConnectionState 2 => 3
[005:391] [10499] (RTCLogging.mm:31): (AudioVideoInterfaceImpl.m:211 -[AudioVideoInterfaceImpl peerConnection:didChangeIceConnectionState:]): ICE state changed: 3
[005:391] [9987] (srtpfilter.cc:150): SRTP activated with negotiated parameters: send cipher_suite 2 recv cipher_suite 2
[005:391] [10243] (channel.cc:1693): Changing voice state, recv=1 send=0
[005:391] [9987] (channel.cc:928): Channel writable (video) for the first time
[005:391] [9987] (channel.cc:936): Using Cand[:3713327443:1:udp:2122260223:10.10.130.88:57893:local::0:GOrQ:Ln5/XQgxfyyYtrCDuPHOihSt:1:50:0]->Cand[:830568635:1:udp:2122260223:10.10.140.187:64832:local::0:dUyp:072oOM1jZ4Ut6qiVdF/DlGc3:1:10:0]
[005:391] [9987] (channel.cc:995): Installing keys from DTLS-SRTP on video RTP
[005:392] [9987] (srtpfilter.cc:150): SRTP activated with negotiated parameters: send cipher_suite 2 recv cipher_suite 2
[005:392] [10243] (channel.cc:1956): Changing video state, send=0
[005:806] [10243] (rtp_receiver_audio.cc:197): Received first audio RTP packet
[005:807] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 1029649540 with payload type: 116, timestamp: 3232293757, sequence number: 3369, arrival time: 293876571
[005:807] [10243] (rtp_receiver_video.cc:75): Received first video RTP packet
[005:821] [10243] (jitter_buffer.cc:1201): Received first complete key frame
[005:833] [37391] (video_receiver.cc:283): Received first complete decodable video frame
[005:833] [37391] (codec_database.cc:527): Initializing decoder with payload type '100'.
[005:980] [10243] (rtp_receiver_audio.cc:197): Received first audio RTP packet
[005:995] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 432006656 with payload type: 116, timestamp: 3232417024, sequence number: 27355, arrival time: 293876760
[005:995] [10243] (rtp_receiver_video.cc:75): Received first video RTP packet
[005:995] [10243] (jitter_buffer.cc:1201): Received first complete key frame
[006:007] [38915] (video_receiver.cc:283): Received first complete decodable video frame
[006:007] [38915] (codec_database.cc:527): Initializing decoder with payload type '100'.
[006:870] [9987] (network.cc:842): Connect failed with 65
[008:872] [9987] (network.cc:842): Connect failed with 65
[010:873] [9987] (network.cc:842): Connect failed with 65
[012:874] [9987] (network.cc:842): Connect failed with 65
[014:879] [9987] (network.cc:842): Connect failed with 65
[015:216] [9987] (port.cc:1419): Connection deleted with number of pings sent: 1
[015:216] [9987] (p2ptransportchannel.cc:1654): Jingle:Channel[audio|1|RW]: Removed connection (2 remaining)
[015:467] [9987] (port.cc:1419): Connection deleted with number of pings sent: 1
[015:467] [9987] (p2ptransportchannel.cc:1654): Jingle:Channel[audio|1|RW]: Removed connection (1 remaining)
[015:811] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 1029649540 with payload type: 116, timestamp: 3233195377, sequence number: 3899, arrival time: 293886576
[016:082] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 432006656 with payload type: 116, timestamp: 3233322874, sequence number: 27743, arrival time: 293886847
[016:880] [9987] (network.cc:842): Connect failed with 65
[018:881] [9987] (network.cc:842): Connect failed with 65
[020:882] [9987] (network.cc:842): Connect failed with 65
[022:884] [9987] (network.cc:842): Connect failed with 65
[024:884] [9987] (network.cc:842): Connect failed with 65
[025:827] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 1029649540 with payload type: 116, timestamp: 3234097537, sequence number: 4617, arrival time: 293896591
[025:982] [35591] (audio_device_buffer.cc:423): [REC : 10000msec, 48kHz] callbacks: 0, samples: 0, rate: 0
[025:982] [35591] (audio_device_buffer.cc:432): [PLAY: 10000msec, 48kHz] callbacks: 1000, samples: 480000, rate: 48000
[026:100] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 432006656 with payload type: 116, timestamp: 3234229894, sequence number: 28359, arrival time: 293896864
[026:885] [9987] (network.cc:842): Connect failed with 65
[028:886] [9987] (network.cc:842): Connect failed with 65
[030:886] [9987] (network.cc:842): Connect failed with 65
[032:886] [9987] (network.cc:842): Connect failed with 65
[034:887] [9987] (network.cc:842): Connect failed with 65
[035:526] [9987] (port.cc:1419): Connection deleted with number of pings sent: 1
[035:526] [9987] (p2ptransportchannel.cc:1654): Jingle:Channel[audio|1|RW]: Removed connection (1 remaining)
[035:829] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 1029649540 with payload type: 116, timestamp: 3234996907, sequence number: 5368, arrival time: 293906593
[036:109] [10243] (rtp_stream_receiver.cc:306): Packet received on SSRC: 432006656 with payload type: 116, timestamp: 3235129084, sequence number: 29111, arrival time: 293906873
[036:247] [35591] (audio_device_buffer.cc:423): [REC : 10265msec, 48kHz] callbacks: 0, samples: 0, rate: 0
[036:247] [35591] (audio_device_buffer.cc:432): [PLAY: 10265msec, 48kHz] callbacks: 1026, samples: 492480, rate: 49248
[036:890] [9987] (network.cc:842): Connect failed with 65
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