When start the momo:
./momo --no-audio-device --log-level 1 test
On the newly installed Jetson Nano when try to connect with H264 or VP8 gave this error message:
Wrong JPEG library version: library is 62, caller expects 80
Not installed the libjpeg 6.2, just the libjpeg8 and libjpeg-turbo8.
[006:972][2747] (websocket.cpp:282): Websocket::OnWrite this=2902f970 ec=Success
Wrong JPEG library version: library is 62, caller expects 80
[006:978][2754] (port.cc:386): Received STUN BINDING request id=6177354536476734786e6835 from unknown address 192.168.10.x:57747
[006:979][2754] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2
[006:979][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:979][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:979][2754] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping.
[006:979][2756] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1
[006:979][2756] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1
[006:979][2756] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[006:980][2754] (connection.cc:1167): Conn[7801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:1pPAcXyO:1:0:local:udp:192.168.10.x:46405->5beevxLi:1:1845501695:prflx:udp:192.168.10.x:57747|CR-W|-|1|0|79263694289820871NG request, id=2b6c51326f334539596c3465, use_candidate=0, nomination=0
[006:980][2754] (connection.cc:1082): Conn[7801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:1pPAcXyO:1:0:local:udp:192.168.10.x:46405->5beevxLi:1:1845501695:prflx:udp:192.168.10.x:57747|CR-I|-|1|0|79263694289820871INDING response, id=2b6c51326f334539596c3465, code=0, rtt=0, pings_since_last_response=2b6c51326f334539596c3465
[006:980][2754] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0
[006:981][2754] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete
[006:981][2754] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[006:981][2754] (channel.cc:354): Network route changed for {mid: 0, media_type: video}
[006:981][2754] (channel.cc:354): Network route changed for {mid: 1, media_type: audio}
[006:981][2755] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ]
[006:981][2754] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 0
[006:981][2754] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake
[006:981][2754] (srtp_transport.cc:365): The params in SRTP transport are reset.
[006:981][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:981][2756] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2
[006:981][2756] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2
[006:981][2756] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected
[006:985][2754] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete.
[006:985][2754] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1.
[006:985][2755] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up
[006:985][2765] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller
[006:985][2756] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable.
[006:985][2756] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2
[006:985][2765] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3
[006:985][2765] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st
[006:985][2765] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st
[006:985][2765] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
[006:986][2765] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled
[006:986][2765] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps
[006:986][2765] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000
[006:986][2765] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5)
[006:986][2765] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)
[006:986][2765] (bitrate_allocator.cc:394): Current BWE 300000
[006:985][2754] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0
root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# clear
root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# ./momo --no-audio-device --log-level 1 --no-google-stun test
[000:000][2807] (v4l2_video_capturer.cpp:76): GetDeviceName(0): device_name=HD Pro Webcam C920, unique_name=usb-70090000.xusb-2.1
[000:003][2807] (v4l2_video_capturer.cpp:230): Video Capture enumerats supported image formats:
[000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = YUYV, description = 'd9f262ac' }
[000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = H264, description = 'd9f262ac' }
[000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = MJPG, description = 'd9f262ac' }
[000:003][2807] (v4l2_video_capturer.cpp:248): We prefer format MJPG
[000:162][2807] (jetson_v4l2_capturer.cpp:29): Get Capture
[000:163][2814] (audio_device_buffer.cc:64): AudioDeviceBuffer::ctor
[000:163][2814] (audio_device_impl.cc:136): current platform is Linux
[000:163][2814] (audio_device_impl.cc:155): CreatePlatformSpecificObjects
[000:163][2814] (audio_device_impl.cc:947): PlatformAudioLayer
[000:163][2814] (audio_device_impl.cc:266): PulseAudio support is enabled.
[000:163][2814] (audio_device_impl.cc:299): Dummy Audio APIs will be utilized.
[000:163][2814] (audio_device_impl.cc:312): AttachAudioBuffer
[000:251][2807] (audio_processing_impl.cc:278): Injected APM submodules:
Echo control factory: 0
Echo detector: 0
Capture analyzer: 0
Capture post processor: 0
Render pre processor: 0
[000:251][2807] (webrtc_voice_engine.cc:268): WebRtcVoiceEngine::WebRtcVoiceEngine
[000:251][2814] (webrtc_voice_engine.cc:290): WebRtcVoiceEngine::Init
[000:252][2814] (audio_device_impl.cc:332): Init
[000:252][2814] (audio_device_impl.cc:676): SetPlayoutDevice(0)
[000:252][2814] (adm_helpers.cc:44): Unable to set playout device.
[000:252][2814] (audio_device_impl.cc:851): RegisterAudioCallback
[000:252][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_y_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, }
[000:252][2814] (audio_device_impl.cc:867): BuiltInAECIsAvailable
[000:252][2814] (audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
[000:252][2814] (audio_device_impl.cc:870): output: 0
[000:252][2814] (audio_device_impl.cc:883): BuiltInAGCIsAvailable
[000:252][2814] (audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
[000:252][2814] (audio_device_impl.cc:886): output: 0
[000:252][2814] (audio_device_impl.cc:899): BuiltInNSIsAvailable
[000:252][2814] (audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
[000:252][2814] (audio_device_impl.cc:902): output: 0
[000:252][2814] (webrtc_voice_engine.cc:495): Stereo swapping enabled? 0
[000:252][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[000:252][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[000:252][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[000:252][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[000:252][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[000:252][2814] (webrtc_voice_engine.cc:589): NS set to 1
[000:252][2814] (webrtc_voice_engine.cc:593): Typing detection is enabled? 1
[000:252][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[000:253][2814] (agc_manager_direct.cc:65): [agc] GetMinMicLevel
[000:253][2814] (agc_manager_direct.cc:69): [agc] Using default min mic level: 12
[003:983][2807] (p2p_websocket_session.cpp:28): P2PWebsocketSession
[003:984][2807] (p2p_websocket_session.cpp:37): Run
[003:984][2807] (p2p_websocket_session.cpp:57): OnAccept: system:0
[006:214][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:214][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:214][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"offer","sdp":"v=0\r\no=- 4331006092786853723 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=extmap-allow-mixed\r\na=msid-so 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r6 F0:38:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/exps-send-time\r\na=extmap:3 urn:3gpp:video-orientation\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:5http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:9 urn:isdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:102 H264/90000\r\na=r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtpllowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcr\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmartcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=rna=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\na=rtpmap:35 flexfec-03/90000\r\na=rtcp-fb:35 goog-remb\r\na=rtcp-fb:35 transport-cc\r\na=f0000000\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r\ F0:38:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrthdrext/abs-send-time\r\na=extmap:4http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpm=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000hone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=application 9 UDP/DTLS/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r\na=fi8:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:2\r\na=sctp-port:5000\r\na=max-message-size:262144\r\n"}
[006:215][2814] (rtc_event_log_impl.cc:43): Creating legacy encoder for RTC event log.
[006:215][2814] (peer_connection_factory.cc:331): Using default network controller factory
[006:215][2814] (bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive
[006:216][2814] (cpu_info.cc:53): Available number of cores: 4
[006:216][2814] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
[006:216][2814] (remote_bitrate_estimator_single_stream.cc:72): RemoteBitrateEstimatorSingleStream: Instantiating.
[006:216][2814] (remote_estimator_proxy.cc:47): Maximum interval between transport feedback RTCP messages (ms): 250
[006:217][2813] (openssl_key_pair.cc:38): Making key pair
[006:217][2815] (rtp_transmission_manager.cc:187): Adding video transceiver in response to a call to AddTrack.
[006:218][2813] (openssl_key_pair.cc:91): Returning key pair
[006:218][2813] (boringssl_certificate.cc:187): Making certificate for WebRTC
[006:218][2813] (boringssl_certificate.cc:243): Returning certificate
[006:219][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[006:219][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[006:219][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[006:219][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[006:219][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[006:219][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[006:219][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[006:219][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[006:219][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0
[006:219][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport 88003020
[006:219][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[006:219][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[006:219][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[006:219][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[006:219][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[006:219][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[006:219][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[006:220][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[006:220][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 1 transport 0
[006:220][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 1 transport 88005e90
[006:220][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[006:220][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[006:220][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[006:220][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[006:220][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[006:220][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[006:220][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[006:220][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[006:220][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 2 transport 0
[006:220][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 2 transport 88008610
[006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled
[006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled
[006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled
[006:220][2814] (webrtc_video_engine.cc:643): CreateMediaChannel. Options: VideoOptions {}
[006:220][2814] (channel.cc:137): Created channel: {mid: 0, media_type: video}
[006:221][2813] (rtp_demuxer.cc:154): Added sink = 80077348 for criteria {mid: 0, rsid: , ssrcs: [], payload_types = []}
[006:221][2815] (sdp_offer_answer.cc:3250): Adding audio transceiver for MID=1 at i=1 in response to the remote description.
[006:221][2814] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jithandling: 0, }
[006:221][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_ji_handling: 0, }
[006:221][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[006:221][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[006:221][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[006:221][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[006:221][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[006:221][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[006:221][2814] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms:r_enable_rtx_handling: 0, }
[006:221][2814] (channel.cc:137): Created channel: {mid: 1, media_type: audio}
[006:221][2813] (rtp_demuxer.cc:154): Added sink = 80077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[006:222][2815] (sdp_offer_answer.cc:3371): Creating data channel, mid=2
[006:222][2813] (peer_connection.cc:2382): Setting up data channel transport for mid=2
[006:222][2815] (sdp_offer_answer.cc:2459): Session: 8528378070752557626 Old state: stable New state: have-remote-offer
[006:222][2814] (webrtc_video_engine.cc:1573): ResetUnsignaledRecvStream.
[006:222][2814] (webrtc_voice_engine.cc:2047): ResetUnsignaledRecvStream.
[006:222][2814] (channel.cc:1082): Setting remote video description for {mid: 0, media_type: video}
[006:222][2814] (webrtc_video_engine.cc:884): SetSendParameters: {codecs: [VideoCodec[102:H264], VideoCodec[121:rtx], VideoCodec[127:H264], VideoCodec[120:rtx], VideoCodec[125:H264], VideoCodec[107:rtx], VideoCodec[109:rtx], VideoCodec[124:H264], VideoCodec[119:rtx], VideoCodec[123:H264], VideoCodec[118:rtx], VideoCodec[114:red], VideoCodec[115:rtx], VideoCodec[116:ulpfec], VideoCodec[35:flexfec-03]], conference_mode: no, extensparams:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timi://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repd: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[102:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[127:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[125:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[108:H264]
[006:222][2814] (webrtc_video_engine.cc:1041): SetFeedbackParameters on all the receive streams because the send codec or RTCP mode has changed.
[006:223][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:223][2814] (channel.cc:889): Setting remote voice description for {mid: 1, media_type: audio}
[006:223][2814] (webrtc_voice_engine.cc:1416): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:ephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {rg/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdesd, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 1, options: AudioOptions {}}
[006:223][2814] (webrtc_voice_engine.cc:1840): Recreate all the receive streams because the send codec has changed.
[006:223][2814] (webrtc_voice_engine.cc:2335): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[006:223][2814] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {}
[006:223][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_ji_handling: 0, }
[006:223][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[006:223][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[006:223][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[006:223][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[006:223][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[006:223][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[006:223][2814] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms:r_enable_rtx_handling: 0, }
[006:223][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:223][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:224][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:224][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51664 typ host generation 0st 999","sdpMid":"0","sdpMLineIndex":0}}
[006:224][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51664 typ host genetwork-cost 999
[006:224][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:224][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:224][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51666 typ host generation 0st 999","sdpMid":"1","sdpMLineIndex":1}}
[006:224][2813] (peer_connection.cc:2587): 0 is not ready to use the remote candidate because the local or remote description is not set.
[006:225][2815] (rtc_connection.cpp:169): Created session description : v=0
o=- 8528378070752557626 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
m=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 2334982981 921515023
a=ssrc:2334982981 cname:9z2Fp9qEvpL87sim
a=ssrc:2334982981 msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:2334982981 mslabel:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
a=ssrc:2334982981 label:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:921515023 cname:9z2Fp9qEvpL87sim
a=ssrc:921515023 msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:921515023 mslabel:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
a=ssrc:921515023 label:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=inactive
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:2
a=sctp-port:5000
a=max-message-size:262144
[006:225][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0
[006:225][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport 88003020
[006:225][2813] (p2p_transport_channel.cc:517): Set ICE ufrag: EnjI pwd: B+h3GCWvL/6K3FyW0S7w92Vd on transport 0
[006:225][2813] (dtls_transport.cc:367): DtlsTransport[0|1|__]: DTLS setup complete.
[006:225][2813] (rtp_demuxer.cc:250): Removed sink = 80077a48 bindings
[006:225][2813] (rtp_demuxer.cc:154): Added sink = 80077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[006:225][2815] (rtp_transceiver.cc:313): Changing transceiver (MID=0) current direction from to kSendOnly.
[006:226][2815] (rtp_transceiver.cc:313): Changing transceiver (MID=1) current direction from to kInactive.
[006:226][2815] (sdp_offer_answer.cc:2459): Session: 8528378070752557626 Old state: have-remote-offer New state: stable
[006:226][2814] (channel.cc:512): Channel enabled: {mid: 0, media_type: video}
[006:226][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:226][2814] (channel.cc:512): Channel enabled: {mid: 1, media_type: audio}
[006:226][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:226][2814] (channel.cc:991): Setting local video description for {mid: 0, media_type: video}
[006:226][2814] (webrtc_video_engine.cc:1341): AddSendStream: {id:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX;ssrcs:[2334982981,921515023];ssrc_groups:{semantics:FID;ssrcs:[2334982981,921515023]};cname:9z2Fp9qEvpL87sim;stream_iYK9xjItzSQkPj;}
[006:226][2814] (webrtc_video_engine.cc:2249): RecreateWebRtcStream (send) because of SetCodec.
[006:226][2814] (paced_sender.cc:187): ProcessThreadAttached 0x80071d60
[006:227][2814] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3
[006:227][2814] (video_stream_encoder.cc:2189): Automatic animation detection experiment is disabled.
[006:227][2824] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[006:227][2824] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[006:227][2824] (video_send_stream_impl.cc:245): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2334982981], rids: [], mid: '0', rtcp_mode: RtcpMode::kReducedSize, mextmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://wnts/rtp-hdrext/color-space, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.orgt/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}, {uri: urn:ietf:params:rtp-hdd, id: 10}], lntf: {enabled: false}, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 116, red_payload_type: 114, red_rtx_payload_type: 115}, payload_name: H264, payload_type: 102, raw_payload: false, flex, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [921515023], payload_type: 121}, c_name: 9z2Fp9qEvpL87sim}, rtcp_report_interval_ms: 1000, send_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, susp off}
[006:227][2827] (video_stream_encoder.cc:798): SetStartBitrate 300000
[006:227][2824] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[006:227][2827] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[006:227][2814] (webrtc_video_engine.cc:1381): SetLocalSsrc on all the receive streams because we added a send stream.
[006:227][2814] (channel.cc:671): Add send stream ssrc: 2334982981 into {mid: 0, media_type: video}
[006:227][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:228][2824] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[006:228][2814] (channel.cc:829): Setting local voice description for {mid: 1, media_type: audio}
[006:228][2814] (webrtc_voice_engine.cc:1462): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:ephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {rg/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdesd, id: 11}]}
[006:228][2814] (webrtc_voice_engine.cc:1630): Setting receive voice codecs.
[006:228][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:228][2813] (p2p_transport_channel.cc:1229): Asynchronously resolving ICE candidate hostname 7fe0e395-c1e7-463a-b210-2344e6824c6e.local
[006:228][2814] (webrtc_video_engine.cc:1300): SetVideoSend (ssrc= 2334982981, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))
[006:228][2827] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[006:228][2813] (basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
[006:229][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51666 typ host genetwork-cost 999
[006:229][2815] (peer_connection.cc:1805): Changing IceConnectionState 0 => 1
[006:229][2813] (basic_port_allocator.cc:111): Filtered out ignored networks:
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[006:229][2813] (basic_port_allocator.cc:861): Network manager has started
[006:229][2813] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[006:229][2813] (basic_port_allocator.cc:111): Filtered out ignored networks:
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[006:229][2813] (basic_port_allocator.cc:776): Allocate ports on 1 networks
[006:229][2813] (basic_port_allocator.cc:1361): Net[eth0:192.168.10.x/24:Ethernet:id=1]: Allocation Phase=Udp
[006:229][2813] (port.cc:187): Port[88005510::1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port created with network cost 0
[006:229][2813] (basic_port_allocator.cc:885): Adding allocated port for 0
[006:229][2813] (basic_port_allocator.cc:907): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Added port to allocator
[006:229][2807] (websocket.cpp:282): Websocket::OnWrite this=18efa970 ec=Success
[006:230][2813] (basic_port_allocator.cc:925): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Gathered candidate: Cand[:1009828554:1:udp:2122260223:192.168.10.x:43907:local::0:EnjI:B+h3GCWvL/6K3F
[006:230][2813] (basic_port_allocator.cc:958): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port ready.
[006:230][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:230][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:230][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51668 typ host generation 0st 999","sdpMid":"2","sdpMLineIndex":2}}
[006:230][2813] (basic_port_allocator.cc:1069): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port completed gathering candidates.
[006:230][2815] (p2p_websocket_session.cpp:171): OnIceCandidate
[006:230][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51668 typ host genetwork-cost 999
[006:230][2813] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[006:230][2807] (websocket.cpp:282): Websocket::OnWrite this=18efa970 ec=Success
[006:236][2813] (port.cc:386): Received STUN BINDING request id=2f66747958694e6350765744 from unknown address 192.168.10.x:51664
[006:237][2813] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2
[006:237][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:237][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:237][2813] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping.
[006:237][2815] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1
[006:237][2815] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1
[006:237][2815] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[006:238][2813] (connection.cc:1167): Conn[8801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:BIp3Al1I:1:0:local:udp:192.168.10.x:43907->0uZrj/Rg:1:1845501695:prflx:udp:192.168.10.x:51664|CR-W|-|1|0|79263694289820871NG request, id=475a65536561683349713235, use_candidate=0, nomination=0
[006:238][2813] (connection.cc:1082): Conn[8801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:BIp3Al1I:1:0:local:udp:192.168.10.x:43907->0uZrj/Rg:1:1845501695:prflx:udp:192.168.10.x:51664|CR-I|-|1|0|79263694289820871INDING response, id=475a65536561683349713235, code=0, rtt=1, pings_since_last_response=475a65536561683349713235
[006:238][2813] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0
[006:238][2813] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete
[006:238][2813] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[006:238][2813] (channel.cc:354): Network route changed for {mid: 0, media_type: video}
[006:238][2813] (channel.cc:354): Network route changed for {mid: 1, media_type: audio}
[006:238][2814] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ]
[006:238][2813] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1
[006:239][2813] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake
[006:239][2813] (srtp_transport.cc:365): The params in SRTP transport are reset.
[006:239][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[006:239][2815] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2
[006:239][2815] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2
[006:239][2815] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected
Wrong JPEG library version: library is 62, caller expects 80
[006:243][2813] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete.
[006:243][2813] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1.
[006:243][2814] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up
[006:243][2824] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller
[006:243][2813] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0
[006:243][2815] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable.
[006:243][2815] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2
[006:243][2824] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3
[006:243][2824] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st
[006:244][2824] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st
[006:244][2824] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
[006:244][2824] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled
[006:244][2824] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps
[006:244][2824] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000
[006:244][2824] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5)
[006:244][2824] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)
[006:244][2824] (bitrate_allocator.cc:394): Current BWE 300000
root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano#
root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# ./momo --no-audio-device --log-level 1 test
[000:000][2927] (v4l2_video_capturer.cpp:76): GetDeviceName(0): device_name=HD Pro Webcam C920, unique_name=usb-70090000.xusb-2.1
[000:003][2927] (v4l2_video_capturer.cpp:230): Video Capture enumerats supported image formats:
[000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = YUYV, description = 'f5cf253c' }
[000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = H264, description = 'f5cf253c' }
[000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = MJPG, description = 'f5cf253c' }
[000:003][2927] (v4l2_video_capturer.cpp:248): We prefer format MJPG
[000:162][2927] (jetson_v4l2_capturer.cpp:29): Get Capture
[000:163][2935] (audio_device_buffer.cc:64): AudioDeviceBuffer::ctor
[000:163][2935] (audio_device_impl.cc:136): current platform is Linux
[000:163][2935] (audio_device_impl.cc:155): CreatePlatformSpecificObjects
[000:163][2935] (audio_device_impl.cc:947): PlatformAudioLayer
[000:163][2935] (audio_device_impl.cc:266): PulseAudio support is enabled.
[000:163][2935] (audio_device_impl.cc:299): Dummy Audio APIs will be utilized.
[000:163][2935] (audio_device_impl.cc:312): AttachAudioBuffer
[000:253][2927] (audio_processing_impl.cc:278): Injected APM submodules:
Echo control factory: 0
Echo detector: 0
Capture analyzer: 0
Capture post processor: 0
Render pre processor: 0
[000:253][2927] (webrtc_voice_engine.cc:268): WebRtcVoiceEngine::WebRtcVoiceEngine
[000:254][2935] (webrtc_voice_engine.cc:290): WebRtcVoiceEngine::Init
[000:254][2935] (audio_device_impl.cc:332): Init
[000:254][2935] (audio_device_impl.cc:676): SetPlayoutDevice(0)
[000:254][2935] (adm_helpers.cc:44): Unable to set playout device.
[000:255][2935] (audio_device_impl.cc:851): RegisterAudioCallback
[000:255][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, }
[000:255][2935] (audio_device_impl.cc:867): BuiltInAECIsAvailable
[000:255][2935] (audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
[000:255][2935] (audio_device_impl.cc:870): output: 0
[000:255][2935] (audio_device_impl.cc:883): BuiltInAGCIsAvailable
[000:255][2935] (audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
[000:255][2935] (audio_device_impl.cc:886): output: 0
[000:255][2935] (audio_device_impl.cc:899): BuiltInNSIsAvailable
[000:255][2935] (audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
[000:255][2935] (audio_device_impl.cc:902): output: 0
[000:255][2935] (webrtc_voice_engine.cc:495): Stereo swapping enabled? 0
[000:255][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[000:255][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[000:255][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[000:255][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[000:255][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[000:255][2935] (webrtc_voice_engine.cc:589): NS set to 1
[000:255][2935] (webrtc_voice_engine.cc:593): Typing detection is enabled? 1
[000:255][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[000:256][2935] (agc_manager_direct.cc:65): [agc] GetMinMicLevel
[000:256][2935] (agc_manager_direct.cc:69): [agc] Using default min mic level: 12
[006:928][2927] (p2p_websocket_session.cpp:28): P2PWebsocketSession
[006:928][2927] (p2p_websocket_session.cpp:37): Run
[006:928][2927] (p2p_websocket_session.cpp:57): OnAccept: system:0
[008:802][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:803][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:803][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"offer","sdp":"v=0\r\no=- 4154535124576789392 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=extmap-allow-mixed\r\na=msid-semantic: WMS\r\nm=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 urn:3gpp:video-orientation\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:5http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:6http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=fmtp:121 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcp-fb:124 transport-cc\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmap:123 H264/90000\r\na=rtcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=rtpmap:118 rtx/90000\r\na=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\na=rtpmap:35 flexfec-03/90000\r\na=rtcp-fb:35 goog-remb\r\na=rtcp-fb:35 transport-cc\r\na=fmtp:35 repair-window=10000000\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=application 9 UDP/DTLS/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:2\r\na=sctp-port:5000\r\na=max-message-size:262144\r\n"}
[008:803][2935] (rtc_event_log_impl.cc:43): Creating legacy encoder for RTC event log.
[008:803][2935] (peer_connection_factory.cc:331): Using default network controller factory
[008:804][2935] (bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive
[008:804][2935] (cpu_info.cc:53): Available number of cores: 4
[008:804][2935] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
[008:804][2935] (remote_bitrate_estimator_single_stream.cc:72): RemoteBitrateEstimatorSingleStream: Instantiating.
[008:804][2935] (remote_estimator_proxy.cc:47): Maximum interval between transport feedback RTCP messages (ms): 250
[008:805][2934] (openssl_key_pair.cc:38): Making key pair
[008:806][2936] (rtp_transmission_manager.cc:187): Adding video transceiver in response to a call to AddTrack.
[008:806][2934] (openssl_key_pair.cc:91): Returning key pair
[008:806][2934] (boringssl_certificate.cc:187): Making certificate for WebRTC
[008:807][2934] (boringssl_certificate.cc:243): Returning certificate
[008:807][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[008:807][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0
[008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport a4003090
[008:808][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[008:808][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 1 transport 0
[008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 1 transport a4005f00
[008:808][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds.
[008:808][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds
[008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0
[008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500
[008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0
[008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000
[008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000
[008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport.
[008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 2 transport 0
[008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 2 transport a4008680
[008:808][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled
[008:808][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled
[008:809][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled
[008:809][2935] (webrtc_video_engine.cc:643): CreateMediaChannel. Options: VideoOptions {}
[008:809][2935] (channel.cc:137): Created channel: {mid: 0, media_type: video}
[008:809][2934] (rtp_demuxer.cc:154): Added sink = 9c077348 for criteria {mid: 0, rsid: , ssrcs: [], payload_types = []}
[008:809][2936] (sdp_offer_answer.cc:3250): Adding audio transceiver for MID=1 at i=1 in response to the remote description.
[008:810][2935] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[008:810][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[008:810][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[008:810][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[008:810][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[008:810][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[008:810][2935] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (channel.cc:137): Created channel: {mid: 1, media_type: audio}
[008:810][2934] (rtp_demuxer.cc:154): Added sink = 9c077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[008:810][2936] (sdp_offer_answer.cc:3371): Creating data channel, mid=2
[008:810][2934] (peer_connection.cc:2382): Setting up data channel transport for mid=2
[008:810][2936] (sdp_offer_answer.cc:2459): Session: 481305330888516940 Old state: stable New state: have-remote-offer
[008:810][2935] (webrtc_video_engine.cc:1573): ResetUnsignaledRecvStream.
[008:810][2935] (webrtc_voice_engine.cc:2047): ResetUnsignaledRecvStream.
[008:810][2935] (channel.cc:1082): Setting remote video description for {mid: 0, media_type: video}
[008:811][2935] (webrtc_video_engine.cc:884): SetSendParameters: {codecs: [VideoCodec[102:H264], VideoCodec[121:rtx], VideoCodec[127:H264], VideoCodec[120:rtx], VideoCodec[125:H264], VideoCodec[107:rtx], VideoCodec[108:H264], VideoCodec[109:rtx], VideoCodec[124:H264], VideoCodec[119:rtx], VideoCodec[123:H264], VideoCodec[118:rtx], VideoCodec[114:red], VideoCodec[115:rtx], VideoCodec[116:ulpfec], VideoCodec[35:flexfec-03]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[102:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[127:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[125:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[108:H264]
[008:811][2935] (webrtc_video_engine.cc:1041): SetFeedbackParameters on all the receive streams because the send codec or RTCP mode has changed.
[008:811][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:811][2935] (channel.cc:889): Setting remote voice description for {mid: 1, media_type: audio}
[008:811][2935] (webrtc_voice_engine.cc:1416): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 1, options: AudioOptions {}}
[008:811][2935] (webrtc_voice_engine.cc:1840): Recreate all the receive streams because the send codec has changed.
[008:811][2935] (webrtc_voice_engine.cc:2335): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[008:811][2935] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {}
[008:811][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:811][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[008:811][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[008:811][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[008:811][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[008:811][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[008:811][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[008:811][2935] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:811][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:812][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:812][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:812][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50485 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"0","sdpMLineIndex":0}}
[008:813][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50485 typ host generation 0 ufrag biOA network-cost 999
[008:813][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:813][2934] (peer_connection.cc:2587): 0 is not ready to use the remote candidate because the local or remote description is not set.
[008:813][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:813][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50487 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"1","sdpMLineIndex":1}}
[008:813][2936] (rtc_connection.cpp:169): Created session description : v=0
o=- 481305330888516940 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS /l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
m=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 142137631 2598199993
a=ssrc:142137631 cname:kz/tKo60LC6c+QVq
a=ssrc:142137631 msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:142137631 mslabel:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
a=ssrc:142137631 label:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:2598199993 cname:kz/tKo60LC6c+QVq
a=ssrc:2598199993 msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:2598199993 mslabel:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
a=ssrc:2598199993 label:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=inactive
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:2
a=sctp-port:5000
a=max-message-size:262144
[008:816][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0
[008:816][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport a4003090
[008:816][2934] (p2p_transport_channel.cc:517): Set ICE ufrag: NdQT pwd: 3Y+jOvlHqoT/UpD1wJbFv8ld on transport 0
[008:816][2934] (dtls_transport.cc:367): DtlsTransport[0|1|__]: DTLS setup complete.
[008:816][2934] (rtp_demuxer.cc:250): Removed sink = 9c077a48 bindings
[008:816][2934] (rtp_demuxer.cc:154): Added sink = 9c077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[008:816][2936] (rtp_transceiver.cc:313): Changing transceiver (MID=0) current direction from to kSendOnly.
[008:816][2936] (rtp_transceiver.cc:313): Changing transceiver (MID=1) current direction from to kInactive.
[008:816][2936] (sdp_offer_answer.cc:2459): Session: 481305330888516940 Old state: have-remote-offer New state: stable
[008:816][2935] (channel.cc:512): Channel enabled: {mid: 0, media_type: video}
[008:817][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:817][2935] (channel.cc:512): Channel enabled: {mid: 1, media_type: audio}
[008:817][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:817][2935] (channel.cc:991): Setting local video description for {mid: 0, media_type: video}
[008:817][2935] (webrtc_video_engine.cc:1341): AddSendStream: {id:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn;ssrcs:[142137631,2598199993];ssrc_groups:{semantics:FID;ssrcs:[142137631,2598199993]};cname:kz/tKo60LC6c+QVq;stream_ids:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX;}
[008:817][2935] (webrtc_video_engine.cc:2249): RecreateWebRtcStream (send) because of SetCodec.
[008:817][2935] (paced_sender.cc:187): ProcessThreadAttached 0x9c071d60
[008:817][2935] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
[008:817][2935] (video_stream_encoder.cc:2189): Automatic animation detection experiment is disabled.
[008:818][2983] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[008:818][2983] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[008:818][2983] (video_send_stream_impl.cc:245): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [142137631], rids: [], mid: '0', rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}], lntf: {enabled: false}, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 116, red_payload_type: 114, red_rtx_payload_type: 115}, payload_name: H264, payload_type: 102, raw_payload: false, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2598199993], payload_type: 121}, c_name: kz/tKo60LC6c+QVq}, rtcp_report_interval_ms: 1000, send_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
[008:818][2986] (video_stream_encoder.cc:798): SetStartBitrate 300000
[008:818][2983] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[008:818][2986] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[008:818][2935] (webrtc_video_engine.cc:1381): SetLocalSsrc on all the receive streams because we added a send stream.
[008:818][2935] (channel.cc:671): Add send stream ssrc: 142137631 into {mid: 0, media_type: video}
[008:819][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:819][2983] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[008:819][2935] (channel.cc:829): Setting local voice description for {mid: 1, media_type: audio}
[008:819][2935] (webrtc_voice_engine.cc:1462): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}]}
[008:819][2935] (webrtc_voice_engine.cc:1630): Setting receive voice codecs.
[008:819][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:819][2934] (p2p_transport_channel.cc:1229): Asynchronously resolving ICE candidate hostname 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local
[008:819][2935] (webrtc_video_engine.cc:1300): SetVideoSend (ssrc= 142137631, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))
[008:819][2986] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[008:821][2934] (basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
[008:821][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50487 typ host generation 0 ufrag biOA network-cost 999
[008:821][2936] (peer_connection.cc:1805): Changing IceConnectionState 0 => 1
[008:821][2927] (websocket.cpp:282): Websocket::OnWrite this=2cc8d970 ec=Success
[008:822][2934] (basic_port_allocator.cc:111): Filtered out ignored networks:
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[008:822][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[008:822][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:822][2934] (basic_port_allocator.cc:861): Network manager has started
[008:822][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50489 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"2","sdpMLineIndex":2}}
[008:822][2934] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[008:822][2934] (basic_port_allocator.cc:111): Filtered out ignored networks:
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[008:822][2934] (basic_port_allocator.cc:776): Allocate ports on 1 networks
[008:822][2934] (basic_port_allocator.cc:1361): Net[eth0:192.168.10.x/24:Ethernet:id=1]: Allocation Phase=Udp
[008:822][2934] (port.cc:187): Port[a4005580::1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port created with network cost 0
[008:822][2934] (basic_port_allocator.cc:1433): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[008:822][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50489 typ host generation 0 ufrag biOA network-cost 999
[008:822][2934] (basic_port_allocator.cc:885): Adding allocated port for 0
[008:822][2934] (basic_port_allocator.cc:907): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Added port to allocator
[008:822][2934] (basic_port_allocator.cc:925): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Gathered candidate: Cand[:1009828554:1:udp:2122260223:192.168.10.x:48944:local::0:NdQT:3Y+jOvlHqoT/UpD1wJbFv8ld:1:0:0]
[008:822][2934] (basic_port_allocator.cc:958): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port ready.
[008:822][2934] (stun_port.cc:441): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Starting STUN host lookup for stun.l.google.com:19302
[008:822][2936] (p2p_websocket_session.cpp:171): OnIceCandidate
[008:822][2934] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[008:822][2927] (websocket.cpp:282): Websocket::OnWrite this=2cc8d970 ec=Success
[008:828][2934] (port.cc:386): Received STUN BINDING request id=6874305a7269345267397a71 from unknown address 192.168.10.x:50485
[008:828][2934] (connection.cc:312): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|C--W|-|0|0|7926369428982087166|-]: Connection created
[008:828][2934] (p2p_transport_channel.cc:1142): Adding connection from peer reflexive candidate: Cand[:411250637:1:udp:1845501695:192.168.10.x:50485:prflx::0:biOA:5DsOfP+Bu3lrMDWQS0D8EyBF:0:999:0]
[008:829][2934] (connection.cc:723): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-W|-|0|0|7926369428982087166|-]: Sent STUN BINDING response, to=192.168.10.x:50485, id=6874305a7269345267397a71
[008:829][2934] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2
[008:829][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[008:829][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[008:829][2934] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping.
[008:829][2936] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1
[008:829][2936] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1
[008:829][2936] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[008:829][2934] (connection.cc:1167): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-W|-|1|0|7926369428982087166|-]: Sent STUN BINDING request, id=576467784a585475442f5157, use_candidate=0, nomination=0
[008:830][2934] (connection.cc:1082): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-I|-|1|0|7926369428982087166|-]: Received STUN BINDING response, id=576467784a585475442f5157, code=0, rtt=1, pings_since_last_response=576467784a585475442f5157
[008:830][2934] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0
[008:830][2934] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete
[008:830][2934] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[008:830][2934] (channel.cc:354): Network route changed for {mid: 0, media_type: video}
[008:830][2934] (channel.cc:354): Network route changed for {mid: 1, media_type: audio}
[008:830][2935] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ]
[008:830][2934] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1
[008:831][2934] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake
[008:831][2934] (srtp_transport.cc:365): The params in SRTP transport are reset.
[008:831][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete.
[008:831][2936] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2
[008:831][2936] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2
[008:831][2936] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected
[008:834][2934] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete.
[008:834][2934] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1.
[008:834][2935] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up
[008:834][2934] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0
[008:834][2983] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller
[008:834][2936] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable.
[008:834][2936] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2
[008:834][2983] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
[008:834][2983] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor
[008:834][2983] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor
[008:835][2983] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
[008:835][2983] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled
[008:835][2983] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps
[008:835][2983] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000
[008:835][2983] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5)
[008:835][2983] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)
[008:835][2983] (bitrate_allocator.cc:394): Current BWE 300000
Wrong JPEG library version: library is 62, caller expects 80
[008:844][2934] (srtp_transport.cc:310): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1
[008:844][2934] (channel.cc:540): Channel writable ({mid: 0, media_type: video}) for the first time
[008:844][2934] (channel.cc:540): Channel writable ({mid: 1, media_type: audio}) for the first time
[008:844][2934] (usrsctp_transport.cc:314): InitializeUsrSctp
[008:844][2935] (video_send_stream.cc:158): UpdateActiveSimulcastLayers: {1}
[008:844][2983] (bitrate_allocator.cc:523): UpdateAllocationLimits : total_requested_min_bitrate: 0 bps, total_requested_padding_bitrate: 0 bps, total_requested_max_bitrate: 10000 kbps
[008:844][2986] (video_stream_encoder.cc:2042): Video suspend state changed to: not suspended
[008:844][2935] (channel.cc:976): Changing video state, send=1 for {mid: 0, media_type: video}
[008:844][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
voluntas
commented
2 years ago
When start the momo: ./momo --no-audio-device --log-level 1 test On the newly installed Jetson Nano when try to connect with H264 or VP8 gave this error message:
Wrong JPEG library version: library is 62, caller expects 80
Not installed the libjpeg 6.2, just the libjpeg8 and libjpeg-turbo8.
[006:972][2747] (websocket.cpp:282): Websocket::OnWrite this=2902f970 ec=Success Wrong JPEG library version: library is 62, caller expects 80 [006:978][2754] (port.cc:386): Received STUN BINDING request id=6177354536476734786e6835 from unknown address 192.168.10.x:57747
[006:979][2754] (p2p_transport_channel.cc:1142): Adding connection from peer reflexive candidate: Cand[:3252266549:1:udp:1845501695:192.168.10.x:57747:prflx::0:ZSii:5hKQ7/1LClMzM1f62PfB0QmW:0:999:0] [006:979][2754] (connection.cc:723): Conn[7801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:1pPAcXyO:1:0:local:udp:192.168.10.x:46405->5beevxLi:1:1845501695:prflx:udp:192.168.10.x:57747|CR-W|-|0|0|792636942898208716G response, to=192.168.10.x:57747, id=6177354536476734786e6835
[006:979][2754] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2 [006:979][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:979][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:979][2754] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping. [006:979][2756] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1 [006:979][2756] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1 [006:979][2756] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[006:980][2754] (connection.cc:1167): Conn[7801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:1pPAcXyO:1:0:local:udp:192.168.10.x:46405->5beevxLi:1:1845501695:prflx:udp:192.168.10.x:57747|CR-W|-|1|0|79263694289820871NG request, id=2b6c51326f334539596c3465, use_candidate=0, nomination=0 [006:980][2754] (connection.cc:1082): Conn[7801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:1pPAcXyO:1:0:local:udp:192.168.10.x:46405->5beevxLi:1:1845501695:prflx:udp:192.168.10.x:57747|CR-I|-|1|0|79263694289820871INDING response, id=2b6c51326f334539596c3465, code=0, rtt=0, pings_since_last_response=2b6c51326f334539596c3465 [006:980][2754] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0 [006:981][2754] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete [006:981][2754] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[006:981][2754] (channel.cc:354): Network route changed for {mid: 0, media_type: video} [006:981][2754] (channel.cc:354): Network route changed for {mid: 1, media_type: audio} [006:981][2755] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ] [006:981][2754] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 0 [006:981][2754] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake [006:981][2754] (srtp_transport.cc:365): The params in SRTP transport are reset. [006:981][2754] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:981][2756] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2 [006:981][2756] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2 [006:981][2756] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected [006:985][2754] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete. [006:985][2754] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1. [006:985][2755] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up [006:985][2765] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller [006:985][2756] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable. [006:985][2756] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2 [006:985][2765] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3 [006:985][2765] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st [006:985][2765] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st [006:985][2765] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85 [006:986][2765] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled [006:986][2765] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps [006:986][2765] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000 [006:986][2765] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5) [006:986][2765] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [006:986][2765] (bitrate_allocator.cc:394): Current BWE 300000 [006:985][2754] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0 root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# clear root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# ./momo --no-audio-device --log-level 1 --no-google-stun test [000:000][2807] (v4l2_video_capturer.cpp:76): GetDeviceName(0): device_name=HD Pro Webcam C920, unique_name=usb-70090000.xusb-2.1 [000:003][2807] (v4l2_video_capturer.cpp:230): Video Capture enumerats supported image formats: [000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = YUYV, description = 'd9f262ac' } [000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = H264, description = 'd9f262ac' } [000:003][2807] (v4l2_video_capturer.cpp:232): { pixelformat = MJPG, description = 'd9f262ac' } [000:003][2807] (v4l2_video_capturer.cpp:248): We prefer format MJPG [000:162][2807] (jetson_v4l2_capturer.cpp:29): Get Capture [000:163][2814] (audio_device_buffer.cc:64): AudioDeviceBuffer::ctor [000:163][2814] (audio_device_impl.cc:136): current platform is Linux [000:163][2814] (audio_device_impl.cc:155): CreatePlatformSpecificObjects [000:163][2814] (audio_device_impl.cc:947): PlatformAudioLayer [000:163][2814] (audio_device_impl.cc:266): PulseAudio support is enabled. [000:163][2814] (audio_device_impl.cc:299): Dummy Audio APIs will be utilized. [000:163][2814] (audio_device_impl.cc:312): AttachAudioBuffer [000:251][2807] (audio_processing_impl.cc:278): Injected APM submodules: Echo control factory: 0 Echo detector: 0 Capture analyzer: 0 Capture post processor: 0 Render pre processor: 0 [000:251][2807] (webrtc_voice_engine.cc:268): WebRtcVoiceEngine::WebRtcVoiceEngine [000:251][2814] (webrtc_voice_engine.cc:290): WebRtcVoiceEngine::Init [000:252][2814] (audio_device_impl.cc:332): Init [000:252][2814] (audio_device_impl.cc:676): SetPlayoutDevice(0) [000:252][2814] (adm_helpers.cc:44): Unable to set playout device. [000:252][2814] (audio_device_impl.cc:851): RegisterAudioCallback [000:252][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_y_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, } [000:252][2814] (audio_device_impl.cc:867): BuiltInAECIsAvailable [000:252][2814] (audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform [000:252][2814] (audio_device_impl.cc:870): output: 0 [000:252][2814] (audio_device_impl.cc:883): BuiltInAGCIsAvailable [000:252][2814] (audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform [000:252][2814] (audio_device_impl.cc:886): output: 0 [000:252][2814] (audio_device_impl.cc:899): BuiltInNSIsAvailable [000:252][2814] (audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform [000:252][2814] (audio_device_impl.cc:902): output: 0 [000:252][2814] (webrtc_voice_engine.cc:495): Stereo swapping enabled? 0 [000:252][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200 [000:252][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0 [000:252][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0 [000:252][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0 [000:252][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0 [000:252][2814] (webrtc_voice_engine.cc:589): NS set to 1 [000:252][2814] (webrtc_voice_engine.cc:593): Typing detection is enabled? 1 [000:252][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }} [000:253][2814] (agc_manager_direct.cc:65): [agc] GetMinMicLevel [000:253][2814] (agc_manager_direct.cc:69): [agc] Using default min mic level: 12 [003:983][2807] (p2p_websocket_session.cpp:28): P2PWebsocketSession [003:984][2807] (p2p_websocket_session.cpp:37): Run [003:984][2807] (p2p_websocket_session.cpp:57): OnAccept: system:0 [006:214][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success [006:214][2807] (p2p_websocket_session.cpp:74): OnRead: system:0 [006:214][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"offer","sdp":"v=0\r\no=- 4331006092786853723 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=extmap-allow-mixed\r\na=msid-so 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r6 F0:38:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/exps-send-time\r\na=extmap:3 urn:3gpp:video-orientation\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:9 urn:isdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:102 H264/90000\r\na=r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtpllowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcr\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmartcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=rna=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\na=rtpmap:35 flexfec-03/90000\r\na=rtcp-fb:35 goog-remb\r\na=rtcp-fb:35 transport-cc\r\na=f0000000\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r\ F0:38:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrthdrext/abs-send-time\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpm=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000hone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=application 9 UDP/DTLS/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:L3Kg\r\na=ice-pwd:95aat8mx9uFiqVHibX8aqOyk\r\na=ice-options:trickle\r\na=fi8:1A:04:F3:D6:68:D6:3C:9A:87:67:85:86:82:68:2B:A6:B5:78:86:FB:EF:F5:66:3A:A6:75:68:41:53:63\r\na=setup:actpass\r\na=mid:2\r\na=sctp-port:5000\r\na=max-message-size:262144\r\n"} [006:215][2814] (rtc_event_log_impl.cc:43): Creating legacy encoder for RTC event log. [006:215][2814] (peer_connection_factory.cc:331): Using default network controller factory [006:215][2814] (bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive [006:216][2814] (cpu_info.cc:53): Available number of cores: 4 [006:216][2814] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85 [006:216][2814] (remote_bitrate_estimator_single_stream.cc:72): RemoteBitrateEstimatorSingleStream: Instantiating. [006:216][2814] (remote_estimator_proxy.cc:47): Maximum interval between transport feedback RTCP messages (ms): 250 [006:217][2813] (openssl_key_pair.cc:38): Making key pair [006:217][2815] (rtp_transmission_manager.cc:187): Adding video transceiver in response to a call to AddTrack. [006:218][2813] (openssl_key_pair.cc:91): Returning key pair [006:218][2813] (boringssl_certificate.cc:187): Making certificate for WebRTC [006:218][2813] (boringssl_certificate.cc:243): Returning certificate [006:219][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [006:219][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [006:219][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [006:219][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [006:219][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [006:219][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [006:219][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [006:219][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [006:219][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0 [006:219][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport 88003020 [006:219][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [006:219][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [006:219][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [006:219][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [006:219][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [006:219][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [006:219][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [006:220][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [006:220][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 1 transport 0 [006:220][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 1 transport 88005e90 [006:220][2813] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [006:220][2813] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [006:220][2813] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [006:220][2813] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [006:220][2813] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [006:220][2813] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [006:220][2813] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [006:220][2813] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [006:220][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 2 transport 0 [006:220][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 2 transport 88008610 [006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled [006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled [006:220][2813] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=L3Kg, renomination disabled [006:220][2814] (webrtc_video_engine.cc:643): CreateMediaChannel. Options: VideoOptions {} [006:220][2814] (channel.cc:137): Created channel: {mid: 0, media_type: video} [006:221][2813] (rtp_demuxer.cc:154): Added sink = 80077348 for criteria {mid: 0, rsid:, ssrcs: [], payload_types = []}
[006:221][2815] (sdp_offer_answer.cc:3250): Adding audio transceiver for MID=1 at i=1 in response to the remote description.
[006:221][2814] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jithandling: 0, }
[006:221][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_ji_handling: 0, }
[006:221][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[006:221][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[006:221][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[006:221][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[006:221][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[006:221][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[006:221][2814] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms:r_enable_rtx_handling: 0, }
[006:221][2814] (channel.cc:137): Created channel: {mid: 1, media_type: audio}
[006:221][2813] (rtp_demuxer.cc:154): Added sink = 80077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[006:222][2815] (sdp_offer_answer.cc:3371): Creating data channel, mid=2
[006:222][2813] (peer_connection.cc:2382): Setting up data channel transport for mid=2
[006:222][2815] (sdp_offer_answer.cc:2459): Session: 8528378070752557626 Old state: stable New state: have-remote-offer
[006:222][2814] (webrtc_video_engine.cc:1573): ResetUnsignaledRecvStream.
[006:222][2814] (webrtc_voice_engine.cc:2047): ResetUnsignaledRecvStream.
[006:222][2814] (channel.cc:1082): Setting remote video description for {mid: 0, media_type: video}
[006:222][2814] (webrtc_video_engine.cc:884): SetSendParameters: {codecs: [VideoCodec[102:H264], VideoCodec[121:rtx], VideoCodec[127:H264], VideoCodec[120:rtx], VideoCodec[125:H264], VideoCodec[107:rtx], VideoCodec[109:rtx], VideoCodec[124:H264], VideoCodec[119:rtx], VideoCodec[123:H264], VideoCodec[118:rtx], VideoCodec[114:red], VideoCodec[115:rtx], VideoCodec[116:ulpfec], VideoCodec[35:flexfec-03]], conference_mode: no, extensparams:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timi://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repd: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[102:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[127:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[125:H264]
[006:222][2814] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[108:H264]
[006:222][2814] (webrtc_video_engine.cc:1041): SetFeedbackParameters on all the receive streams because the send codec or RTCP mode has changed.
[006:223][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:223][2814] (channel.cc:889): Setting remote voice description for {mid: 1, media_type: audio}
[006:223][2814] (webrtc_voice_engine.cc:1416): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:ephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {rg/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdesd, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 1, options: AudioOptions {}}
[006:223][2814] (webrtc_voice_engine.cc:1840): Recreate all the receive streams because the send codec has changed.
[006:223][2814] (webrtc_voice_engine.cc:2335): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[006:223][2814] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {}
[006:223][2814] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_ji_handling: 0, }
[006:223][2814] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[006:223][2814] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[006:223][2814] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[006:223][2814] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[006:223][2814] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[006:223][2814] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_ampfixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_d_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }} enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, _dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[006:223][2814] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms:r_enable_rtx_handling: 0, }
[006:223][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:223][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:224][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:224][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51664 typ host generation 0st 999","sdpMid":"0","sdpMLineIndex":0}}
[006:224][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51664 typ host genetwork-cost 999
[006:224][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:224][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:224][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51666 typ host generation 0st 999","sdpMid":"1","sdpMLineIndex":1}}
[006:224][2813] (peer_connection.cc:2587): 0 is not ready to use the remote candidate because the local or remote description is not set.
[006:225][2815] (rtc_connection.cpp:169): Created session description : v=0
o=- 8528378070752557626 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
m=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 2334982981 921515023
a=ssrc:2334982981 cname:9z2Fp9qEvpL87sim
a=ssrc:2334982981 msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:2334982981 mslabel:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
a=ssrc:2334982981 label:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:921515023 cname:9z2Fp9qEvpL87sim
a=ssrc:921515023 msid:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
a=ssrc:921515023 mslabel:uuQyAmicHdz3pQwRDmmYK9xjItzSQkPj
a=ssrc:921515023 label:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=inactive
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:EnjI
a=ice-pwd:B+h3GCWvL/6K3FyW0S7w92Vd
a=ice-options:trickle
a=fingerprint:sha-256 AF:CE:EA:11:AA:33:FD:81:84:49:36:0B:BB:2E:79:2C:A8:76:AC:11:E0:EF:CC:CD:74:91:B2:DE:1D:07:53:E8
a=setup:active
a=mid:2
a=sctp-port:5000
a=max-message-size:262144
[006:225][2813] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0 [006:225][2813] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport 88003020 [006:225][2813] (p2p_transport_channel.cc:517): Set ICE ufrag: EnjI pwd: B+h3GCWvL/6K3FyW0S7w92Vd on transport 0 [006:225][2813] (dtls_transport.cc:367): DtlsTransport[0|1|__]: DTLS setup complete. [006:225][2813] (rtp_demuxer.cc:250): Removed sink = 80077a48 bindings [006:225][2813] (rtp_demuxer.cc:154): Added sink = 80077a48 for criteria {mid: 1, rsid:, ssrcs: [], payload_types = []}
[006:225][2815] (rtp_transceiver.cc:313): Changing transceiver (MID=0) current direction from to kSendOnly.
[006:226][2815] (rtp_transceiver.cc:313): Changing transceiver (MID=1) current direction from to kInactive.
[006:226][2815] (sdp_offer_answer.cc:2459): Session: 8528378070752557626 Old state: have-remote-offer New state: stable
[006:226][2814] (channel.cc:512): Channel enabled: {mid: 0, media_type: video}
[006:226][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:226][2814] (channel.cc:512): Channel enabled: {mid: 1, media_type: audio}
[006:226][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:226][2814] (channel.cc:991): Setting local video description for {mid: 0, media_type: video}
[006:226][2814] (webrtc_video_engine.cc:1341): AddSendStream: {id:IfdObIJ8OUlr5Gd4yrp5wTDS9VcM6LoX;ssrcs:[2334982981,921515023];ssrc_groups:{semantics:FID;ssrcs:[2334982981,921515023]};cname:9z2Fp9qEvpL87sim;stream_iYK9xjItzSQkPj;}
[006:226][2814] (webrtc_video_engine.cc:2249): RecreateWebRtcStream (send) because of SetCodec.
[006:226][2814] (paced_sender.cc:187): ProcessThreadAttached 0x80071d60
[006:227][2814] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3
[006:227][2814] (video_stream_encoder.cc:2189): Automatic animation detection experiment is disabled.
[006:227][2824] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[006:227][2824] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[006:227][2824] (video_send_stream_impl.cc:245): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2334982981], rids: [], mid: '0', rtcp_mode: RtcpMode::kReducedSize, mextmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://wnts/rtp-hdrext/color-space, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.orgt/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}, {uri: urn:ietf:params:rtp-hdd, id: 10}], lntf: {enabled: false}, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 116, red_payload_type: 114, red_rtx_payload_type: 115}, payload_name: H264, payload_type: 102, raw_payload: false, flex, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [921515023], payload_type: 121}, c_name: 9z2Fp9qEvpL87sim}, rtcp_report_interval_ms: 1000, send_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, susp off}
[006:227][2827] (video_stream_encoder.cc:798): SetStartBitrate 300000
[006:227][2824] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[006:227][2827] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[006:227][2814] (webrtc_video_engine.cc:1381): SetLocalSsrc on all the receive streams because we added a send stream.
[006:227][2814] (channel.cc:671): Add send stream ssrc: 2334982981 into {mid: 0, media_type: video}
[006:227][2814] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[006:228][2824] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[006:228][2814] (channel.cc:829): Setting local voice description for {mid: 1, media_type: audio}
[006:228][2814] (webrtc_voice_engine.cc:1462): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:ephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {rg/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdesd, id: 11}]}
[006:228][2814] (webrtc_voice_engine.cc:1630): Setting receive voice codecs.
[006:228][2814] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[006:228][2813] (p2p_transport_channel.cc:1229): Asynchronously resolving ICE candidate hostname 7fe0e395-c1e7-463a-b210-2344e6824c6e.local
[006:228][2814] (webrtc_video_engine.cc:1300): SetVideoSend (ssrc= 2334982981, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))
[006:228][2827] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[006:228][2813] (basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
[006:229][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51666 typ host genetwork-cost 999
[006:229][2815] (peer_connection.cc:1805): Changing IceConnectionState 0 => 1
[006:229][2813] (basic_port_allocator.cc:111): Filtered out ignored networks:
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[006:229][2813] (basic_port_allocator.cc:861): Network manager has started
[006:229][2813] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[006:229][2813] (basic_port_allocator.cc:111): Filtered out ignored networks:
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[006:229][2813] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[006:229][2813] (basic_port_allocator.cc:776): Allocate ports on 1 networks
[006:229][2813] (basic_port_allocator.cc:1361): Net[eth0:192.168.10.x/24:Ethernet:id=1]: Allocation Phase=Udp
[006:229][2813] (port.cc:187): Port[88005510::1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port created with network cost 0
[006:229][2813] (basic_port_allocator.cc:885): Adding allocated port for 0
[006:229][2813] (basic_port_allocator.cc:907): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Added port to allocator
[006:229][2807] (websocket.cpp:282): Websocket::OnWrite this=18efa970 ec=Success
[006:230][2813] (basic_port_allocator.cc:925): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Gathered candidate: Cand[:1009828554:1:udp:2122260223:192.168.10.x:43907:local::0:EnjI:B+h3GCWvL/6K3F
[006:230][2813] (basic_port_allocator.cc:958): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port ready.
[006:230][2807] (websocket.cpp:225): Websocket::OnRead this=18efa970 ec=Success
[006:230][2807] (p2p_websocket_session.cpp:74): OnRead: system:0
[006:230][2807] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51668 typ host generation 0st 999","sdpMid":"2","sdpMLineIndex":2}}
[006:230][2813] (basic_port_allocator.cc:1069): Port[88005510:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port completed gathering candidates.
[006:230][2815] (p2p_websocket_session.cpp:171): OnIceCandidate
[006:230][2815] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 7fe0e395-c1e7-463a-b210-2344e6824c6e.local 51668 typ host genetwork-cost 999
[006:230][2813] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[006:230][2807] (websocket.cpp:282): Websocket::OnWrite this=18efa970 ec=Success
[006:236][2813] (port.cc:386): Received STUN BINDING request id=2f66747958694e6350765744 from unknown address 192.168.10.x:51664
[006:236][2813] (p2p_transport_channel.cc:1142): Adding connection from peer reflexive candidate: Cand[:1281963569:1:udp:1845501695:192.168.10.x:51664:prflx::0:L3Kg:95aat8mx9uFiqVHibX8aqOyk:0:999:0] [006:237][2813] (connection.cc:723): Conn[8801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:BIp3Al1I:1:0:local:udp:192.168.10.x:43907->0uZrj/Rg:1:1845501695:prflx:udp:192.168.10.x:51664|CR-W|-|0|0|792636942898208716G response, to=192.168.10.x:51664, id=2f66747958694e6350765744
[006:237][2813] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2 [006:237][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:237][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:237][2813] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping. [006:237][2815] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1 [006:237][2815] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1 [006:237][2815] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[006:238][2813] (connection.cc:1167): Conn[8801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:BIp3Al1I:1:0:local:udp:192.168.10.x:43907->0uZrj/Rg:1:1845501695:prflx:udp:192.168.10.x:51664|CR-W|-|1|0|79263694289820871NG request, id=475a65536561683349713235, use_candidate=0, nomination=0 [006:238][2813] (connection.cc:1082): Conn[8801b860:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:BIp3Al1I:1:0:local:udp:192.168.10.x:43907->0uZrj/Rg:1:1845501695:prflx:udp:192.168.10.x:51664|CR-I|-|1|0|79263694289820871INDING response, id=475a65536561683349713235, code=0, rtt=1, pings_since_last_response=475a65536561683349713235 [006:238][2813] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0 [006:238][2813] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete [006:238][2813] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[006:238][2813] (channel.cc:354): Network route changed for {mid: 0, media_type: video} [006:238][2813] (channel.cc:354): Network route changed for {mid: 1, media_type: audio} [006:238][2814] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ] [006:238][2813] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [006:239][2813] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake [006:239][2813] (srtp_transport.cc:365): The params in SRTP transport are reset. [006:239][2813] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [006:239][2815] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2 [006:239][2815] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2 [006:239][2815] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected Wrong JPEG library version: library is 62, caller expects 80 [006:243][2813] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete. [006:243][2813] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1. [006:243][2814] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up [006:243][2824] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller [006:243][2813] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0 [006:243][2815] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable. [006:243][2815] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2 [006:243][2824] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percentgroup ID: 3 [006:243][2824] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st [006:244][2824] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network st [006:244][2824] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85 [006:244][2824] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled [006:244][2824] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps [006:244][2824] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000 [006:244][2824] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5) [006:244][2824] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [006:244][2824] (bitrate_allocator.cc:394): Current BWE 300000 root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# root@robi-jetsonout:~/momo/momo-2021.4.3_ubuntu-18.04_armv8_jetson_nano# ./momo --no-audio-device --log-level 1 test [000:000][2927] (v4l2_video_capturer.cpp:76): GetDeviceName(0): device_name=HD Pro Webcam C920, unique_name=usb-70090000.xusb-2.1 [000:003][2927] (v4l2_video_capturer.cpp:230): Video Capture enumerats supported image formats: [000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = YUYV, description = 'f5cf253c' } [000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = H264, description = 'f5cf253c' } [000:003][2927] (v4l2_video_capturer.cpp:232): { pixelformat = MJPG, description = 'f5cf253c' } [000:003][2927] (v4l2_video_capturer.cpp:248): We prefer format MJPG [000:162][2927] (jetson_v4l2_capturer.cpp:29): Get Capture [000:163][2935] (audio_device_buffer.cc:64): AudioDeviceBuffer::ctor [000:163][2935] (audio_device_impl.cc:136): current platform is Linux [000:163][2935] (audio_device_impl.cc:155): CreatePlatformSpecificObjects [000:163][2935] (audio_device_impl.cc:947): PlatformAudioLayer [000:163][2935] (audio_device_impl.cc:266): PulseAudio support is enabled. [000:163][2935] (audio_device_impl.cc:299): Dummy Audio APIs will be utilized. [000:163][2935] (audio_device_impl.cc:312): AttachAudioBuffer [000:253][2927] (audio_processing_impl.cc:278): Injected APM submodules: Echo control factory: 0 Echo detector: 0 Capture analyzer: 0 Capture post processor: 0 Render pre processor: 0 [000:253][2927] (webrtc_voice_engine.cc:268): WebRtcVoiceEngine::WebRtcVoiceEngine [000:254][2935] (webrtc_voice_engine.cc:290): WebRtcVoiceEngine::Init [000:254][2935] (audio_device_impl.cc:332): Init [000:254][2935] (audio_device_impl.cc:676): SetPlayoutDevice(0) [000:254][2935] (adm_helpers.cc:44): Unable to set playout device. [000:255][2935] (audio_device_impl.cc:851): RegisterAudioCallback [000:255][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, } [000:255][2935] (audio_device_impl.cc:867): BuiltInAECIsAvailable [000:255][2935] (audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform [000:255][2935] (audio_device_impl.cc:870): output: 0 [000:255][2935] (audio_device_impl.cc:883): BuiltInAGCIsAvailable [000:255][2935] (audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform [000:255][2935] (audio_device_impl.cc:886): output: 0 [000:255][2935] (audio_device_impl.cc:899): BuiltInNSIsAvailable [000:255][2935] (audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform [000:255][2935] (audio_device_impl.cc:902): output: 0 [000:255][2935] (webrtc_voice_engine.cc:495): Stereo swapping enabled? 0 [000:255][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200 [000:255][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0 [000:255][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0 [000:255][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0 [000:255][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0 [000:255][2935] (webrtc_voice_engine.cc:589): NS set to 1 [000:255][2935] (webrtc_voice_engine.cc:593): Typing detection is enabled? 1 [000:255][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }} [000:256][2935] (agc_manager_direct.cc:65): [agc] GetMinMicLevel [000:256][2935] (agc_manager_direct.cc:69): [agc] Using default min mic level: 12 [006:928][2927] (p2p_websocket_session.cpp:28): P2PWebsocketSession [006:928][2927] (p2p_websocket_session.cpp:37): Run [006:928][2927] (p2p_websocket_session.cpp:57): OnAccept: system:0 [008:802][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success [008:803][2927] (p2p_websocket_session.cpp:74): OnRead: system:0 [008:803][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"offer","sdp":"v=0\r\no=- 4154535124576789392 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=extmap-allow-mixed\r\na=msid-semantic: WMS\r\nm=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 urn:3gpp:video-orientation\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=fmtp:121 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcp-fb:124 transport-cc\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmap:123 H264/90000\r\na=rtcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=rtpmap:118 rtx/90000\r\na=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\na=rtpmap:35 flexfec-03/90000\r\na=rtcp-fb:35 goog-remb\r\na=rtcp-fb:35 transport-cc\r\na=fmtp:35 repair-window=10000000\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=application 9 UDP/DTLS/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:biOA\r\na=ice-pwd:5DsOfP+Bu3lrMDWQS0D8EyBF\r\na=ice-options:trickle\r\na=fingerprint:sha-256 E4:55:4F:14:F6:38:0F:DE:B5:11:56:A0:A7:7E:D6:DE:25:5A:43:68:20:3C:3B:D9:0A:E2:C0:6B:A0:0D:46:5D\r\na=setup:actpass\r\na=mid:2\r\na=sctp-port:5000\r\na=max-message-size:262144\r\n"} [008:803][2935] (rtc_event_log_impl.cc:43): Creating legacy encoder for RTC event log. [008:803][2935] (peer_connection_factory.cc:331): Using default network controller factory [008:804][2935] (bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive [008:804][2935] (cpu_info.cc:53): Available number of cores: 4 [008:804][2935] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85 [008:804][2935] (remote_bitrate_estimator_single_stream.cc:72): RemoteBitrateEstimatorSingleStream: Instantiating. [008:804][2935] (remote_estimator_proxy.cc:47): Maximum interval between transport feedback RTCP messages (ms): 250 [008:805][2934] (openssl_key_pair.cc:38): Making key pair [008:806][2936] (rtp_transmission_manager.cc:187): Adding video transceiver in response to a call to AddTrack. [008:806][2934] (openssl_key_pair.cc:91): Returning key pair [008:806][2934] (boringssl_certificate.cc:187): Making certificate for WebRTC [008:807][2934] (boringssl_certificate.cc:243): Returning certificate [008:807][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [008:807][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0 [008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport a4003090 [008:808][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [008:808][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 1 transport 0 [008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 1 transport a4005f00 [008:808][2934] (p2p_transport_channel.cc:583): Set backup connection ping interval to 25000 milliseconds. [008:808][2934] (p2p_transport_channel.cc:592): Set ICE receiving timeout to 2500 milliseconds [008:808][2934] (p2p_transport_channel.cc:599): Set ping most likely connection to 0 [008:808][2934] (p2p_transport_channel.cc:606): Set stable_writable_connection_ping_interval to 2500 [008:808][2934] (p2p_transport_channel.cc:619): Set presume writable when fully relayed to 0 [008:808][2934] (p2p_transport_channel.cc:637): Set regather_on_failed_networks_interval to 300000 [008:808][2934] (p2p_transport_channel.cc:644): Set receiving_switching_delay to 1000 [008:808][2934] (jsep_transport_controller.cc:1094): Creating DtlsSrtpTransport. [008:808][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 2 transport 0 [008:808][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 2 transport a4008680 [008:808][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled [008:808][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled [008:809][2934] (p2p_transport_channel.cc:528): Received remote ICE parameters: ufrag=biOA, renomination disabled [008:809][2935] (webrtc_video_engine.cc:643): CreateMediaChannel. Options: VideoOptions {} [008:809][2935] (channel.cc:137): Created channel: {mid: 0, media_type: video} [008:809][2934] (rtp_demuxer.cc:154): Added sink = 9c077348 for criteria {mid: 0, rsid:, ssrcs: [], payload_types = []}
[008:809][2936] (sdp_offer_answer.cc:3250): Adding audio transceiver for MID=1 at i=1 in response to the remote description.
[008:810][2935] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[008:810][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[008:810][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[008:810][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[008:810][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[008:810][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[008:810][2935] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:810][2935] (channel.cc:137): Created channel: {mid: 1, media_type: audio}
[008:810][2934] (rtp_demuxer.cc:154): Added sink = 9c077a48 for criteria {mid: 1, rsid: , ssrcs: [], payload_types = []}
[008:810][2936] (sdp_offer_answer.cc:3371): Creating data channel, mid=2
[008:810][2934] (peer_connection.cc:2382): Setting up data channel transport for mid=2
[008:810][2936] (sdp_offer_answer.cc:2459): Session: 481305330888516940 Old state: stable New state: have-remote-offer
[008:810][2935] (webrtc_video_engine.cc:1573): ResetUnsignaledRecvStream.
[008:810][2935] (webrtc_voice_engine.cc:2047): ResetUnsignaledRecvStream.
[008:810][2935] (channel.cc:1082): Setting remote video description for {mid: 0, media_type: video}
[008:811][2935] (webrtc_video_engine.cc:884): SetSendParameters: {codecs: [VideoCodec[102:H264], VideoCodec[121:rtx], VideoCodec[127:H264], VideoCodec[120:rtx], VideoCodec[125:H264], VideoCodec[107:rtx], VideoCodec[108:H264], VideoCodec[109:rtx], VideoCodec[124:H264], VideoCodec[119:rtx], VideoCodec[123:H264], VideoCodec[118:rtx], VideoCodec[114:red], VideoCodec[115:rtx], VideoCodec[116:ulpfec], VideoCodec[35:flexfec-03]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[102:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[127:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[125:H264]
[008:811][2935] (webrtc_video_engine.cc:892): Negotiated codec: VideoCodec[108:H264]
[008:811][2935] (webrtc_video_engine.cc:1041): SetFeedbackParameters on all the receive streams because the send codec or RTCP mode has changed.
[008:811][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:811][2935] (channel.cc:889): Setting remote voice description for {mid: 1, media_type: audio}
[008:811][2935] (webrtc_voice_engine.cc:1416): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 1, options: AudioOptions {}}
[008:811][2935] (webrtc_voice_engine.cc:1840): Recreate all the receive streams because the send codec has changed.
[008:811][2935] (webrtc_voice_engine.cc:2335): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
[008:811][2935] (webrtc_voice_engine.cc:1602): Setting voice channel options: AudioOptions {}
[008:811][2935] (webrtc_voice_engine.cc:386): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:811][2935] (webrtc_voice_engine.cc:500): NetEq capacity is 200
[008:811][2935] (webrtc_voice_engine.cc:506): NetEq fast mode? 0
[008:811][2935] (webrtc_voice_engine.cc:512): NetEq minimum delay is 0
[008:811][2935] (webrtc_voice_engine.cc:518): NetEq handle reordered packets? 0
[008:811][2935] (webrtc_voice_engine.cc:538): Experimental ns is enabled? 0
[008:811][2935] (audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
[008:811][2935] (webrtc_voice_engine.cc:1620): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
[008:811][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:812][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:812][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:812][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50485 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"0","sdpMLineIndex":0}}
[008:813][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50485 typ host generation 0 ufrag biOA network-cost 999
[008:813][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:813][2934] (peer_connection.cc:2587): 0 is not ready to use the remote candidate because the local or remote description is not set.
[008:813][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:813][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50487 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"1","sdpMLineIndex":1}}
[008:813][2936] (rtc_connection.cpp:169): Created session description : v=0
o=- 481305330888516940 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS /l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
m=video 9 UDP/TLS/RTP/SAVPF 102 121 127 120 125 107 108 109 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 142137631 2598199993
a=ssrc:142137631 cname:kz/tKo60LC6c+QVq
a=ssrc:142137631 msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:142137631 mslabel:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
a=ssrc:142137631 label:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:2598199993 cname:kz/tKo60LC6c+QVq
a=ssrc:2598199993 msid:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
a=ssrc:2598199993 mslabel:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX
a=ssrc:2598199993 label:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=inactive
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:NdQT
a=ice-pwd:3Y+jOvlHqoT/UpD1wJbFv8ld
a=ice-options:trickle
a=fingerprint:sha-256 E5:B3:5C:4E:D1:7E:DB:DB:FF:C6:63:60:59:FD:47:6B:F6:98:7E:9D:87:3C:7D:AE:07:86:4C:B6:E3:B9:D3:9D
a=setup:active
a=mid:2
a=sctp-port:5000
a=max-message-size:262144
[008:816][2934] (dtls_srtp_transport.cc:62): Setting RTCP Transport on 0 transport 0 [008:816][2934] (dtls_srtp_transport.cc:67): Setting RTP Transport on 0 transport a4003090 [008:816][2934] (p2p_transport_channel.cc:517): Set ICE ufrag: NdQT pwd: 3Y+jOvlHqoT/UpD1wJbFv8ld on transport 0 [008:816][2934] (dtls_transport.cc:367): DtlsTransport[0|1|__]: DTLS setup complete. [008:816][2934] (rtp_demuxer.cc:250): Removed sink = 9c077a48 bindings [008:816][2934] (rtp_demuxer.cc:154): Added sink = 9c077a48 for criteria {mid: 1, rsid:, ssrcs: [], payload_types = []}
[008:816][2936] (rtp_transceiver.cc:313): Changing transceiver (MID=0) current direction from to kSendOnly.
[008:816][2936] (rtp_transceiver.cc:313): Changing transceiver (MID=1) current direction from to kInactive.
[008:816][2936] (sdp_offer_answer.cc:2459): Session: 481305330888516940 Old state: have-remote-offer New state: stable
[008:816][2935] (channel.cc:512): Channel enabled: {mid: 0, media_type: video}
[008:817][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:817][2935] (channel.cc:512): Channel enabled: {mid: 1, media_type: audio}
[008:817][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:817][2935] (channel.cc:991): Setting local video description for {mid: 0, media_type: video}
[008:817][2935] (webrtc_video_engine.cc:1341): AddSendStream: {id:q8O6/KaSJuDzaJUN3BDbRtmFJI0o1Nbn;ssrcs:[142137631,2598199993];ssrc_groups:{semantics:FID;ssrcs:[142137631,2598199993]};cname:kz/tKo60LC6c+QVq;stream_ids:/l9FtwnVLifhpJNnuwjtQgJRFMJat4EX;}
[008:817][2935] (webrtc_video_engine.cc:2249): RecreateWebRtcStream (send) because of SetCodec.
[008:817][2935] (paced_sender.cc:187): ProcessThreadAttached 0x9c071d60
[008:817][2935] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
[008:817][2935] (video_stream_encoder.cc:2189): Automatic animation detection experiment is disabled.
[008:818][2983] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[008:818][2983] (rtp_video_sender.cc:106): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
[008:818][2983] (video_send_stream_impl.cc:245): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [142137631], rids: [], mid: '0', rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}], lntf: {enabled: false}, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 116, red_payload_type: 114, red_rtx_payload_type: 115}, payload_name: H264, payload_type: 102, raw_payload: false, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2598199993], payload_type: 121}, c_name: kz/tKo60LC6c+QVq}, rtcp_report_interval_ms: 1000, send_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
[008:818][2986] (video_stream_encoder.cc:798): SetStartBitrate 300000
[008:818][2983] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[008:818][2986] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[008:818][2935] (webrtc_video_engine.cc:1381): SetLocalSsrc on all the receive streams because we added a send stream.
[008:818][2935] (channel.cc:671): Add send stream ssrc: 142137631 into {mid: 0, media_type: video}
[008:819][2935] (channel.cc:976): Changing video state, send=0 for {mid: 0, media_type: video}
[008:819][2983] (video_send_stream_impl.cc:404): VideoSendStreamImpl::Stop
[008:819][2935] (channel.cc:829): Setting local voice description for {mid: 1, media_type: audio}
[008:819][2935] (webrtc_voice_engine.cc:1462): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}]}
[008:819][2935] (webrtc_voice_engine.cc:1630): Setting receive voice codecs.
[008:819][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}
[008:819][2934] (p2p_transport_channel.cc:1229): Asynchronously resolving ICE candidate hostname 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local
[008:819][2935] (webrtc_video_engine.cc:1300): SetVideoSend (ssrc= 142137631, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))
[008:819][2986] (video_stream_encoder.cc:813): ConfigureEncoder requested.
[008:821][2934] (basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
[008:821][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50487 typ host generation 0 ufrag biOA network-cost 999
[008:821][2936] (peer_connection.cc:1805): Changing IceConnectionState 0 => 1
[008:821][2927] (websocket.cpp:282): Websocket::OnWrite this=2cc8d970 ec=Success
[008:822][2934] (basic_port_allocator.cc:111): Filtered out ignored networks:
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[008:822][2927] (websocket.cpp:225): Websocket::OnRead this=2cc8d970 ec=Success
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[008:822][2927] (p2p_websocket_session.cpp:74): OnRead: system:0
[008:822][2934] (basic_port_allocator.cc:861): Network manager has started
[008:822][2927] (p2p_websocket_session.cpp:91): OnRead: recv_string={"type":"candidate","ice":{"candidate":"candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50489 typ host generation 0 ufrag biOA network-cost 999","sdpMid":"2","sdpMLineIndex":2}}
[008:822][2934] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[008:822][2934] (basic_port_allocator.cc:111): Filtered out ignored networks:
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
[008:822][2934] (basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
[008:822][2934] (basic_port_allocator.cc:776): Allocate ports on 1 networks
[008:822][2934] (basic_port_allocator.cc:1361): Net[eth0:192.168.10.x/24:Ethernet:id=1]: Allocation Phase=Udp
[008:822][2934] (port.cc:187): Port[a4005580::1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port created with network cost 0
[008:822][2934] (basic_port_allocator.cc:1433): AllocationSequence: UDPPort will be handling the STUN candidate generation.
[008:822][2936] (rtc_connection.cpp:215): operator() Failed to apply the received candidate. type=NONE message= sdp=candidate:4078457168 1 udp 2113937151 6c0230b5-4026-4e42-8a5b-a299fe1c8eb8.local 50489 typ host generation 0 ufrag biOA network-cost 999
[008:822][2934] (basic_port_allocator.cc:885): Adding allocated port for 0
[008:822][2934] (basic_port_allocator.cc:907): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Added port to allocator
[008:822][2934] (basic_port_allocator.cc:925): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Gathered candidate: Cand[:1009828554:1:udp:2122260223:192.168.10.x:48944:local::0:NdQT:3Y+jOvlHqoT/UpD1wJbFv8ld:1:0:0]
[008:822][2934] (basic_port_allocator.cc:958): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Port ready.
[008:822][2934] (stun_port.cc:441): Port[a4005580:0:1:0:local:Net[eth0:192.168.10.x/24:Ethernet:id=1]]: Starting STUN host lookup for stun.l.google.com:19302
[008:822][2936] (p2p_websocket_session.cpp:171): OnIceCandidate
[008:822][2934] (jsep_transport_controller.cc:293): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
[008:822][2927] (websocket.cpp:282): Websocket::OnWrite this=2cc8d970 ec=Success
[008:828][2934] (port.cc:386): Received STUN BINDING request id=6874305a7269345267397a71 from unknown address 192.168.10.x:50485
[008:828][2934] (connection.cc:312): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|C--W|-|0|0|7926369428982087166|-]: Connection created
[008:828][2934] (p2p_transport_channel.cc:1142): Adding connection from peer reflexive candidate: Cand[:411250637:1:udp:1845501695:192.168.10.x:50485:prflx::0:biOA:5DsOfP+Bu3lrMDWQS0D8EyBF:0:999:0]
[008:829][2934] (connection.cc:723): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-W|-|0|0|7926369428982087166|-]: Sent STUN BINDING response, to=192.168.10.x:50485, id=6874305a7269345267397a71
[008:829][2934] (p2p_transportchannel.cc:1917): Channel[0|1|R]: Transport channel state changed from 0 to 2 [008:829][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [008:829][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [008:829][2934] (p2p_transportchannel.cc:1697): Channel[0|1|R]: Have a pingable connection for the first time; starting to ping. [008:829][2936] (peer_connection.cc:1824): Changing standardized IceConnectionState 0 => 1 [008:829][2936] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :1 [008:829][2936] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state unknown -> checking
[008:829][2934] (connection.cc:1167): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-W|-|1|0|7926369428982087166|-]: Sent STUN BINDING request, id=576467784a585475442f5157, use_candidate=0, nomination=0 [008:830][2934] (connection.cc:1082): Conn[a401bab0:0:Net[eth0:192.168.10.x/24:Ethernet:id=1]:hthNM5+l:1:0:local:udp:192.168.10.x:48944->nI2s+6d2:1:1845501695:prflx:udp:192.168.10.x:50485|CR-I|-|1|0|7926369428982087166|-]: Received STUN BINDING response, id=576467784a585475442f5157, code=0, rtt=1, pings_since_last_response=576467784a585475442f5157 [008:830][2934] (basic_port_allocator.cc:1144): All candidates gathered for 0:1:0 [008:830][2934] (p2p_transport_channel.cc:1004): P2PTransportChannel: 0, component 1 gathering complete [008:830][2934] (p2p_transport_channel.cc:328): Switching selected connection due to: candidate pair state changed
[008:830][2934] (channel.cc:354): Network route changed for {mid: 0, media_type: video} [008:830][2934] (channel.cc:354): Network route changed for {mid: 1, media_type: audio} [008:830][2935] (rtp_transport_controller_send.cc:301): Network route changed on transport 0: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 32/0 Wildcard turn: 0 ] packet_overhead_bytes: 28 ] [008:830][2934] (dtls_transport.cc:818): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [008:831][2934] (dtls_transport.cc:723): DtlsTransport[0|1|]: DtlsTransport: Started DTLS handshake [008:831][2934] (srtp_transport.cc:365): The params in SRTP transport are reset. [008:831][2934] (jsep_transport_controller.cc:1272): 0 Transport 1 state changed. Check if state is complete. [008:831][2936] (peer_connection.cc:1824): Changing standardized IceConnectionState 1 => 2 [008:831][2936] (peer_connection_observer.cpp:22): OnStandardizedIceConnectionChange :2 [008:831][2936] (p2p_websocket_session.cpp:161): OnIceConnectionStateChange rtc_state checking -> connected [008:834][2934] (dtls_transport.cc:651): DtlsTransport[0|1|]: DTLS handshake complete. [008:834][2934] (jsep_transport_controller.cc:1210): Transport 0 writability changed to 1. [008:834][2935] (call.cc:1245): UpdateAggregateNetworkState: aggregate_state change to up [008:834][2934] (dtls_srtp_transport.cc:218): Extracting keys from transport: 0 [008:834][2983] (rtp_transport_controller_send.cc:603): Creating fallback congestion controller [008:834][2936] (peer_connection.cc:2260): Changing to ICE connected state because all transports are writable. [008:834][2936] (peer_connection.cc:1805): Changing IceConnectionState 1 => 2 [008:834][2983] (alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3 [008:834][2983] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor [008:834][2983] (trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor [008:835][2983] (aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85 [008:835][2983] (delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled [008:835][2983] (delay_based_bwe.cc:357): BWE Setting start bitrate to: 300 kbps [008:835][2983] (probe_controller.cc:280): Measured bitrate: 300000 Minimum to probe further: 1260000 [008:835][2983] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1688:5) [008:835][2983] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [008:835][2983] (bitrate_allocator.cc:394): Current BWE 300000 Wrong JPEG library version: library is 62, caller expects 80 [008:844][2934] (srtp_transport.cc:310): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [008:844][2934] (channel.cc:540): Channel writable ({mid: 0, media_type: video}) for the first time [008:844][2934] (channel.cc:540): Channel writable ({mid: 1, media_type: audio}) for the first time [008:844][2934] (usrsctp_transport.cc:314): InitializeUsrSctp [008:844][2935] (video_send_stream.cc:158): UpdateActiveSimulcastLayers: {1} [008:844][2983] (bitrate_allocator.cc:523): UpdateAllocationLimits : total_requested_min_bitrate: 0 bps, total_requested_padding_bitrate: 0 bps, total_requested_max_bitrate: 10000 kbps [008:844][2986] (video_stream_encoder.cc:2042): Video suspend state changed to: not suspended [008:844][2935] (channel.cc:976): Changing video state, send=1 for {mid: 0, media_type: video} [008:844][2935] (channel.cc:820): Changing voice state, recv=0 send=0 for {mid: 1, media_type: audio}