Connection through a TURN server (iOS)
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Anton Tutarinov
May 17, 2017, 3:43:55 AM5/17/17
to discuss-webrtc
Hello,
I have a trouble with iOS <-> browser(Chrome) and iOS <-> iOS connections through a turn server (webrtc lib revision 76d9c9c3826f3c3622d3e8869fa54f7b39e70427). I used RTCIceTransportPolicyRelay as an ice transport policy to simulate NAT. A direct connection (with host candidates) woks as expected. Browser <-> browser connection with the same turn server works as expected. Here is a full log:
(webrtcvoiceengine.cc:570): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtcvoiceengine.cc:577): Supported send codecs in order of preference:
(webrtcvoiceengine.cc:304): Adding supported codec: opus/48000/2 (111)
(webrtcvoiceengine.cc:304): Adding supported codec: ISAC/16000/1 (103)
(webrtcvoiceengine.cc:304): Adding supported codec: ISAC/32000/1 (104)
(webrtcvoiceengine.cc:304): Adding supported codec: G722/8000/1 (9)
(webrtcvoiceengine.cc:304): Adding supported codec: ILBC/8000/1 (102)
(webrtcvoiceengine.cc:304): Adding supported codec: PCMU/8000/1 (0)
(webrtcvoiceengine.cc:304): Adding supported codec: PCMA/8000/1 (8)
(webrtcvoiceengine.cc:304): Adding supported codec: CN/32000/1 (106)
(webrtcvoiceengine.cc:304): Adding supported codec: CN/16000/1 (105)
(webrtcvoiceengine.cc:304): Adding supported codec: CN/8000/1 (13)
(webrtcvoiceengine.cc:304): Adding supported codec: telephone-event/48000/1 (110)
(webrtcvoiceengine.cc:304): Adding supported codec: telephone-event/32000/1 (112)
(webrtcvoiceengine.cc:304): Adding supported codec: telephone-event/16000/1 (113)
(webrtcvoiceengine.cc:304): Adding supported codec: telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:580): opus/48000/2 (111)
(webrtcvoiceengine.cc:580): ISAC/16000/1 (103)
(webrtcvoiceengine.cc:580): ISAC/32000/1 (104)
(webrtcvoiceengine.cc:580): G722/8000/1 (9)
(webrtcvoiceengine.cc:580): ILBC/8000/1 (102)
(webrtcvoiceengine.cc:580): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:580): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:580): CN/32000/1 (106)
(webrtcvoiceengine.cc:580): CN/16000/1 (105)
(webrtcvoiceengine.cc:580): CN/8000/1 (13)
(webrtcvoiceengine.cc:580): telephone-event/48000/1 (110)
(webrtcvoiceengine.cc:580): telephone-event/32000/1 (112)
(webrtcvoiceengine.cc:580): telephone-event/16000/1 (113)
(webrtcvoiceengine.cc:580): telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:583): Supported recv codecs in order of preference:
(webrtcvoiceengine.cc:586): opus/48000/2 (111)
(webrtcvoiceengine.cc:586): isac/16000/1 (103)
(webrtcvoiceengine.cc:586): isac/32000/1 (104)
(webrtcvoiceengine.cc:586): G722/8000/1 (9)
(webrtcvoiceengine.cc:586): iLBC/8000/1 (102)
(webrtcvoiceengine.cc:586): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:586): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:586): cn/32000/1 (106)
(webrtcvoiceengine.cc:586): cn/16000/1 (105)
(webrtcvoiceengine.cc:586): cn/8000/1 (13)
(webrtcvoiceengine.cc:586): telephone-event/48000/1 (110)
(webrtcvoiceengine.cc:586): telephone-event/32000/1 (112)
(webrtcvoiceengine.cc:586): telephone-event/16000/1 (113)
(webrtcvoiceengine.cc:586): telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:594): VoiceEngine 4.1.0
(audio_device_impl.cc:84): Create
(audio_device_buffer.cc:62): AudioDeviceBuffer::ctor
(audio_device_impl.cc:128): AudioDeviceModuleImpl
(audio_device_impl.cc:136): CheckPlatform
(audio_device_impl.cc:153): current platform is IOS
(audio_device_impl.cc:177): CreatePlatformSpecificObjects
(audio_device_impl.cc:1844): PlatformAudioLayer
(audio_device_ios.mm:102): AudioDeviceIOS::ctor<NSThread: 0x17007a480>{number = 3, name = (null)}
(audio_device_impl.cc:297): iPhone Audio APIs will be utilized
(audio_device_impl.cc:344): AttachAudioBuffer
(audio_device_ios.mm:116): AudioDeviceIOS::AttachAudioBuffer
(audio_device_impl.cc:1452): RegisterEventObserver
(audio_device_impl.cc:1465): RegisterAudioCallback
(audio_device_buffer.cc:77): RegisterAudioCallback
(audio_device_impl.cc:467): Init
(audio_device_ios.mm:123): AudioDeviceIOS::Init
(audio_device_ios.mm:74): LogDeviceInfo
(audio_device_ios.mm:76): system name: iOS
(audio_device_ios.mm:77): system version 1(2): 10.3.1
(audio_device_ios.mm:78): system version 2(2): 10.3
(audio_device_ios.mm:79): device type: iPhone
(audio_device_ios.mm:80): device name: iPhone6,2
(audio_device_ios.mm:81): process name: Test
(audio_device_ios.mm:82): process ID: 829
(audio_device_ios.mm:83): OS version: Version 10.3.1 (Build 14E304)
(audio_device_ios.mm:84): processing cores: 2
(audio_device_ios.mm:87): low power mode: 0
(audio_device_ios.mm:582): AudioDeviceIOS::UpdateAudioDevicebuffer
(audio_device_buffer.cc:185): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:209): SetPlayoutChannels(1)
(audio_device_buffer.cc:178): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:202): SetRecordingChannels(1)
(audio_device_impl.cc:1196): SetPlayoutDevice(0)
(audio_device_not_implemented_ios.mm:111): virtual int32_t webrtc::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented
(audio_device_impl.cc:516): InitSpeaker
(audio_device_impl.cc:1291): SetRecordingDevice(0)
(audio_device_not_implemented_ios.mm:260): virtual int32_t webrtc::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented
(audio_device_impl.cc:526): InitMicrophone
(audio_device_impl.cc:1005): StereoPlayoutIsAvailable
(audio_device_impl.cc:1015): output: 0
(audio_device_impl.cc:1024): SetStereoPlayout(0)
(audio_device_not_implemented_ios.mm:195): virtual int32_t webrtc::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented
(audio_device_impl.cc:1034): stereo playout is not supported
(webrtcvoiceengine.cc:974): webrtc: Init() failed to set mono/stereo playout mode (error=8090)
(audio_device_impl.cc:891): StereoRecordingIsAvailable
(audio_device_impl.cc:901): output: 0
(audio_device_impl.cc:910): SetStereoRecording(0)
(audio_device_not_implemented_ios.mm:180): virtual int32_t webrtc::AudioDeviceIOS::SetStereoRecording(bool): Not implemented
(audio_device_impl.cc:919): failed to change stereo recording
(webrtcvoiceengine.cc:974): webrtc: Init() failed to set mono/stereo recording mode (error=8090)
(webrtcvoiceengine.cc:974): webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x4681fa00)
(webrtcvoiceengine.cc:974): webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x4681fa00)
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:631): Level controller activated: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:638): Highpass filter activated: 1
(webrtcvoiceengine.cc:670): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, typing: true, agc_delta: 0, experimental_agc: false, extended_filter_aec: false, delay_agnostic_aec: false, experimental_ns: false, intelligibility_enhancer: false, level_control: false, residual_echo_detector: true, }
(webrtcvoiceengine.cc:686): Always disable AEC, NS and AGC on iOS. Use built-in instead.
(audio_device_impl.cc:1760): BuiltInAECIsAvailable
(audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1763): output: 0
(apm_helpers.cc:106): Echo control set to 0 with mode 0
(apm_helpers.cc:116): EC metrics set to 0
(audio_device_impl.cc:1776): BuiltInAGCIsAvailable
(audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1779): output: 0
(audio_device_impl.cc:1071): SetAGC(0)
(apm_helpers.cc:67): Failed to set AGC mode in ADM: 0
(webrtcvoiceengine.cc:784): Adjusting AGC level from default -3dB to -3dB
(audio_device_impl.cc:1792): BuiltInNSIsAvailable
(audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1795): output: 0
(apm_helpers.cc:141): NS set to 0
(webrtcvoiceengine.cc:815): Stereo swapping enabled? 0
(webrtcvoiceengine.cc:820): NetEq capacity is 50
(webrtcvoiceengine.cc:826): NetEq fast mode? 0
(webrtcvoiceengine.cc:833): Typing detection is enabled? 0
(webrtcvoiceengine.cc:844): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:853): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:862): Experimental ns is enabled? 0
(webrtcvoiceengine.cc:868): Intelligibility Enhancer is enabled? 0
(webrtcvoiceengine.cc:878): Level control: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:631): Level controller activated: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:638): Highpass filter activated: 1
(audio_device_impl.cc:1465): RegisterAudioCallback
(audio_device_buffer.cc:77): RegisterAudioCallback
(audio_device_impl.cc:1465): RegisterAudioCallback
(audio_device_buffer.cc:77): RegisterAudioCallback
(webrtcvideoengine2.cc:461): WebRtcVideoEngine2::WebRtcVideoEngine2()
(webrtcvideoengine2.cc:469): WebRtcVideoEngine2::Init
(messagequeue.cc:542): Message took 63ms to dispatch. Posted from: CreatePeerConnectionFactoryWithAudioMixer@../../webrtc/pc/peerconnectionfactory.cc:110
2017-05-16 21:18:25.931781+0200 Test[829:176559] [MC] System group container for systemgroup.com.apple.configurationprofiles path is /private/var/containers/Shared/SystemGroup/systemgroup.com.apple.configurationprofiles
2017-05-16 21:18:25.933896+0200 Test[829:176559] [MC] Reading from public effective user settings.
2017-05-16 21:18:26.031260+0200 Test[829:176559] libMobileGestalt MobileGestaltSupport.m:153: pid 829 (Test) does not have sandbox access for frZQaeyWLUvLjeuEK43hmg and IS NOT appropriately entitled
2017-05-16 21:18:26.031369+0200 Test[829:176559] libMobileGestalt MobileGestalt.c:550: no access to InverseDeviceID (see <rdar://problem/11744455>)
(RTCLogging.mm:31): (RTCAVFoundationVideoCapturerInternal.mm:284 -[RTCAVFoundationVideoCapturerInternal handleCaptureSessionDidStartRunning:]): Capture session started.
2017-05-16 21:18:38.404596+0200 Test[829:176559] [ARLRoomServerClientImpl.m joinRoomWithId: 44] [ARLRoomServerClientImpl::joinRoomWithId] url: https://webrtc.lost.in.ua/join/200119
2017-05-16 21:18:39.160117+0200 Test[829:176600] [ARLRoomServerClientImpl.m joinRoomWithId: 65] [ARLRoomServerClientImpl::joinRoomWithId] response: {"params": {"is_initiator": "true", "room_link": "https://webrtc.lost.in.ua/r/200119", "turn_server_override": [{"username": "1", "credential": "1", "urls": ["turn:webrtc.lost.in.ua:3478?transport=udp", "turn:webrtc.lost.in.ua:3478?transport=tcp"]}, {"urls": ["stun:webrtc.lost.in.ua:3478"]}], "ice_server_transports": "", "media_constraints": "{\"audio\": true, \"video\": true}", "include_loopback_js": "", "turn_url": "https://webrtc.lost.in.ua/nopage/turn?username=30398841&key=111111", "wss_url": "wss://webrtc.lost.in.ua/ws", "pc_constraints": "{\"optional\": []}", "pc_config": "{\"rtcpMuxPolicy\": \"require\", \"bundlePolicy\": \"max-bundle\", \"iceServers\": []}", "wss_post_url": "https://webrtc.lost.in.ua", "ice_server_url": "https://webrtc.lost.in.ua/getice/iceconfig?key=None", "warning_messages": [], "room_id": "200119", "include_rtstats_js": "", "version_info": "{\"gitHash\": \"3aaa27dbb025f53e00e19172b7d3704ce719271e\", \"branch\": \"master\", \"time\": \"Mon Mar 6 07:49:29 2017 +0100\"}", "error_messages": [], "client_id": "30398841", "bypass_join_confirmation": "false", "room_token": "tok-52905449-11261956-17071321", "is_loopback": "false", "offer_options": "{}", "messages": [], "callstats_params": "{\"appSecret\": null, \"appId\": null}"}, "result": "SUCCESS"}
(webrtcvideoengine2.cc:601): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
(webrtcvideoengine2.cc:611): Codecs supported by the external encoder factory: {VideoCodec[0:H264]}
(webrtcsessiondescriptionfactory.cc:163): DTLS-SRTP enabled; sending DTLS identity request (key type: 1).
(opensslidentity.cc:41): Making key pair
(RTCLogging.mm:31): (RTCRtpSender.mm:89 -[RTCRtpSender initWithNativeRtpSender:]): RTCRtpSender(0x174014510): created sender: RTCRtpSender {
senderId: da3a30e8-f2a2-4c7e-a906-ca89527178d9
}
(RTCLogging.mm:31): (RTCRtpSender.mm:89 -[RTCRtpSender initWithNativeRtpSender:]): RTCRtpSender(0x170012970): created sender: RTCRtpSender {
senderId: f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07
}
(opensslidentity.cc:89): Returning key pair
(opensslidentity.cc:96): Making certificate for WebRTC
2017-05-16 21:18:39.189276+0200 Test[829:176600] SocketRocket: In debug mode. Allowing connection to any root cert
(opensslidentity.cc:143): Returning certificate
(opensslidentity.cc:276): Certificate:
Data:
Version: 3 (0x2)
Serial Number:
98:46:87:a6:65:02:8b:83
Signature Algorithm: ecdsa-with-SHA256
Issuer: CN=WebRTC
Validity
Not Before: May 15 19:18:39 2017 GMT
Not After : Jun 15 19:18:39 2017 GMT
Subject: CN=WebRTC
Subject Public Key Info:
Public Key Algorithm: id-ecPublicKey
Public-Key: (256 bit)
00000000 04 18 4a 25 6e c7 51 2f c5 ae 84 86 d2 b4 93 a3 |..J%n.Q/........|
00000010 3c 03 7e 5f 3a 1c 8a c3 a7 b8 08 f9 04 74 35 9b |<.~_:........t5.|
00000020 48 03 62 bc ba 34 b3 fd f4 d0 2c e8 00 9c ad c0 |H.b..4....,.....|
00000030 de 95 b1 d1 ce 0e c2 f7 14 d7 09 55 7a cb 5b c3 |...........Uz.[.|
00000040 f2 |.|
Signature Algorithm: ecdsa-with-SHA256
30:45:02:21:00:9d:fc:85:48:71:58:cd:65:b4:53:fa:94:55:
2a:b2:e7:7a:f1:5c:af:cf:0d:9a:e4:86:a6:ac:70:d8:f9:9e:
92:02:20:16:6e:82:b5:e7:f5:a0:70:ec:44:6b:bb:d2:c6:3b:
31:71:65:3b:3d:2a:38:cd:70:8e:cd:97:28:a1:61:5f:ac
(webrtcsessiondescriptionfactory.cc:492): Setting new certificate.
(mediasession.cc:347): Duplicate id found. Reassigning from 102 to 127
(mediasession.cc:347): Duplicate id found. Reassigning from 103 to 125
(mediasession.cc:347): Duplicate id found. Reassigning from 104 to 124
(webrtcsession.cc:1698): Bundling audio on audio
(p2ptransportchannel.cc:394): Set ICE receiving timeout to 1000 milliseconds
(p2ptransportchannel.cc:400): Set ping most likely connection to 0
(p2ptransportchannel.cc:420): Set presume writable when fully relayed to 0
(cpu_info.cc:50): Available number of cores: 2
(remote_bitrate_estimator_single_stream.cc:58): RemoteBitrateEstimatorSingleStream: Instantiating.
(bitrate_prober.cc:63): Bandwidth probing enabled, set to inactive
(delay_based_bwe.cc:167): Using Trendline filter for delay change estimation.
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(delay_based_bwe.cc:351): BWE Setting start bitrate to: 300000
(paced_sender.cc:477): ProcessThreadAttached 0x0x17418efb0
(webrtcvoiceengine.cc:1611): WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel
(webrtcvoiceengine.cc:1790): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(webrtcvoiceengine.cc:670): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(webrtcvoiceengine.cc:686): Always disable AEC, NS and AGC on iOS. Use built-in instead.
(audio_device_impl.cc:1760): BuiltInAECIsAvailable
(audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1763): output: 0
(apm_helpers.cc:106): Echo control set to 0 with mode 0
(apm_helpers.cc:116): EC metrics set to 0
(audio_device_impl.cc:1776): BuiltInAGCIsAvailable
(audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1779): output: 0
(audio_device_impl.cc:1071): SetAGC(0)
(apm_helpers.cc:67): Failed to set AGC mode in ADM: 0
(audio_device_impl.cc:1792): BuiltInNSIsAvailable
(audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1795): output: 0
(apm_helpers.cc:141): NS set to 0
(webrtcvoiceengine.cc:820): NetEq capacity is 50
(webrtcvoiceengine.cc:826): NetEq fast mode? 0
(webrtcvoiceengine.cc:833): Typing detection is enabled? 0
(webrtcvoiceengine.cc:844): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:853): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:862): Experimental ns is enabled? 0
(webrtcvoiceengine.cc:868): Intelligibility Enhancer is enabled? 0
(webrtcvoiceengine.cc:878): Level control: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:631): Level controller activated: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:638): Highpass filter activated: 1
(webrtcvoiceengine.cc:1809): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(channel.cc:178): Created channel for audio
(channel.cc:338): Setting RTP Transport for audio on audio transport 0x145d9d980
(webrtcvoiceengine.cc:2484): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(webrtcsession.cc:1698): Bundling video on audio
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(webrtcvoiceengine.cc:2484): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(webrtcvideoengine2.cc:478): CreateChannel. Options: VideoOptions {}
(webrtcvideoengine2.cc:601): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
(webrtcvideoengine2.cc:611): Codecs supported by the external encoder factory: {VideoCodec[0:H264]}
(channel.cc:178): Created channel for video
(channel.cc:338): Setting RTP Transport for video on audio transport 0x145d9d980
(webrtcvideoengine2.cc:1449): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(webrtcvideoengine2.cc:1449): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(transportcontroller.cc:611): Set local transport description on audio
(p2ptransportchannel.cc:323): Set ICE ufrag: kGwq pwd: 9uDawFR6FMcP/JNqej40NRxn on transport audio
(webrtcsession.cc:844): Session:8610189052997493170 Old state:STATE_INIT New state:STATE_SENTOFFER
2017-05-16 21:18:39.252834+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeSignalingState: 149] Signaling state changed: RTCSignalingStateHaveLocalOffer
(channel.cc:1797): Setting local voice description
(webrtcvoiceengine.cc:1672): 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[102:ILBC:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}
(webrtcvoiceengine.cc:1819): Setting receive voice codecs.
(webrtcvoiceengine.cc:2120): AddSendStream: {id:da3a30e8-f2a2-4c7e-a906-ca89527178d9;ssrcs:[3150167243];ssrc_groups:;cname:kblAp4EIPh6Csu7q;sync_label:LocalMediaStream}
(neteq_impl.cc:109): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, background_noise_mode=2, playout_mode=0, enable_fast_accelerate=false, enable_muted_state= true
(neteq_impl.cc:2064): SetSampleRateAndChannels 16000 1
(neteq_impl.cc:529): FlushBuffers
(neteq_impl.cc:529): FlushBuffers
(audio_send_stream.cc:66): AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: false, enable_codec_fec: false, enable_opus_dtx: false, opus_max_playback_rate: 0, cng_payload_type: -1, cng_plfreq: -1, min_ptime: -1, max_ptime: -1, codec_inst: {pltype: -1, plname: "", plfreq: 0, pacsize: 0, channels: 0, rate: 0}}}
(audio_send_stream.cc:344): SetSendCodec() failed.
(audio_send_stream.cc:103): Failed to set up send codec state.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(channel.cc:1378): Add send stream ssrc: 3150167243
(channel.cc:1784): Changing voice state, recv=0 send=0
(channel.cc:2075): Setting local video description
(webrtcvideoengine2.cc:950): SetRecvParameters: {codecs: [VideoCodec[125:H264], VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:red], VideoCodec[127:ulpfec], VideoCodec[97:rtx], VideoCodec[99:rtx], VideoCodec[101:rtx], VideoCodec[124:rtx]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}]}
(webrtcvideoengine2.cc:601): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
(webrtcvideoengine2.cc:611): Codecs supported by the external encoder factory: {VideoCodec[0:H264]}
(webrtcvideoengine2.cc:959): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[103:H264]} to {VideoCodec[125:H264], VideoCodec[96:VP8], VideoCodec[98:VP9]}
(webrtcvideoengine2.cc:1065): AddSendStream: {id:f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07;ssrcs:[272048794,3445604732];ssrc_groups:{semantics:FID;ssrcs:[272048794,3445604732]};cname:kblAp4EIPh6Csu7q;sync_label:LocalMediaStream}
(webrtcvideoengine2.cc:1093): SetLocalSsrc on all the receive streams because we added a send stream.
(channel.cc:1378): Add send stream ssrc: 272048794
(webrtcvideoengine2.cc:1000): SetSend: false
(channel.cc:2042): Changing video state, send=0
(webrtcsession.cc:657): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(webrtcvoiceengine.cc:1790): Setting voice channel options: AudioOptions {}
(webrtcvoiceengine.cc:670): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(webrtcvoiceengine.cc:686): Always disable AEC, NS and AGC on iOS. Use built-in instead.
(audio_device_impl.cc:1760): BuiltInAECIsAvailable
(audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1763): output: 0
(apm_helpers.cc:106): Echo control set to 0 with mode 0
(apm_helpers.cc:116): EC metrics set to 0
(audio_device_impl.cc:1776): BuiltInAGCIsAvailable
(audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1779): output: 0
(audio_device_impl.cc:1071): SetAGC(0)
(apm_helpers.cc:67): Failed to set AGC mode in ADM: 0
(audio_device_impl.cc:1792): BuiltInNSIsAvailable
(audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1795): output: 0
(apm_helpers.cc:141): NS set to 0
(webrtcvoiceengine.cc:820): NetEq capacity is 50
(webrtcvoiceengine.cc:826): NetEq fast mode? 0
(webrtcvoiceengine.cc:833): Typing detection is enabled? 0
(webrtcvoiceengine.cc:844): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:853): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:862): Experimental ns is enabled? 0
(webrtcvoiceengine.cc:868): Intelligibility Enhancer is enabled? 0
(webrtcvoiceengine.cc:878): Level control: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:631): Level controller activated: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:638): Highpass filter activated: 1
(webrtcvoiceengine.cc:1809): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(webrtcvideoengine2.cc:1025): SetVideoSend (ssrc= 272048794, enable = 1, options: VideoOptions {is_screencast : false, }, source = (source))
(basicportallocator.cc:274): Start getting ports with prune_turn_ports disabled
(messagequeue.cc:542): Message took 60ms to dispatch. Posted from: SetLocalDescription@../../webrtc/api/peerconnectionproxy.h:85
2017-05-16 21:18:39.282537+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeIceGatheringState: 224] ICE gathering state changed: RTCIceGatheringStateGathering
2017-05-16 21:18:39.282564+0200 Test[829:176559] [ARLWebSocketChannel.m sendMessage: 125] {"type":"offer","sdp":"v=0\r\no=- 8610189052997493170 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS LocalMediaStream\r\nm=audio 9 UDP\/TLS\/RTP\/SAVPF 111 103 104 9 102 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:kGwq\r\na=ice-pwd:9uDawFR6FMcP\/JNqej40NRxn\r\na=ice-options:renomination\r\na=fingerprint:sha-256 2A:F7:9D:89:85:A9:6B:4D:08:FB:AD:DC:22:64:F3:3B:19:A4:8B:26:83:35:92:A2:D1:28:AA:B2:4C:FB:1F:E1\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\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:102 ILBC\/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\na=ssrc:3150167243 cname:kblAp4EIPh6Csu7q\r\na=ssrc:3150167243 msid:LocalMediaStream da3a30e8-f2a2-4c7e-a906-ca89527178d9\r\na=ssrc:3150167243 mslabel:LocalMediaStream\r\na=ssrc:3150167243 label:da3a30e8-f2a2-4c7e-a906-ca89527178d9\r\nm=video 9 UDP\/TLS\/RTP\/SAVPF 125 96 98 100 127 97 99 101 124\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:kGwq\r\na=ice-pwd:9uDawFR6FMcP\/JNqej40NRxn\r\na=ice-options:renomination\r\na=fingerprint:sha-256 2A:F7:9D:89:85:A9:6B:4D:08:FB:AD:DC:22:64:F3:3B:19:A4:8B:26:83:35:92:A2:D1:28:AA:B2:4C:FB:1F:E1\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/playout-delay\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8\/90000\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtpmap:98 VP9\/90000\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtpmap:100 red\/90000\r\na=rtpmap:127 ulpfec\/90000\r\na=rtpmap:125 H264\/90000\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:97 rtx\/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:99 rtx\/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:101 rtx\/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:124 rtx\/90000\r\na=fmtp:124 apt=125\r\na=ssrc-group:FID 272048794 3445604732\r\na=ssrc:272048794 cname:kblAp4EIPh6Csu7q\r\na=ssrc:272048794 msid:LocalMediaStream f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:272048794 mslabel:LocalMediaStream\r\na=ssrc:272048794 label:f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:3445604732 cname:kblAp4EIPh6Csu7q\r\na=ssrc:3445604732 msid:LocalMediaStream f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:3445604732 mslabel:LocalMediaStream\r\na=ssrc:3445604732 label:f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\n"}
(RTCLogging.mm:31): (RTCRtpSender.mm:89 -[RTCRtpSender initWithNativeRtpSender:]): RTCRtpSender(0x1700147a0): created sender: RTCRtpSender {
senderId: da3a30e8-f2a2-4c7e-a906-ca89527178d9
}
(RTCLogging.mm:31): (RTCRtpSender.mm:89 -[RTCRtpSender initWithNativeRtpSender:]): RTCRtpSender(0x170014770): created sender: RTCRtpSender {
senderId: f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07
}
(basicportallocator.cc:676): Network manager has started
(basicportallocator.cc:594): Allocate ports on 2 networks
(basicportallocator.cc:1143): Jingle:Net[en2:169.254.0.0/16:Wifi]: Allocation Phase=Udp
(port.cc:214): Jingle:Port[0x1468a1600::1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Port created with network cost 10
(basicportallocator.cc:698): Adding allocated port for audio
(basicportallocator.cc:718): Jingle:Port[0x1468a1600:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Added port to allocator
(basicportallocator.cc:735): Jingle:Port[0x1468a1600:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Gathered candidate: Cand[:1212275373:1:udp:2122260223:169.254.218.97:56317:local::0:kGwq:9uDawFR6FMcP/JNqej40NRxn:2:10:0]
(basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:844): Jingle:Port[0x1468a1600:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Port completed gathering candidates.
(basicportallocator.cc:1143): Jingle:Net[en0:192.168.0.0/24:Wifi]: Allocation Phase=Udp
(port.cc:214): Jingle:Port[0x1468aec00::1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Port created with network cost 10
(basicportallocator.cc:698): Adding allocated port for audio
(basicportallocator.cc:718): Jingle:Port[0x1468aec00:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Added port to allocator
(basicportallocator.cc:735): Jingle:Port[0x1468aec00:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Gathered candidate: Cand[:1002794138:1:udp:2122194687:192.168.0.245:59518:local::0:kGwq:9uDawFR6FMcP/JNqej40NRxn:1:10:0]
(basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:844): Jingle:Port[0x1468aec00:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Port completed gathering candidates.
(basicportallocator.cc:1143): Jingle:Net[en2:169.254.0.0/16:Wifi]: Allocation Phase=Relay
(basicportallocator.cc:1298): AllocationSequence: Relay ports disabled, skipping.
(basicportallocator.cc:1143): Jingle:Net[en0:192.168.0.0/24:Wifi]: Allocation Phase=Relay
(basicportallocator.cc:1298): AllocationSequence: Relay ports disabled, skipping.
(basicportallocator.cc:1143): Jingle:Net[en2:169.254.0.0/16:Wifi]: Allocation Phase=Tcp
(port.cc:214): Jingle:Port[0x145da8d50::1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Port created with network cost 10
(basicportallocator.cc:698): Adding allocated port for audio
(basicportallocator.cc:718): Jingle:Port[0x145da8d50:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Added port to allocator
(tcpport.cc:174): Preparing TCP address, current state: 2
(basicportallocator.cc:735): Jingle:Port[0x145da8d50:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Gathered candidate: Cand[:113387101:1:tcp:1518280447:169.254.218.97:55256:local::0:kGwq:9uDawFR6FMcP/JNqej40NRxn:2:10:0]
(basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:844): Jingle:Port[0x145da8d50:audio:1:0:local:Net[en2:169.254.0.0/16:Wifi]]: Port completed gathering candidates.
(basicportallocator.cc:1143): Jingle:Net[en0:192.168.0.0/24:Wifi]: Allocation Phase=Tcp
(port.cc:214): Jingle:Port[0x145d75b60::1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Port created with network cost 10
(basicportallocator.cc:698): Adding allocated port for audio
(basicportallocator.cc:718): Jingle:Port[0x145d75b60:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Added port to allocator
(tcpport.cc:174): Preparing TCP address, current state: 2
(basicportallocator.cc:735): Jingle:Port[0x145d75b60:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Gathered candidate: Cand[:1967558762:1:tcp:1518214911:192.168.0.245:55257:local::0:kGwq:9uDawFR6FMcP/JNqej40NRxn:1:10:0]
(basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
(basicportallocator.cc:844): Jingle:Port[0x145d75b60:audio:1:0:local:Net[en0:192.168.0.0/24:Wifi]]: Port completed gathering candidates.
(basicportallocator.cc:1143): Jingle:Net[en2:169.254.0.0/16:Wifi]: Allocation Phase=SslTcp
(basicportallocator.cc:1143): Jingle:Net[en0:192.168.0.0/24:Wifi]: Allocation Phase=SslTcp
(basicportallocator.cc:968): All candidates gathered for audio:1:0
(p2ptransportchannel.cc:582): P2PTransportChannel: audio, component 1 gathering complete
2017-05-16 21:18:39.452985+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeIceGatheringState: 224] ICE gathering state changed: RTCIceGatheringStateComplete
2017-05-16 21:18:39.689351+0200 Test[829:176559] [ARLRoomServerClientImpl.m webSocketChannel:didChangeState: 180]
2017-05-16 21:18:39.689628+0200 Test[829:176559] [ARLRoomServerClientImpl.m webSocketChannel:didChangeState: 180]
2017-05-16 21:18:39.689859+0200 Test[829:176559] [ARLWebSocketChannel.m sendMessage: 125] {"type":"offer","sdp":"v=0\r\no=- 8610189052997493170 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS LocalMediaStream\r\nm=audio 9 UDP\/TLS\/RTP\/SAVPF 111 103 104 9 102 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:kGwq\r\na=ice-pwd:9uDawFR6FMcP\/JNqej40NRxn\r\na=ice-options:renomination\r\na=fingerprint:sha-256 2A:F7:9D:89:85:A9:6B:4D:08:FB:AD:DC:22:64:F3:3B:19:A4:8B:26:83:35:92:A2:D1:28:AA:B2:4C:FB:1F:E1\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\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:102 ILBC\/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\na=ssrc:3150167243 cname:kblAp4EIPh6Csu7q\r\na=ssrc:3150167243 msid:LocalMediaStream da3a30e8-f2a2-4c7e-a906-ca89527178d9\r\na=ssrc:3150167243 mslabel:LocalMediaStream\r\na=ssrc:3150167243 label:da3a30e8-f2a2-4c7e-a906-ca89527178d9\r\nm=video 9 UDP\/TLS\/RTP\/SAVPF 125 96 98 100 127 97 99 101 124\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:kGwq\r\na=ice-pwd:9uDawFR6FMcP\/JNqej40NRxn\r\na=ice-options:renomination\r\na=fingerprint:sha-256 2A:F7:9D:89:85:A9:6B:4D:08:FB:AD:DC:22:64:F3:3B:19:A4:8B:26:83:35:92:A2:D1:28:AA:B2:4C:FB:1F:E1\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/playout-delay\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8\/90000\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtpmap:98 VP9\/90000\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtpmap:100 red\/90000\r\na=rtpmap:127 ulpfec\/90000\r\na=rtpmap:125 H264\/90000\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:97 rtx\/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:99 rtx\/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:101 rtx\/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:124 rtx\/90000\r\na=fmtp:124 apt=125\r\na=ssrc-group:FID 272048794 3445604732\r\na=ssrc:272048794 cname:kblAp4EIPh6Csu7q\r\na=ssrc:272048794 msid:LocalMediaStream f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:272048794 mslabel:LocalMediaStream\r\na=ssrc:272048794 label:f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:3445604732 cname:kblAp4EIPh6Csu7q\r\na=ssrc:3445604732 msid:LocalMediaStream f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\na=ssrc:3445604732 mslabel:LocalMediaStream\r\na=ssrc:3445604732 label:f02ffc3e-e6a1-4e4b-b7cb-2fd9dfe0ef07\r\n"}
2017-05-16 21:18:39.694892+0200 Test[829:176559] [ARLWebSocketChannel.m createKeepAliveTimer 282] keepalive
2017-05-16 21:18:44.651818+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"type":"answer","sdp":"v=0\r\no=- 6659742695368938815 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB\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:TG0t\r\na=ice-pwd:CkCxTFgvd4QFlM3pKXej//Oq\r\na=fingerprint:sha-256 B7:3F:D7:45:A8:5E:C8:93:96:9E:13:52:28:3C:A3:C2:A2:2C:B1:65:A6:21:5E:DC:29:B8:45:15:11:B3:2D:9D\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=recvonly\r\na=rtcp-mux\r\na=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=video 9 UDP/TLS/RTP/SAVPF 125 96 98 100 127 97 99 101 124\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:TG0t\r\na=ice-pwd:CkCxTFgvd4QFlM3pKXej//Oq\r\na=fingerprint:sha-256 B7:3F:D7:45:A8:5E:C8:93:96:9E:13:52:28:3C:A3:C2:A2:2C:B1:65:A6:21:5E:DC:29:B8:45:15:11:B3:2D:9D\r\na=setup:active\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtpmap:100 red/90000\r\na=rtpmap:127 ulpfec/90000\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=125\r\na=ssrc-group:FID 863849452 52816360\r\na=ssrc:863849452 cname:mpKG5R6g/TKLwCmx\r\na=ssrc:863849452 msid:uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB 9c49c137-1f40-4622-acbd-0d03e1f9cfbe\r\na=ssrc:863849452 mslabel:uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB\r\na=ssrc:863849452 label:9c49c137-1f40-4622-acbd-0d03e1f9cfbe\r\na=ssrc:52816360 cname:mpKG5R6g/TKLwCmx\r\na=ssrc:52816360 msid:uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB 9c49c137-1f40-4622-acbd-0d03e1f9cfbe\r\na=ssrc:52816360 mslabel:uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB\r\na=ssrc:52816360 label:9c49c137-1f40-4622-acbd-0d03e1f9cfbe\r\n"}
(webrtcvideoengine2.cc:1315): Call stats: 67777117, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
(webrtcsession.cc:1079): BUNDLE already enabled for audio on audio.
(webrtcsession.cc:1079): BUNDLE already enabled for video on audio.
(transportcontroller.cc:639): Set remote transport description on audio
(p2ptransportchannel.cc:334): Remote supports ICE renomination ? 0
(dtlstransportchannel.cc:313): Jingle:DtlsTransport[audio|1|__]: DTLS setup complete.
(p2ptransportchannel.cc:1264): Sorting 0 available connections:
(channel.cc:953): Channel enabled
(channel.cc:1784): Changing voice state, recv=1 send=0
(channel.cc:953): Channel enabled
(webrtcvideoengine2.cc:1000): SetSend: false
(channel.cc:2042): Changing video state, send=0
(webrtcsession.cc:844): Session:8610189052997493170 Old state:STATE_SENTOFFER New state:STATE_INPROGRESS
2017-05-16 21:18:44.702563+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeSignalingState: 149] Signaling state changed: RTCSignalingStateStable
(channel.cc:1842): Setting remote voice description
(channel.cc:1275): Enabling rtcp-mux for audio; no longer need RTCP transport for audio
(webrtcvoiceengine.cc:1640): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], max_bandwidth_bps: -1, options: AudioOptions {}}
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_send_stream.cc:111): ~AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: false, enable_codec_fec: false, enable_opus_dtx: false, opus_max_playback_rate: 0, cng_payload_type: -1, cng_plfreq: -1, min_ptime: -1, max_ptime: -1, codec_inst: {pltype: -1, plname: "", plfreq: 0, pacsize: 0, channels: 0, rate: 0}}}
(audio_send_stream.cc:66): AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: 48000, cng_payload_type: -1, cng_plfreq: -1, min_ptime: 10, max_ptime: 120, codec_inst: {pltype: 111, plname: "opus", plfreq: 48000, pacsize: 960, channels: 1, rate: 32000}}}
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(webrtcvoiceengine.cc:2020): Recreate all the receive streams because the send codec has changed.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_send_stream.cc:111): ~AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: 48000, cng_payload_type: -1, cng_plfreq: -1, min_ptime: 10, max_ptime: 120, codec_inst: {pltype: 111, plname: "opus", plfreq: 48000, pacsize: 960, channels: 1, rate: 32000}}}
(audio_send_stream.cc:66): AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: 48000, cng_payload_type: -1, cng_plfreq: -1, min_ptime: 10, max_ptime: 120, codec_inst: {pltype: 111, plname: "opus", plfreq: 48000, pacsize: 960, channels: 1, rate: 32000}}}
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(webrtcvoiceengine.cc:2471): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
(webrtcvoiceengine.cc:1790): Setting voice channel options: AudioOptions {}
(webrtcvoiceengine.cc:670): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(webrtcvoiceengine.cc:686): Always disable AEC, NS and AGC on iOS. Use built-in instead.
(audio_device_impl.cc:1760): BuiltInAECIsAvailable
(audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1763): output: 0
(apm_helpers.cc:106): Echo control set to 0 with mode 0
(apm_helpers.cc:116): EC metrics set to 0
(audio_device_impl.cc:1776): BuiltInAGCIsAvailable
(audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1779): output: 0
(audio_device_impl.cc:1071): SetAGC(0)
(apm_helpers.cc:67): Failed to set AGC mode in ADM: 0
(audio_device_impl.cc:1792): BuiltInNSIsAvailable
(audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
(audio_device_impl.cc:1795): output: 0
(apm_helpers.cc:141): NS set to 0
(webrtcvoiceengine.cc:820): NetEq capacity is 50
(webrtcvoiceengine.cc:826): NetEq fast mode? 0
(webrtcvoiceengine.cc:833): Typing detection is enabled? 0
(webrtcvoiceengine.cc:844): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:853): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:862): Experimental ns is enabled? 0
(webrtcvoiceengine.cc:868): Intelligibility Enhancer is enabled? 0
(webrtcvoiceengine.cc:878): Level control: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:631): Level controller activated: 0
(webrtcvoiceengine.cc:974): webrtc: (audio_processing_impl.cc:638): Highpass filter activated: 1
(webrtcvoiceengine.cc:1809): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(channel.cc:1784): Changing voice state, recv=1 send=0
(webrtcvoiceengine.cc:2484): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(channel.cc:2120): Setting remote video description
(channel.cc:1275): Enabling rtcp-mux for video; no longer need RTCP transport for audio
(webrtcvideoengine2.cc:748): SetSendParameters: {codecs: [VideoCodec[125:H264], VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:red], VideoCodec[127:ulpfec], VideoCodec[97:rtx], VideoCodec[99:rtx], VideoCodec[101:rtx], VideoCodec[124:rtx]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}], max_bandwidth_bps: -1, }
(webrtcvideoengine2.cc:601): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
(webrtcvideoengine2.cc:611): Codecs supported by the external encoder factory: {VideoCodec[0:H264]}
(webrtcvideoengine2.cc:757): Using codec: VideoCodec[125:H264]
(videotoolboxvideocodecfactory.cc:39): Creating HW encoder for H264
(h264_video_toolbox_encoder.mm:351): Using profile H264_Baseline_3_1
(webrtcvideoengine2.cc:1762): RecreateWebRtcStream (send) because of SetCodec.
(video_send_stream.cc:800): VideoSendStreamInternal: {encoder_settings: {payload_name: H264, payload_type: 125, encoder: (VideoEncoder)}, rtp: {ssrcs: [272048794], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 127, red_payload_type: 100, red_rtx_payload_type: 101}, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3445604732], payload_type: 124}, c_name: kblAp4EIPh6Csu7q}, pre_encode_callback: nullptr, post_encode_callback: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
(video_send_stream.cc:1103): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.
(vie_encoder.cc:393): ConfigureEncoder requested.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(video_send_stream.cc:675): VideoSendStream::Stop
(webrtcvideoengine2.cc:805): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
(webrtcvideoengine2.cc:1157): AddRecvStream: {id:9c49c137-1f40-4622-acbd-0d03e1f9cfbe;ssrcs:[863849452,52816360];ssrc_groups:{semantics:FID;ssrcs:[863849452,52816360]};cname:mpKG5R6g/TKLwCmx;sync_label:uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB}
(videotoolboxvideocodecfactory.cc:102): Creating HW decoder for H264
(videotoolboxvideocodecfactory.cc:105): No HW decoder found for codec VP8
(videotoolboxvideocodecfactory.cc:105): No HW decoder found for codec VP9
(video_send_stream.cc:938): VideoSendStream::Stop
(quality_scaler.cc:66): Created CheckQPTask. Scheduling on queue...
(quality_scaler.cc:115): QP thresholds: low: 28, high: 39
(video_receive_stream.cc:206): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 125, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 863849452, local_ssrc: 272048794, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 127, red_payload_type: 100, red_rtx_payload_type: 101}, rtx_ssrc: 52816360, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), 125 (apt) -> 124 (pt), }, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB, pre_decode_callback: nullptr, target_delay_ms: 0}
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(channel.cc:1453): Add remote ssrc: 863849452
(webrtcvideoengine2.cc:1000): SetSend: false
(video_send_stream.cc:675): VideoSendStream::Stop
(channel.cc:2042): Changing video state, send=0
(webrtcvideoengine2.cc:1449): OnReadyToSend: Not ready.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(video_send_stream.cc:938): VideoSendStream::Stop
(webrtcsession.cc:1422): Changing IceConnectionState 0 => 1
2017-05-16 21:18:44.732995+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeIceConnectionState: 181] ICE state changed: RTCIceConnectionStateChecking
(webrtcsession.cc:662): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.
(webrtcvideoengine2.cc:1272): SetSink: ssrc:863849452 (ptr)
2017-05-16 21:18:44.735312+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:1062608808 1 udp 2122260223 169.254.229.79 62401 typ host generation 0 ufrag TG0t network-id 2","sdpMid":"audio","sdpMLineIndex":0}
2017-05-16 21:18:44.735543+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:1749024283 1 udp 2122194687 192.168.0.64 63560 typ host generation 0 ufrag TG0t network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
2017-05-16 21:18:44.735700+0200 Test[829:176559] [ARLWebSocketChannel.m createKeepAliveTimer 282] keepalive
2017-05-16 21:18:44.747346+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:2653915087 1 udp 1685987071 83.219.136.180 3148 typ srflx raddr 192.168.0.64 rport 63560 generation 0 ufrag TG0t network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
2017-05-16 21:18:44.747539+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:1909842776 1 tcp 1518280447 169.254.229.79 9 typ host tcptype active generation 0 ufrag TG0t network-id 2","sdpMid":"audio","sdpMLineIndex":0}
2017-05-16 21:18:44.747653+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:649855723 1 tcp 1518214911 192.168.0.64 9 typ host tcptype active generation 0 ufrag TG0t network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
(vie_encoder.cc:579): Video frame parameters changed: dimensions=640x480, rotation=90, texture=1
(video_sender.cc:129): max bitrate 1700 start bitrate 300 max frame rate 60 max payload size 1200
2017-05-16 21:18:44.832367+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:1829204744 1 udp 41820159 94.244.16.46 57472 typ relay raddr 83.219.136.180 rport 3148 generation 0 ufrag TG0t network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
2017-05-16 21:18:44.942400+0200 Test[829:176559] [ARLWebSocketChannel.m webSocket:didReceiveMessage: 216] {"candidate":"candidate:595890168 1 udp 25042687 94.244.16.46 49444 typ relay raddr 83.219.136.180 rport 6950 generation 0 ufrag TG0t network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
(video_receive_stream.cc:493): No decodable frame in 3000 ms, requesting keyframe.
(rtcp_sender.cc:713): Can't send rtcp if it is disabled.
2017-05-16 21:18:49.691216+0200 Test[829:176559] [ARLWebSocketChannel.m createKeepAliveTimer 282] keepalive
(video_receive_stream.cc:493): No decodable frame in 3000 ms, requesting keyframe.
(rtcp_sender.cc:713): Can't send rtcp if it is disabled.
2017-05-16 21:18:50.883520+0200 Test[829:176559] [ARLWebSocketChannel.m sendMessage: 125] {
"type" : "bye"
}
2017-05-16 21:18:50.883955+0200 Test[829:176559] [ARLRoomServerClientImpl.m webSocketChannel:didChangeState: 180]
(webrtcsession.cc:844): Session:8610189052997493170 Old state:STATE_INPROGRESS New state:STATE_CLOSED
2017-05-16 21:18:50.895623+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeIceConnectionState: 181] ICE state changed: RTCIceConnectionStateClosed
2017-05-16 21:18:50.896748+0200 Test[829:176616] [ARLConnectionImpl.m stringForRTCSignalingState: 352] unexpected RTCSignalingState: 5
2017-05-16 21:18:50.897335+0200 Test[829:176616] [ARLConnectionImpl.m peerConnection:didChangeSignalingState: 149] Signaling state changed:
(webrtcvideoengine2.cc:1272): SetSink: ssrc:863849452 nullptr
(webrtcvideoengine2.cc:1272): SetSink: ssrc:863849452 nullptr
(channel.cc:963): Channel disabled
(webrtcvideoengine2.cc:1000): SetSend: false
(video_send_stream.cc:675): VideoSendStream::Stop
(video_send_stream.cc:938): VideoSendStream::Stop
(channel.cc:2042): Changing video state, send=0
(video_send_stream.cc:675): VideoSendStream::Stop
(video_send_stream.cc:938): VideoSendStream::Stop
(quality_scaler.cc:72): Stopping QP Check task.
(video_send_stream.cc:938): VideoSendStream::Stop
(video_send_stream.cc:892): ~VideoSendStreamInternal: {encoder_settings: {payload_name: H264, payload_type: 125, encoder: (VideoEncoder)}, rtp: {ssrcs: [272048794], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 127, red_payload_type: 100, red_rtx_payload_type: 101}, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3445604732], payload_type: 124}, c_name: kblAp4EIPh6Csu7q}, pre_encode_callback: nullptr, post_encode_callback: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(video_receive_stream.cc:236): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 125, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 863849452, local_ssrc: 272048794, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 127, red_payload_type: 100, red_rtx_payload_type: 101}, rtx_ssrc: 52816360, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), 125 (apt) -> 124 (pt), }, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: uDAi770C24jcDyvguH8IPXqLqOxgfhYwyodB, pre_decode_callback: nullptr, target_delay_ms: 0}
(channel.cc:193): Destroyed channel: video
(channel.cc:963): Channel disabled
(channel.cc:1784): Changing voice state, recv=0 send=0
(webrtcvoiceengine.cc:1619): WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel
(webrtcvoiceengine.cc:2168): RemoveSendStream: 3150167243
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(webrtcvoiceengine.cc:2185): Removing audio send stream 3150167243 with VoiceEngine channel #0.
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(call.cc:1050): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:168): SignalNetworkState Down
(paced_sender.cc:278): PacedSender paused.
(audio_send_stream.cc:111): ~AudioSendStream: {rtp: {ssrc: 3150167243, extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], nack: {rtp_history_ms: 0}, c_name: kblAp4EIPh6Csu7q}, send_transport: (Transport), voe_channel_id: 0, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: 48000, cng_payload_type: -1, cng_plfreq: -1, min_ptime: 10, max_ptime: 120, codec_inst: {pltype: 111, plname: "opus", plfreq: 48000, pacsize: 960, channels: 1, rate: 32000}}}
(audio_device_impl.cc:1426): StopRecording
(audio_device_ios.mm:251): AudioDeviceIOS::StopRecording
(audio_device_impl.cc:1430): output: 0
(audio_device_impl.cc:1384): StopPlayout
(audio_device_ios.mm:216): AudioDeviceIOS::StopPlayout
(audio_device_impl.cc:1388): output: 0
(channel.cc:193): Destroyed channel: audio
(opensslstreamadapter.cc:894): Cleanup
(paced_sender.cc:477): ProcessThreadAttached 0x0x0
(paced_sender.cc:477): ProcessThreadAttached 0x0x0
(messagequeue.cc:542): Message took 74ms to dispatch. Posted from: Close@../../webrtc/api/peerconnectionproxy.h:108
(rtpsender.cc:275): ClearAudioSend: No audio channel exists.
(rtpsender.cc:463): SetVideoSend: No video channel exists.
(webrtcsession.cc:528): Session: 8610189052997493170 is destroyed.
(dtmfsender.cc:230): The Dtmf provider is deleted. Clear the sending queue.
What's the problem here?
Thanks.
Anton Tutarinov
May 17, 2017, 5:54:53 AM5/17/17
to discuss-webrtc
The problem is solved. The reason was in the incorrect turn configuration.
среда, 17 мая 2017 г., 9:43:55 UTC+2 пользователь Anton Tutarinov написал:
среда, 17 мая 2017 г., 9:43:55 UTC+2 пользователь Anton Tutarinov написал:
...
Apparao Mulpuri
Jun 5, 2017, 11:09:52 AM6/5/17
to discuss-webrtc
Hi Anton,
Is this the configuration issue in iOS side code base? I am also facing this issue. It would greatly help, what exactly the change was.
- Apparao
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