astaikos316
commented
4 years ago @OceanJune what Android/iOS app are you trying to use? It does sound like a codec negotiation mismatch.
Open OceanJune opened 4 years ago
astaikos316
commented
4 years ago @OceanJune what Android/iOS app are you trying to use? It does sound like a codec negotiation mismatch.
OceanJune
commented
4 years ago @OceanJune what Android/iOS app are you trying to use? It does sound like a codec negotiation mismatch.
Hi @astaikos316 , thanks for the response. Both the Android/iOS apps with this feature are under development and has not been released yet. Also the video works fine between these apps and browser too, the problem arises only between Android/iOS with UWP. I can furnish any required details regarding the apps to debug the issue. Would setting the PreferredVideoCodec to any specific value sort out the codec mismatch? Tried VP8,VP9,H264 but no luck. Is there anything else that would resolve it? Thanks again for helping out.
drejx
commented
4 years ago Hi,
I don't have a smoking gun answer for you, but a suggestion. I don't recall, but do the Nuget packages include debug versions of the libs? If not it is probably worthwhile to build the MixedReality libs from source code so you can get debug libs and as a result get all the debugging information on the webrtc side. It's a lot, but hopefully there is some useful information for your case.
Best of luck! Andrej
OceanJune
commented
4 years ago Thanks Andrej, I'll dig in and try to find the more details on it.
marcodiniz
commented
4 years ago I'm facing the exact same issue. I had built from master.
My UWP is running on a robot with Windows IoT core. It works fine from chorme desktop, android chrome and safari desktop. But on iOS I get only audio from iOS -> UWP, no video and no audio from UWP -> iOS. Datachannel works fine.
Things I've tried: -Setting video codec preference to H264 and VP8 on UWP; -Setting video codec preference to profile 42e01f on iOS (I read somewhere that's is the H264 profile iOS supports); -Loading adapter.js on the page;
Also it worked from iOS -> chrome, so I thinks it's note iOS not supporting webrtc.
OceanJune
commented
4 years ago Hi,
I tried building from the source but the test app MixedReality.WebRTC.TestAppUWP crashes soon after it is deployed (x64 Debug configuration)at PeerConnectionInterop.EnumVideoCaptureDevicesAsync( PeerConnectionInterop.VideoCaptureDevice_EnumCallback, userData, PeerConnectionInterop.VideoCaptureDevice_EnumCompletedCallback, userData);
at GetVideoCaptureDevicesAsync
at PeerConnection.cs
I believe it happens whenever the app tries to access c code. Any thoughts on what could be missing?
djee-ms
commented
4 years ago Hi @OceanJune,
Any number of things, but for start is mrwebrtc.dll at least correctly deployed and found? Otherwise it's difficult to tell without more information. You can try to run some other code first (like PeerConnection.InitializeAsync()) to rule out a problem with video, and confirm this is a problem with the first call to the library (and so likely with C# not finding the DLL).
OceanJune
commented
4 years ago Hello @djee-ms! Many thanks for the prompt reply.
Yes, the mrwebrtc.dll is found under '\bin\UWP\x64\Debug' & 'bin\netstandard2.0\Debug' and so is Microsoft.MixedReality.WebRTC.dll.
I tried to create the peer connection on a button click to confirm the issue and the app crashes whenever it tries to make a call to the library, in this case on
uint res = PeerConnectionInterop.PeerConnection_Create(nativeConfig, out _nativePeerhandle);
Is there a way to sort this out?
djee-ms
commented
4 years ago Then if it crashes on any call I expect it's not found. You're on NuGet 1.0.3 or from a local build (and if so, which commit)?
OceanJune
commented
4 years ago I'm using the master branch at this commit. The same happens even with an older commit
OceanJune
commented
4 years ago I'd be really grateful for any assistance on this issue, inputs on where to look or what to check. I've tried to fix every possible cause I could think of but still can't seem to find out where it is going wrong.
djee-ms
commented
4 years ago So if I summarize:
mrwebrtc.dll builtThe next steps would be to:
mrwebrtc.dll and Microsoft.MixedReality.WebRTC.dll are deployed to the same folder.Unfortunately DLL loading errors (due to the DLL being absent itself, or due another DLL it depends on being missing) is painfully annoying to debug, as the OS does not provide any context information on what caused the DLL to fail. Your only hints are those "module loaded" lines from the VS output window, and if you see mrwebrtc.dll being loaded then try to guess which other DLL it depends on is missing. Often this is due to Debug-only dependencies; so I would suggest also to try if the app crashes both in Debug and Release. I know we had back in the days some issue with Debug build due to missing Debug-only dependencies on device that wouldn't get always installed, and although that should be fixed, it's worth double-checking this is not the issue here. On the other hand if you cannot see mrwebrtc.dll then it's easier, it means the OS cannot find the DLL, so it's a problem while deploying it (wrong path, or not copied correctly or at all). Sometimes uninstalling via Device Portal and reinstalling via VS may help.
Sorry that it is all so difficult, those deploy issues shouldn't happen. 😢
OceanJune
commented
4 years ago Thank you very much @djee-ms ! The crash doesn't occur in Release mode, though the actual issue of android/iOS stream callback not being received still persists :( I'm not using HoloLens 2 but just a Dell Precision, not sure if that makes a difference. I'm trying to find out what could be going wrong with the available information and hope to figure it out soon 🤞 Thanks a million for the help!! 🙏 😊
djee-ms
commented
4 years ago HoloLens 2 vs. a PC shouldn't make much difference as long as you stay on UWP. The codepath for Windows Desktop (Win32) is quite different though. But TestAppUWP is UWP so should be the same.
If you deploy on a PC then it's much easier to confirm that the mrwebrtc.dll is indeed at the right place; it should be deployed by Visual Studio into the AppX folder. Note that UPW apps need to be deployed on the PC and won't use the DLLs in bin/ but instead have their own copy. During debugging with Visual Studio however a temporary deploy folder is used instead of the regular system one, which is under bin/UWP/x64/Debug/AppX for example, and inside this AppX folder you should have a copy of mrwebrtc.dll alongside TestAppUWP.exe.
OceanJune
commented
4 years ago The mrwebrtc.dll is indeed available under bin/UWP/x64/Debug/AppX along with TestAppUWP.exe . Is there something specific or obvious to lookout for with the dependent dlls?
fibann
commented
3 years ago @OceanJune the problem with the Debug build crashing might be related to the debug Visual C++ runtime not being available on the machine you have deployed your app to. A quick-and-dirty workaround for this is to copy the dlls in C:\Program Files (x86)\Microsoft Visual Studio\<your_version>\Enterprise\VC\Redist\MSVC\<your_version>\debug_nonredist\x64\Microsoft.VC141.DebugCRT to bin/UWP/x64/Debug/AppX. If this is the problem it should be enough to get you going.
I am not sure about the original issue, the SDP messages look fine to me and the fact that you get an OnTrack event should mean that the negotiation succeeds. Just to be sure that the problem is not in the mobile app, have you double-checked that video capture is working properly on the phone? Have you verified that the mobile app streams frames correctly to a different WebRTC implementation?
OceanJune
commented
3 years ago Hi @fibann , the video from the mobile app plays without issues in the web browser, so I doubt it has anything to do with the mobile implementation.
I tried copying the dlls to the Appx folder, but the app still crashes on the first call to c++ library. Would reinstalling the c++ related workload in Visual Studio help by any chance?
fibann
commented
3 years ago Would reinstalling the c++ related workload in Visual Studio help by any chance?
I don't think so.
First I'd double-check if this is actually a missing DLL problem. Can you attach the VS debug output for a run of the Debug build and/or look for errors about DLL loading?
Second, I'd double-check that you are moving the DLLs to the correct folder - as djee-ms wrote above, the launch folder varies depending on the deploy process. Look for TestAppUWP in the Task Manager and check its containing folder.
If the dll is indeed failing to load despite the debug VCRT being in the correct folder, you can try to parse the mrwebrtc dll with Dependencies or similar to see what dependency is missing.
From what you described it seems that the main issue might be the WebRTC-UWP implementation failing somewhere in the video receiving process - there is not much we can do without more info unfortunately, your best bet is getting the Debug build running and looking at the WebRTC logs for video errors.
xiemeilong
commented
3 years ago Video stream from android is also not rendered in unity. I don't know whether is it the same issue or not.
OceanJune
commented
3 years ago @djee-ms @fibann The debug build of version 1.0.3 started working out of the blue after signing up for the Windows Insider program (still no luck with version 2.x.x). I'm hoping the attached WebRTC logs has some clue about the issue of remote video stream from Android(/iOS) app not being rendered in the UWP app that uses the v1.0.3 of mixed reality WebRTC. The audio works fine both ways, also the video stream is received when the call is made from a browser of the Android device. Something amiss?
onecore\base\AppModel\Runtime\Src\PackagePath.hpp(144)\kernelbase.dll!00007FFE5AB1DE88: (caller: 00007FFE5AB1B13E) ReturnHr(3) tid(b51c) 80073D5B The package does not have a mutable directory.
onecore\base\AppModel\Runtime\Src\PackagePath.hpp(144)\kernelbase.dll!00007FFE5AB1DE88: (caller: 00007FFE5AB1B13E) ReturnHr(4) tid(b51c) 80073D5B The package does not have a mutable directory.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\Windows.Web.dll'.
'TestAppUwp.exe' (Win32): Loaded 'E:\MRRTCLatest\bin\UWP\x64\Debug\AppX\mrwebrtc.dll'. Symbols loaded.
'TestAppUwp.exe' (Win32): Loaded 'E:\MRRTCLatest\bin\UWP\x64\Debug\AppX\ucrtbased.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\MMDevAPI.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Program Files\WindowsApps\Microsoft.VCLibs.140.00.Debug_14.0.27810.0_x64__8wekyb3d8bbwe\vcruntime140d_app.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Program Files\WindowsApps\Microsoft.VCLibs.140.00.Debug_14.0.27810.0_x64__8wekyb3d8bbwe\msvcp140d_app.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\mfplat.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\mfreadwrite.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\Windows.Media.Devices.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\devobj.dll'.
(audio_processing_impl.cc:423): Capture analyzer activated: 0
Capture post processor activated: 0
Render pre processor activated: 0
(webrtcvoiceengine.cc:204): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtcvideoengine.cc:473): WebRtcVideoEngine::WebRtcVideoEngine()
(webrtcvoiceengine.cc:227): WebRtcVoiceEngine::Init
(webrtcvoiceengine.cc:234): Supported send codecs in order of preference:
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\userenv.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\profext.dll'.
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\profext.dll'
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\userenv.dll'
(webrtcvoiceengine.cc:237): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtcvoiceengine.cc:237): ISAC/16000/1 (103)
(webrtcvoiceengine.cc:237): ISAC/32000/1 (104)
(webrtcvoiceengine.cc:237): G722/8000/1 (9)
(webrtcvoiceengine.cc:237): ILBC/8000/1 (102)
(webrtcvoiceengine.cc:237): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:237): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:237): CN/32000/1 (106)
(webrtcvoiceengine.cc:237): CN/16000/1 (105)
(webrtcvoiceengine.cc:237): CN/8000/1 (13)
(webrtcvoiceengine.cc:237): telephone-event/48000/1 (110)
(webrtcvoiceengine.cc:237): telephone-event/32000/1 (112)
(webrtcvoiceengine.cc:237): telephone-event/16000/1 (113)
(webrtcvoiceengine.cc:237): telephone-event/8000/1 (126)
(webrtcvoiceengine.cc:240): Supported recv codecs in order of preference:
(webrtcvoiceengine.cc:243): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
(webrtcvoiceengine.cc:243): ISAC/16000/1 (103)
(webrtcvoiceengine.cc:243): ISAC/32000/1 (104)
(webrtcvoiceengine.cc:243): G722/8000/1 (9)
(webrtcvoiceengine.cc:243): ILBC/8000/1 (102)
(webrtcvoiceengine.cc:243): PCMU/8000/1 (0)
(webrtcvoiceengine.cc:243): PCMA/8000/1 (8)
(webrtcvoiceengine.cc:243): CN/32000/1 (106)
(webrtcvoiceengine.cc:243): CN/16000/1 (105)
(webrtcvoiceengine.cc:243): CN/8000/1 (13)
(webrtcvoiceengine.cc:243): telephone-event/48000/1 (110)
(webrtcvoiceengine.cc:243): telephone-event/32000/1 (112)
(webrtcvoiceengine.cc:243): telephone-event/16000/1 (113)
(webrtcvoiceengine.cc:243): telephone-event/8000/1 (126)
(audio_device_buffer.cc:64): AudioDeviceBuffer::ctor
(audio_device_buffer.cc:181): SetRecordingSampleRate(0)
(audio_device_buffer.cc:187): SetPlayoutSampleRate(0)
(audio_device_buffer.cc:201): SetRecordingChannels(0)
(audio_device_buffer.cc:207): SetPlayoutChannels(0)
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\wintrust.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\Windows.Devices.Enumeration.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\DevDispItemProvider.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\DDORes.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\DefaultDeviceManager.dll'.
(impl_webrtc_audiodevicewasapi.cpp:1082): Using communications audio playout device: Speakers/Headphones (Realtek(R) Audio)
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\AudioSes.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\ResourcePolicyClient.dll'.
(impl_webrtc_audiodevicewasapi.cpp:1110): Output audio device activated Speakers/Headphones (Realtek(R) Audio)
(audio_device_buffer.cc:187): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:207): SetPlayoutChannels(2)
(audio_device_buffer.cc:207): SetPlayoutChannels(2)
(impl_webrtc_audiodevicewasapi.cpp:1211): Using communications audio capture device: Microphone (Realtek(R) Audio)
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\mlang.dll'.
(impl_webrtc_audiodevicewasapi.cpp:1239): Input audio device activated Microphone (Realtek(R) Audio)
(audio_device_buffer.cc:181): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:201): SetRecordingChannels(2)
(audio_device_buffer.cc:201): SetRecordingChannels(2)
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_device_buffer.cc:82): webrtc::AudioDeviceBuffer::RegisterAudioCallback
(webrtcvoiceengine.cc:316): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, typing: 1, experimental_agc: 0, extended_filter_aec: 0, delay_agnostic_aec: 0, experimental_ns: 0, residual_echo_detector: 1, }
(webrtcvoiceengine.cc:422): Disabling EC since built-in EC will be used instead
(audio_processing_impl.cc:688): Highpass filter activated: 0
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
(apm_helpers.cc:114): Echo control set to 0 with mode 0
(apm_helpers.cc:124): EC metrics set to 0
The thread 0x6904 has exited with code 0 (0x0).
(apm_helpers.cc:81): AGC set to 1 with mode 0
(webrtcvoiceengine.cc:476): Disabling NS since built-in NS will be used instead
(apm_helpers.cc:149): NS set to 0
(webrtcvoiceengine.cc:484): Stereo swapping enabled? 0
(webrtcvoiceengine.cc:489): NetEq capacity is 50
(webrtcvoiceengine.cc:495): NetEq fast mode? 0
(webrtcvoiceengine.cc:502): Typing detection is enabled? 1
(apm_helpers.cc:163): VAD set to 1 for typing detection.
(webrtcvoiceengine.cc:513): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:523): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:533): Experimental ns is enabled? 0
(audio_processing_impl.cc:688): Highpass filter activated: 1
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
(messagequeue.cc:513): Message took 1591ms to dispatch. Posted from: cricket::ChannelManager::Init@../../pc/channelmanager.cc:133
(messagequeue.cc:513): Message took 1592ms to dispatch. Posted from: webrtc::CreateModularPeerConnectionFactory@../../pc/peerconnectionfactory.cc:104
(peerconnectionfactory.cc:479): Using default network controller factory
(alr_experiment.cc:65): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
(bitrate_prober.cc:64): Bandwidth probing enabled, set to inactive
(rtp_transport_controller_send.cc:48): Using Legacy SSCC
(aimd_rate_control.cc:108): Using aimd rate control with back off factor 0.85
(delay_based_bwe.cc:104): Using Trendline filter for delay change estimation with window size 20
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(delay_based_bwe.cc:312): BWE Setting start bitrate to: 300000
(paced_sender.cc:366): ProcessThreadAttached 0x9c30b690
(cpu_info.cc:46): Available number of cores: 12
(aimd_rate_control.cc:108): Using aimd rate control with back off factor 0.85
(remote_bitrate_estimator_single_stream.cc:55): RemoteBitrateEstimatorSingleStream: Instantiating.
(messagequeue.cc:513): Message took 93ms to dispatch. Posted from: webrtc::PeerConnectionFactory::CreatePeerConnection@../../pc/peerconnectionfactory.cc:379
(paced_sender.cc:95): PacedSender paused.
(opensslidentity.cc:45): Making key pair
(messagequeue.cc:513): Message took 174ms to dispatch. Posted from: webrtc::PeerConnectionFactoryProxyWithInternal::CreatePeerConnection@D:\azp2\1\s\external\webrtc-uwp-sdk\webrtc\xplatform\webrtc\api/peerconnectionfactoryproxy.h:43
(opensslidentity.cc:93): Returning key pair
(opensslcertificate.cc:58): Making certificate for WebRTC
(opensslcertificate.cc:105): Returning certificate
(p2ptransportchannel.cc:455): Set backup connection ping interval to 25000 milliseconds.
(p2ptransportchannel.cc:464): Set ICE receiving timeout to 2500 milliseconds
(p2ptransportchannel.cc:471): Set ping most likely connection to 0
(p2ptransportchannel.cc:478): Set stable_writable_connection_ping_interval to 2500
(p2ptransportchannel.cc:491): Set presume writable when fully relayed to 0
(p2ptransportchannel.cc:500): Set regather_on_failed_networks_interval to 300000
(p2ptransportchannel.cc:519): Set receiving_switching_delay to 1000
(dtlssrtptransport.cc:60): Setting RTCP Transport on audio transport 0
(dtlssrtptransport.cc:65): Setting RTP Transport on audio transport 9c33ba90
(p2ptransportchannel.cc:455): Set backup connection ping interval to 25000 milliseconds.
(p2ptransportchannel.cc:464): Set ICE receiving timeout to 2500 milliseconds
(p2ptransportchannel.cc:471): Set ping most likely connection to 0
(p2ptransportchannel.cc:478): Set stable_writable_connection_ping_interval to 2500
(p2ptransportchannel.cc:491): Set presume writable when fully relayed to 0
(p2ptransportchannel.cc:500): Set regather_on_failed_networks_interval to 300000
(p2ptransportchannel.cc:519): Set receiving_switching_delay to 1000
(dtlssrtptransport.cc:60): Setting RTCP Transport on video transport 0
(dtlssrtptransport.cc:65): Setting RTP Transport on video transport 9bbab320
(p2ptransportchannel.cc:403): Received remote ICE parameters: ufrag=f5DQ, renomination enabled
(p2ptransportchannel.cc:403): Received remote ICE parameters: ufrag=f5DQ, renomination enabled
(peerconnection.cc:2730): Adding audio transceiver for MID=audio at i=0 in response to the remote description.
(webrtcvoiceengine.cc:1471): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:316): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:489): NetEq capacity is 50
(webrtcvoiceengine.cc:495): NetEq fast mode? 0
(webrtcvoiceengine.cc:513): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:523): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:533): Experimental ns is enabled? 0
(audio_processing_impl.cc:688): Highpass filter activated: 1
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
(webrtcvoiceengine.cc:1489): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(channel.cc:114): Created channel for audio
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(peerconnection.cc:2730): Adding video transceiver for MID=video at i=1 in response to the remote description.
(webrtcvideoengine.cc:493): CreateChannel. Options: VideoOptions {}
(channel.cc:114): Created channel for video
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(peerconnection.cc:3512): Session: 8067918082610112391 Old state: kStable New state: kHaveRemoteOffer
(channel.cc:838): Setting remote voice description
(webrtcvoiceengine.cc:1293): 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[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}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}], max_bandwidth_bps: -1, mid: audio, options: AudioOptions {}}
(webrtcvoiceengine.cc:1688): Recreate all the receive streams because the send codec has changed.
(webrtcvoiceengine.cc:2084): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
(webrtcvoiceengine.cc:1471): Setting voice channel options: AudioOptions {}
(webrtcvoiceengine.cc:316): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:489): NetEq capacity is 50
(webrtcvoiceengine.cc:495): NetEq fast mode? 0
(webrtcvoiceengine.cc:513): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:523): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:533): Experimental ns is enabled? 0
(audio_processing_impl.cc:688): Highpass filter activated: 1
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
(webrtcvoiceengine.cc:1489): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:1833): AddRecvStream: {id:AVcall_local_audio_track;ssrcs:[3953683229];ssrc_groups:;cname:VZlgqObc5Oamy/zX;stream_ids:ANDROID_AV_STREAM_1;}
(neteq_impl.cc:114): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, enable_fast_accelerate=false, enable_muted_state= true
(audio_coding_module.cc:451): Created
(audio_receive_stream.cc:106): AudioReceiveStream: 3953683229
(audio_receive_stream.cc:327): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3953683229, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), sync_group: ANDROID_AV_STREAM_1}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(channel.cc:656): Add remote ssrc: 3953683229
(channel.cc:779): Changing voice state, recv=0 send=0
(channel.cc:973): Setting remote video description
(webrtcvideoengine.cc:686): SetSendParameters: {codecs: [VideoCodec[96:VP8]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {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://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}], max_bandwidth_bps: -1, mid: video}
(webrtcmediaengine.cc:170): Unsupported RTP extension: {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}
(webrtcvideoengine.cc:695): Using codec: VideoCodec[96:VP8]
(webrtcvideoengine.cc:744): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
(webrtcvideoengine.cc:1137): AddRecvStream: {id:AVcall_local_video_track;ssrcs:[2220358426,1637978611];ssrc_groups:{semantics:FID;ssrcs:[2220358426,1637978611]};cname:VZlgqObc5Oamy/zX;stream_ids:ANDROID_AV_STREAM_1;}
(video_receive_stream.cc:119): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, name: VP9, codec_params: {x-google-profile-id: 0}}, {decoder: (VideoDecoder), payload_type: 100, name: H264, codec_params: {packetization-mode: 1}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 104, red_type: 102, rtx_ssrc: 1637978611, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(channel.cc:656): Add remote ssrc: 2220358426
(channel.cc:909): Changing video state, send=0
(messagequeue.cc:513): Message took 58ms to dispatch. Posted from: cricket::BaseChannel::SetRemoteContent@../../pc/channel.cc:279
(peerconnection.cc:4763): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(peerconnection.cc:2362): Processing the addition of a new track for MID=audio (added to streams=[ANDROID_AV_STREAM_1]).
(webrtcvoiceengine.cc:1942): SetOutputVolume() to 1 for recv stream with ssrc 3953683229
(peerconnection.cc:2362): Processing the addition of a new track for MID=video (added to streams=[ANDROID_AV_STREAM_1]).
(webrtcvideoengine.cc:1267): SetSink: ssrc:2220358426 (ptr)
(peer_connection.cpp:963): Added transceiver mid=#audio of type 'audio' with desired direction kRecvOnly
(peer_connection.cpp:972): Send with NULL track
(peer_connection.cpp:979): Recv with track #AVcall_local_audio_track enabled=true
(peer_connection.cpp:915): Added receiver #AVcall_local_audio_track of type 0
(peer_connection.cpp:918): + Track #AVcall_local_audio_track with stream #ANDROID_AV_STREAM_1
(peer_connection.cpp:963): Added transceiver mid=#video of type 'video' with desired direction kRecvOnly
(peer_connection.cpp:972): Send with NULL track
(peer_connection.cpp:979): Recv with track #AVcall_local_video_track enabled=true
(peer_connection.cpp:915): Added receiver #AVcall_local_video_track of type 1
(peer_connection.cpp:918): + Track #AVcall_local_video_track with stream #ANDROID_AV_STREAM_1
(peer_connection.cpp:786): Added stream #ANDROID_AV_STREAM_1 with 1 audio tracks and 1 video tracks.
(peer_connection.cpp:202): Remote description successfully set.
(peer_connection.cpp:1194): Synchronizing 2 RTP transceivers with 2 transceiver wrappers (remote = 1).
(messagequeue.cc:513): Message took 210ms to dispatch. Posted from: webrtc::PeerConnectionProxyWithInternal::SetRemoteDescription@D:\azp2\1\s\external\webrtc-uwp-sdk\webrtc\xplatform\webrtc\api/peerconnectionproxy.h:107
(peerconnection.cc:4785): Local and Remote descriptions must be applied to get the SSL Role of the session.
(peerconnection.cc:4785): Local and Remote descriptions must be applied to get the SSL Role of the session.
(impl_webrtc_videocapturer.cpp:1172): Using local detection for orientation source
(impl_webrtc_videocapturer.cpp:1232): Init called for device \\?\USB#VID_0C45&PID_671D&MI_00#6&4b9a392&0&0000#{e5323777-f976-4f5b-9b55-b94699c46e44}\global
(webrtcvideoengine.cc:2524): VideoReceiveStream stats: 1603716918410, {ssrc: 2220358426, total_bps: 0, width: 0, height: 0, key: 0, delta: 0, network_fps: 0, decode_fps: 0, render_fps: 0, decode_ms: 0, max_decode_ms: 0, cur_delay_ms: 0, targ_delay_ms: 0, jb_delay_ms: 0, min_playout_delay_ms: 0, discarded: 0, sync_offset_ms: 2147483647, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0}
(webrtcvideoengine.cc:1313): Call stats: 1603716918410, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
(dtlssrtptransport.cc:60): Setting RTCP Transport on audio transport 0
(dtlssrtptransport.cc:65): Setting RTP Transport on audio transport 9c33ba90
(p2ptransportchannel.cc:392): Set ICE ufrag: otZH pwd: wJm8xzkNmIQ6lbmLdkbsoSZv on transport audio
(dtlstransport.cc:376): DtlsTransport[audio|1|__]: DTLS setup complete.
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(rtptransceiver.cc:174): Changing transceiver (MID=audio) current direction from to kSendRecv.
(rtptransceiver.cc:174): Changing transceiver (MID=video) current direction from to kSendRecv.
(impl_webrtc_videocapturer.cpp:529): CaptureDevice::Initialize(\\?\USB#VID_0C45&PID_671D&MI_00#6&4b9a392&0&0000#{e5323777-f976-4f5b-9b55-b94699c46e44}\global, w=0, h=0, f=0)
(channel.cc:530): Channel enabled
(channel.cc:779): Changing voice state, recv=0 send=0
(channel.cc:530): Channel enabled
(channel.cc:909): Changing video state, send=0
(peerconnection.cc:3512): Session: 8067918082610112391 Old state: kHaveRemoteOffer New state: kStable
(channel.cc:787): Setting local voice description
(webrtcvoiceengine.cc:1330): 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:1499): Setting receive voice codecs.
(audio_receive_stream.cc:327): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3953683229, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), sync_group: ANDROID_AV_STREAM_1}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(audio_receive_stream.cc:129): ~AudioReceiveStream: 3953683229
(neteq_impl.cc:114): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, enable_fast_accelerate=false, enable_muted_state= true
(audio_coding_module.cc:451): Created
(audio_receive_stream.cc:106): AudioReceiveStream: 3953683229
(audio_receive_stream.cc:327): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3953683229, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), sync_group: ANDROID_AV_STREAM_1}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(webrtcvoiceengine.cc:1770): AddSendStream: {id:8289eb69-8c56-4376-86d0-0a983a598651;ssrcs:[1392525940];ssrc_groups:;cname:uKW4QFepWmYz77ac;stream_ids:;}
(neteq_impl.cc:114): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=0, enable_fast_accelerate=false, enable_muted_state= true
(audio_coding_module.cc:451): Created
(audio_send_stream.cc:137): AudioSendStream: 1392525940
(audio_send_stream.cc:199): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1392525940, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], nack: {rtp_history_ms: 0}, c_name: uKW4QFepWmYz77ac}, send_transport: (Transport), min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: , payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(audio_receive_stream.cc:327): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3953683229, local_ssrc: 1392525940, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), sync_group: ANDROID_AV_STREAM_1}
(channel.cc:610): Add send stream ssrc: 1392525940
(impl_webrtc_audiodevicewasapi.cpp:1082): Using communications audio playout device: Speakers/Headphones (Realtek(R) Audio)
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\MFCaptureEngine.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\mfcore.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\CompPkgSup.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\CapabilityAccessManagerClient.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\UserMgrProxy.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\usermgrcli.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\FrameServerClient.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\mfsensorgroup.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\avrt.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\Windows.Media.MediaControl.dll'.
(impl_webrtc_audiodevicewasapi.cpp:1110): Output audio device activated Speakers/Headphones (Realtek(R) Audio)
(audio_device_buffer.cc:187): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:207): SetPlayoutChannels(2)
(audio_device_buffer.cc:187): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:207): SetPlayoutChannels(2)
(audio_device_buffer.cc:99): webrtc::AudioDeviceBuffer::StartPlayout
(channel.cc:779): Changing voice state, recv=1 send=0
(messagequeue.cc:513): Message took 212ms to dispatch. Posted from: cricket::BaseChannel::SetLocalContent@../../pc/channel.cc:270
(channel.cc:922): Setting local video description
(audio_device_buffer.cc:288): Size of playout buffer: 960
(webrtcvideoengine.cc:922): SetRecvParameters: {codecs: [VideoCodec[96:VP8]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {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://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}]}
(webrtcvideoengine.cc:937): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:H264]} to {VideoCodec[96:VP8]}
(webrtcvideoengine.cc:2377): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters
(webrtcvideoengine.cc:2382): RecreateWebRtcVideoStream (recv) because of SetRecvParameters
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(video_receive_stream.cc:162): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, name: VP9, codec_params: {x-google-profile-id: 0}}, {decoder: (VideoDecoder), payload_type: 100, name: H264, codec_params: {packetization-mode: 1}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 104, red_type: 102, rtx_ssrc: 1637978611, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
The thread 0x5920 has exited with code 0 (0x0).
The thread 0x8010 has exited with code 0 (0x0).
(receive_statistics_proxy.cc:487): Frames decoded 0
WebRTC.Video.DroppedFrames.Receiver 0
(video_receive_stream.cc:119): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: -1, red_type: -1, rtx_ssrc: 1637978611, rtx_payload_types: {-1 (pt) -> 96 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {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:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
WinUWPH264DecoderImpl::Release()
WinUWPH264DecoderImpl::~WinUWPH264DecoderImpl()
WinUWPH264DecoderImpl::Release()
(webrtcvideoengine.cc:1041): AddSendStream: {id:cb63e1f0-35a3-4a81-ac1b-c836e2288bee;ssrcs:[939973652];ssrc_groups:;cname:uKW4QFepWmYz77ac;stream_ids:;}
(webrtcvideoengine.cc:1694): RecreateWebRtcStream (send) because of SetCodec.
(alr_experiment.cc:65): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
(video_send_stream_impl.cc:285): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [939973652], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {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}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: -1, red_payload_type: -1, red_rtx_payload_type: -1}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [], payload_type: -1}, c_name: uKW4QFepWmYz77ac}, rtcp: {video_report_interval_ms: 1000, audio_report_interval_ms: 5000}, pre_encode_callback: nullptr, post_encode_callback: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
(video_send_stream_impl.cc:304): ERROR: Initial encoder max bitrate = -1 which is <= 0!
(video_send_stream_impl.cc:306): Using default encoder max bitrate = 10 Mbps
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(video_stream_encoder.cc:487): ConfigureEncoder requested.
(video_send_stream.cc:150): VideoSendStream::Stop
(video_send_stream_impl.cc:442): VideoSendStream::Stop
(webrtcvideoengine.cc:1072): SetLocalSsrc on all the receive streams because we added a send stream.
(webrtcvideoengine.cc:2317): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=939973652
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(video_receive_stream.cc:162): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: -1, red_type: -1, rtx_ssrc: 1637978611, rtx_payload_types: {-1 (pt) -> 96 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {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:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
The thread 0xac4c has exited with code 0 (0x0).
The thread 0xcb2c has exited with code 0 (0x0).
(receive_statistics_proxy.cc:487): Frames decoded 0
WebRTC.Video.DroppedFrames.Receiver 0
(video_receive_stream.cc:119): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 939973652, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: -1, red_type: -1, rtx_ssrc: 1637978611, rtx_payload_types: {-1 (pt) -> 96 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {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:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(channel.cc:610): Add send stream ssrc: 939973652
(video_send_stream.cc:150): VideoSendStream::Stop
(channel.cc:909): Changing video state, send=0
(video_send_stream_impl.cc:442): VideoSendStream::Stop
(messagequeue.cc:513): Message took 265ms to dispatch. Posted from: cricket::BaseChannel::SetLocalContent@../../pc/channel.cc:270
(peerconnection.cc:4769): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.
(basicportallocator.cc:336): Start getting ports with prune_turn_ports disabled
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\DriverStore\FileRepository\43_dell_coffeelake_hws_iigd_dch.inf_amd64_b8b61605a0c26134\igd11dxva64.dll'.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\setupapi.dll'.
(basicportallocator.cc:107): Filtered out ignored networks:
(basicportallocator.cc:109): Net[Software:::1/128:Loopback:id=2]
(basicportallocator.cc:109): Net[Software:127.0.0.1/32:Loopback:id=1]
(basicportallocator.cc:825): Network manager has started
(messagequeue.cc:513): Message took 55ms to dispatch. Posted from: rtc::BasicNetworkManager::StartUpdating@../../rtc_base/network.cc:852
(basicportallocator.cc:107): Filtered out ignored networks:
(messagequeue.cc:513): Message took 550ms to dispatch. Posted from: webrtc::WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionSucceeded@../../pc/webrtcsessiondescriptionfactory.cc:451
(basicportallocator.cc:109): Net[Software:::1/128:Loopback:id=2]
(basicportallocator.cc:109): Net[Software:127.0.0.1/32:Loopback:id=1]
(basicportallocator.cc:740): Allocate ports on 1 networks
(peer_connection.cpp:1194): Synchronizing 2 RTP transceivers with 2 transceiver wrappers (remote = 0).
(basicportallocator.cc:1276): Net[Intel(R):192.168.0.194/32:Wifi:id=3]: Allocation Phase=Udp
(port.cc:322): Port[9c3ce950::1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port created with network cost 10
(basicportallocator.cc:1347): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(basicportallocator.cc:847): Adding allocated port for audio
(basicportallocator.cc:866): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Added port to allocator
(basicportallocator.cc:883): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Gathered candidate: Cand[:4252876256:1:udp:2122260223:192.168.0.194:59478:local::0:otZH:wJm8xzkNmIQ6lbmLdkbsoSZv:3:10:0]
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\DriverStore\FileRepository\43_dell_coffeelake_hws_iigd_dch.inf_amd64_b8b61605a0c26134\igdinfo64.dll'.
(basicportallocator.cc:911): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port ready.
(physicalsocketserver.cc:562): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:738): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: SetOption(5, 0) failed: 0
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\CoreMmRes.dll'. Module was built without symbols.
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\CoreMmRes.dll'
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\CoreMmRes.dll'. Module was built without symbols.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(1) tid(a2c) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(2) tid(a2c) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(3) tid(a2c) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3BB9) ReturnHr(4) tid(a2c) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3C2C) ReturnHr(5) tid(a2c) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3C78) ReturnHr(6) tid(a2c) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(1138)\MFSENSORGROUP.dll!00007FFE09AD3A87: (caller: 00007FFE09AA3CC1) ReturnHr(7) tid(a2c) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(8) tid(73a8) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(9) tid(73a8) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(273)\MFSENSORGROUP.dll!00007FFE09AD3072: (caller: 00007FFE09AA097C) ReturnHr(10) tid(73a8) 80070490 Element not found.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3BB9) ReturnHr(11) tid(73a8) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3C2C) ReturnHr(12) tid(73a8) 80070002 The system cannot find the file specified.
avcore\mf\frameserver\fsutils\fsutils.cpp(495)\MFSENSORGROUP.dll!00007FFE09AD3619: (caller: 00007FFE09AA3C78) ReturnHr(13) tid(73a8) 80070002 The system cannot find the file specified.
(basicportallocator.cc:883): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Gathered candidate: Cand[:2083896148:1:udp:1686052607:49.207.133.244:39285:stun:192.168.0.194:59478:otZH:wJm8xzkNmIQ6lbmLdkbsoSZv:3:10:0]
avcore\mf\frameserver\fsutils\fsutils.cpp(1138)\MFSENSORGROUP.dll!00007FFE09AD3A87: (caller: 00007FFE09AA3CC1) ReturnHr(14) tid(73a8) 80070002 The system cannot find the file specified.
(basicportallocator.cc:985): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port completed gathering candidates.
(messagequeue.cc:513): Message took 61ms to dispatch. Posted from: webrtc::PeerConnection::PostSetSessionDescriptionSuccess@../../pc/peerconnection.cc:3567
candidate: candidate:4252876256 1 udp 2122260223 192.168.0.194 59478 typ host generation 0 ufrag otZH network-id 3 network-cost 10 spdMid: audio spdMlineIndex: 0
(device_video_track_source.cpp:125): Using video capture device 'Integrated Webcam' (id=\\?\USB#VID_0C45&PID_671D&MI_00#6&4b9a392&0&0000#{e5323777-f976-4f5b-9b55-b94699c46e44}\global)
(device_video_track_source.cpp:130): Supported video formats:
(basicportallocator.cc:1276): Net[Intel(R):192.168.0.194/32:Wifi:id=3]: Allocation Phase=Relay
(device_video_track_source.cpp:133): - NV12 1280x720x30
(device_video_track_source.cpp:133): - MJPG 1280x720x30
(device_video_track_source.cpp:133): - NV12 960x540x30
(port.cc:322): Port[9c3b2cc0::1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port created with network cost 10
(device_video_track_source.cpp:133): - MJPG 960x540x30
(device_video_track_source.cpp:133): - NV12 848x480x30
(basicportallocator.cc:847): Adding allocated port for audio
(device_video_track_source.cpp:133): - MJPG 848x480x30
(basicportallocator.cc:866): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Added port to allocator
(device_video_track_source.cpp:133): - NV12 640x480x30
(device_video_track_source.cpp:133): - MJPG 640x480x30
(turnport.cc:336): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Trying to connect to TURN server via udp @ 119.81.194.156:443
(device_video_track_source.cpp:133): - NV12 640x360x30
(device_video_track_source.cpp:133): - MJPG 640x360x30
candidate: candidate:2083896148 1 udp 1686052607 49.207.133.244 39285 typ srflx raddr 192.168.0.194 rport 59478 generation 0 ufrag otZH network-id 3 network-cost 10 spdMid: audio spdMlineIndex: 0
(device_video_track_source.cpp:133): - YUY2 640x480x30
(device_video_track_source.cpp:133): - YUY2 640x360x30
(device_video_track_source.cpp:133): - YUY2 424x240x30
(turnport.cc:1282): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate request sent, id=7730712f4b355075517a3839
(device_video_track_source.cpp:133): - YUY2 320x240x30
(device_video_track_source.cpp:133): - YUY2 320x180x30
(device_video_track_source.cpp:133): - YUY2 160x120x30
(port.cc:322): Port[9c53e790::1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port created with network cost 10
(basicportallocator.cc:847): Adding allocated port for audio
(basicportallocator.cc:866): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Added port to allocator
(turnport.cc:336): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Trying to connect to TURN server via tcp @ 119.81.194.156:443
(impl_webrtc_videocapturer.cpp:1508): Trying to start video capture with video format Nv12 (640 x 480 @ 30 FPS)
(impl_webrtc_videocapturer.cpp:1574): Found closest video encoding format (640 x 480 @ 30 FPS)
(impl_webrtc_videocapturer.cpp:654): Starting device capture with sink video format NV12 (640 x 480 @ 30 FPS)
(messagequeue.cc:513): Message took 90ms to dispatch. Posted from: cricket::AllocationSequence::OnMessage@../../p2p/client/basicportallocator.cc:1300
(turnport.cc:1334): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Received TURN allocate error response, id=7730712f4b355075517a3839, code=401, rtt=48
(turnport.cc:1282): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate request sent, id=444c7139364d4f39556e3444
Exception thrown at 0x00007FFE5AB4276C (KernelBase.dll) in TestAppUwp.exe: WinRT originate error - 0xC00D36B3 : 'The stream number provided was invalid.'.
(turnport.cc:471): TurnPort connected to 119.81.194.156:443 using tcp.
(turnport.cc:1282): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate request sent, id=766a68457473375865374e50
(turnport.cc:1288): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate requested successfully, id=444c7139364d4f39556e3444, code=0, rtt=40
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\userenv.dll'.
(basicportallocator.cc:883): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Gathered candidate: Cand[:1218638837:1:udp:41885695:119.81.194.156:32105:relay:49.207.133.244:39285:otZH:wJm8xzkNmIQ6lbmLdkbsoSZv:3:10:0]
(basicportallocator.cc:911): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port ready.
(physicalsocketserver.cc:562): Socket::OPT_DSCP not supported.
'TestAppUwp.exe' (Win32): Loaded 'C:\Windows\System32\profext.dll'.
(p2ptransportchannel.cc:738): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:985): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port completed gathering candidates.
(turnport.cc:1040): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Received response with long lifetime: 10800 seconds.
(turnport.cc:1051): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled refresh in 3540000ms.
(basicportallocator.cc:1276): Net[Intel(R):192.168.0.194/32:Wifi:id=3]: Allocation Phase=Tcp
(port.cc:322): Port[9c540840::1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port created with network cost 10
(basicportallocator.cc:847): Adding allocated port for audio
(basicportallocator.cc:866): Port[9c540840:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Added port to allocator
(basicportallocator.cc:883): Port[9c540840:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Gathered candidate: Cand[:3019784464:1:tcp:1518280447:192.168.0.194:55190:local::0:otZH:wJm8xzkNmIQ6lbmLdkbsoSZv:3:10:0]
(basicportallocator.cc:911): Port[9c540840:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port ready.
(physicalsocketserver.cc:562): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:738): Port[9c540840:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:985): Port[9c540840:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port completed gathering candidates.
(turnport.cc:1334): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Received TURN allocate error response, id=766a68457473375865374e50, code=401, rtt=39
(turnport.cc:1282): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate request sent, id=6c2f6f5866432b6577706531
(turnport.cc:1288): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN allocate requested successfully, id=6c2f6f5866432b6577706531, code=0, rtt=38
(basicportallocator.cc:883): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Gathered candidate: Cand[:102829829:1:udp:25108223:119.81.194.156:39301:relay:49.207.133.244:38476:otZH:wJm8xzkNmIQ6lbmLdkbsoSZv:3:10:0]
(basicportallocator.cc:911): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port ready.
(physicalsocketserver.cc:562): Socket::OPT_DSCP not supported.
(p2ptransportchannel.cc:738): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: SetOption(5, 0) failed: 0
(basicportallocator.cc:985): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Port completed gathering candidates.
(basicportallocator.cc:1081): All candidates gathered for audio:1:0
(p2ptransportchannel.cc:793): P2PTransportChannel: audio, component 1 gathering complete
(turnport.cc:1040): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Received response with long lifetime: 10800 seconds.
(turnport.cc:1051): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled refresh in 3540000ms.
(port.cc:511): Received STUN ping id=3161727072446e6b3855714c from unknown address 192.168.0.171:45223
(port.cc:1104): Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|C--W|-|0|0|7962116751024340478|-]: Connection created
(p2ptransportchannel.cc:927): Adding connection from peer reflexive candidate: Cand[:3565642410:1:udp:1853824767:192.168.0.171:45223:prflx::0:f5DQ:GDpCnAy5oRlw+8hXDGJfQxQ/:3:10:0]
(port.cc:831): Port[9c3ce950:audio:1:0:local:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Sent STUN ping response, to=192.168.0.171:45223, id=3161727072446e6b3855714c
(p2ptransportchannel.cc:977): Not switching the selected connection on controlled side yet: Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|CR-W|-|1|0|7962116751024340478|-]
(p2ptransportchannel.cc:1831): Channel[audio|1|__]: Transport channel state changed from 0 to 2
(jseptransportcontroller.cc:1113): audio Transport 1 state changed. Check if state is complete.
(p2ptransportchannel.cc:1412): Channel[audio|1|R_]: Have a pingable connection for the first time; starting to ping.
(p2ptransportchannel.cc:2374): Selecting connection for triggered check: Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|CR-W|-|1|0|7962116751024340478|-]
(port.cc:1726): Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|CR-W|-|1|0|7962116751024340478|-]: Sent STUN ping, id=3334596a77576c7a376e6444, use_candidate=0, nomination=0
(port.cc:1672): Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|CR-I|-|1|0|7962116751024340478|-]: Received STUN ping response, id=3334596a77576c7a376e6444, code=0, rtt=4, pings_since_last_response=3334596a77576c7a376e6444
(p2ptransportchannel.cc:271): Switching selected connection due to: candidate pair state changed
(p2ptransportchannel.cc:1789): Channel[audio|1|R_]: New selected connection: Conn[9c3b4070:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->pZ6yP25S:1:1853824767:prflx:udp:192.168.0.171:45223|CRWS|S|1|0|7962116751024340478|4]
(dtlstransport.cc:791): DtlsTransport[audio|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 4
(opensslstreamadapter.cc:766): BeginSSL with peer.
(openssladapter.cc:819): SSL_connect:TLS client enter_early_data
(openssladapter.cc:819): SSL_connect:TLS client read_hello_verify_request
(openssladapter.cc:829): SSL_connect:error in TLS client read_hello_verify_request
(dtlstransport.cc:704): DtlsTransport[audio|1|__]: DtlsTransport: Started DTLS handshake
(srtptransport.cc:349): The params in SRTP transport are reset.
(openssladapter.cc:819): SSL_connect:TLS client read_server_hello
(openssladapter.cc:819): SSL_connect:TLS client read_server_certificate
(openssladapter.cc:819): SSL_connect:TLS client read_certificate_status
(openssladapter.cc:819): SSL_connect:TLS client verify_server_certificate
(opensslstreamadapter.cc:1087): Accepted peer certificate.
(openssladapter.cc:819): SSL_connect:TLS client read_server_key_exchange
(openssladapter.cc:819): SSL_connect:TLS client read_certificate_request
(openssladapter.cc:819): SSL_connect:TLS client read_server_hello_done
(openssladapter.cc:819): SSL_connect:TLS client send_client_certificate
(openssladapter.cc:819): SSL_connect:TLS client send_client_key_exchange
(openssladapter.cc:819): SSL_connect:TLS client send_client_certificate_verify
(openssladapter.cc:819): SSL_connect:TLS client send_client_finished
(openssladapter.cc:819): SSL_connect:TLS client finish_flight
(openssladapter.cc:819): SSL_connect:TLS client read_session_ticket
(openssladapter.cc:829): SSL_connect:error in TLS client read_session_ticket
(openssladapter.cc:819): SSL_connect:TLS client process_change_cipher_spec
(openssladapter.cc:819): SSL_connect:TLS client read_server_finished
(openssladapter.cc:819): SSL_connect:TLS client finish_client_handshake
(openssladapter.cc:819): SSL_connect:TLS client done
(dtlstransport.cc:636): DtlsTransport[audio|1|__]: DTLS handshake complete.
(jseptransportcontroller.cc:1050): Transport audio writability changed to 1.
(dtlssrtptransport.cc:218): Extracting keys from transport: audio
(srtptransport.cc:294): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1
(channel.cc:560): Channel writable (audio) for the first time
(channel.cc:560): Channel writable (video) for the first time
(impl_webrtc_videocapturer.cpp:734): CaptureDevice::StartCapture: returning
(messagequeue.cc:513): Message took 953ms to dispatch. Posted from: webrtc::VideoCapturerTrackSource::Initialize@../../pc/videocapturertracksource.cc:357
(messagequeue.cc:513): Message took 954ms to dispatch. Posted from: webrtc::PeerConnectionFactoryProxyWithInternal::CreateVideoSource@D:\azp2\1\s\external\webrtc-uwp-sdk\webrtc\xplatform\webrtc\api/peerconnectionfactoryproxy.h:59
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=up
(send_side_congestion_controller.cc:308): SignalNetworkState Up
(bitrate_prober.cc:110): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5)
(bitrate_prober.cc:110): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)
(send_side_congestion_controller.cc:555): Bitrate estimate state changed, BWE: 300000 bps.
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=up
(bitrate_allocator.cc:104): Current BWE 300000
(send_side_congestion_controller.cc:308): SignalNetworkState Up
(webrtcvoiceengine.cc:1471): Setting voice channel options: AudioOptions {}
(webrtcvoiceengine.cc:316): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:489): NetEq capacity is 50
(webrtcvoiceengine.cc:495): NetEq fast mode? 0
(webrtcvoiceengine.cc:513): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:523): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:533): Experimental ns is enabled? 0
(audio_processing_impl.cc:688): Highpass filter activated: 1
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
(webrtcvoiceengine.cc:1489): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:316): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, }
(webrtcvoiceengine.cc:489): NetEq capacity is 50
(webrtcvoiceengine.cc:495): NetEq fast mode? 0
(webrtcvoiceengine.cc:513): Delay agnostic aec is enabled? 0
(webrtcvoiceengine.cc:523): Extended filter aec is enabled? 0
(webrtcvoiceengine.cc:533): Experimental ns is enabled? 0
candidate: candidate:1218638837 1 udp 41885695 119.81.194.156 32105 typ relay raddr 49.207.133.244 rport 39285 generation 0 ufrag otZH network-id 3 network-cost 10 spdMid: audio spdMlineIndex: 0
(audio_processing_impl.cc:688): Highpass filter activated: 1
(fixed_gain_controller.cc:60): Gain to apply: 0 db.
(audio_processing_impl.cc:702): Gain Controller 2 activated: 0
(audio_processing_impl.cc:704): Pre-amplifier activated: 0
candidate: candidate:3019784464 1 tcp 1518280447 192.168.0.194 55190 typ host tcptype passive generation 0 ufrag otZH network-id 3 network-cost 10 spdMid: audio spdMlineIndex: 0
candidate: candidate:102829829 1 udp 25108223 119.81.194.156 39301 typ relay raddr 49.207.133.244 rport 38476 generation 0 ufrag otZH network-id 3 network-cost 10 spdMid: audio spdMlineIndex: 0
(impl_webrtc_audiodevicewasapi.cpp:1211): Using communications audio capture device: Microphone (Realtek(R) Audio)
(peerconnection.cc:5229): Changing to ICE connected state because all transports are writable.
(peerconnection.cc:3464): Changing IceConnectionState 0 => 2
(turnport.cc:1547): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=686253486a334f4e747a644a
(port.cc:1104): Conn[fff75a90:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->eJteEn0Z:1:2122260223:local:udp:192.168.0.171:45223|C--W|-|0|0|107839000890195455|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 2
(paced_sender.cc:110): PacedSender resumed.
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=674a33476552502b4934314a
(port.cc:1104): Conn[9c33be90:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->eJteEn0Z:1:2122260223:local:udp:192.168.0.171:45223|C--W|-|0|0|179897694439751167|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 3
(port.cc:1372): Conn[9c33be90:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->eJteEn0Z:1:2122260223:local:udp:192.168.0.171:45223|C--W|-|0|0|179897694439751167|-]: Connection pruned
(port.cc:1372): Conn[fff75a90:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->eJteEn0Z:1:2122260223:local:udp:192.168.0.171:45223|C--W|-|0|0|107839000890195455|-]: Connection pruned
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(turnport.cc:1547): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=5771675467664434796f6136
(port.cc:1104): Conn[9c3b5560:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|107839000017780223|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 4
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=6b474e4c59475442316b7258
(port.cc:1104): Conn[9c5ab010:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|179897693567335935|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 5
(port.cc:1104): Conn[9c5abcb0:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|7241540810645061118|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 6
(port.cc:1372): Conn[9c5abcb0:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|7241540810645061118|-]: Connection pruned
(port.cc:1372): Conn[9c5ab010:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|179897693567335935|-]: Connection pruned
(port.cc:1372): Conn[9c3b5560:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->oa9X2mJv:1:1686052607:stun:udp:49.207.133.244:37555|C--W|-|0|0|107839000017780223|-]: Connection pruned
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(turnport.cc:1547): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=73397045686d4a4b7a436a33
(port.cc:1104): Conn[9c57e200:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|107838996729446399|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 7
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=66784d787035626a4a525554
(port.cc:1104): Conn[9c57eea0:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|179897690279002110|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 8
(port.cc:1104): Conn[9c6db090:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|179897694439751166|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 9
(port.cc:1372): Conn[9c6db090:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|179897694439751166|-]: Connection pruned
(impl_webrtc_audiodevicewasapi.cpp:1239): Input audio device activated Microphone (Realtek(R) Audio)
(audio_device_buffer.cc:181): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:201): SetRecordingChannels(2)
(port.cc:1372): Conn[9c57eea0:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|179897690279002110|-]: Connection pruned
(port.cc:1372): Conn[9c57e200:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->tRWMjC9k:1:41885695:relay:udp:119.81.194.156:47618|C--W|-|0|0|107838996729446399|-]: Connection pruned
(turnport.cc:1547): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=317a735747525044454e795a
(port.cc:1104): Conn[9c6dbd30:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107838996695891454|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 10
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=71756a506f6c58682b655a76
(port.cc:1104): Conn[9c6f5a70:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107838996729446398|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 11
(port.cc:1104): Conn[9c6f4d70:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107839000890195454|-]: Connection created
(p2ptransportchannel.cc:1198): Channel[audio|1|RW]: Created connection with origin: 2, total: 12
(audio_device_buffer.cc:181): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:201): SetRecordingChannels(2)
(audio_device_buffer.cc:118): webrtc::AudioDeviceBuffer::StartRecording
(port.cc:1372): Conn[9c6f4d70:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:vy29WCeV:1:0:local:udp:192.168.0.194:59478->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107839000890195454|-]: Connection pruned
(port.cc:1372): Conn[9c6f5a70:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:TlHd4Otp:1:0:relay:udp:119.81.194.156:32105->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107838996729446398|-]: Connection pruned
(port.cc:1372): Conn[9c6dbd30:audio:Net[Intel(R):192.168.0.194/32:Wifi:id=3]:+jeTWT3/:1:0:relay:udp:119.81.194.156:39301->W53DneZ2:1:25108223:relay:udp:119.81.194.156:48765|C--W|-|0|0|107838996695891454|-]: Connection pruned
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(turnport.cc:1554): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=686253486a334f4e747a644a, code=0, rtt=75
(turnport.cc:1729): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 192.168.0.171:45223 succeeded
(turnport.cc:1741): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(turnport.cc:1554): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=5771675467664434796f6136, code=0, rtt=62
(turnport.cc:1729): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 49.207.133.244:37555 succeeded
(turnport.cc:1741): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(turnport.cc:1554): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=73397045686d4a4b7a436a33, code=0, rtt=51
(turnport.cc:1729): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 119.81.194.156:47618 succeeded
(turnport.cc:1741): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(channel.cc:779): Changing voice state, recv=1 send=1
(messagequeue.cc:513): Message took 240ms to dispatch. Posted from: cricket::BaseChannel::UpdateMediaSendRecvState@../../pc/channel.cc:764
(video_send_stream.cc:123): VideoSendStream::UpdateActiveSimulcastLayers
(video_send_stream_impl.cc:396): VideoSendStream::UpdateActiveSimulcastLayers
(video_stream_encoder.cc:1007): Video suspend state changed to: not suspended
(bitrate_allocator.cc:220): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps
(channel.cc:909): Changing video state, send=1
(audio_device_buffer.cc:239): Size of recording buffer: 960
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1461895327, sequence number: 16036, arrival time: 1603716919967
(decoder_database.cc:140): Initializing decoder with payload type '96'.
(agc_manager_direct.cc:382): [agc] Initial GetMicVolume()=0
(agc_manager_direct.cc:387): [agc] Initial volume too low, raising to 12
(channel.cc:1324): GetPlayoutTimestamp() failed to retrieve timestamp
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(turnport.cc:1554): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=317a735747525044454e795a, code=0, rtt=57
(turnport.cc:1729): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 119.81.194.156:48765 succeeded
(turnport.cc:1741): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(jseptransportcontroller.cc:288): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
(agc_manager_direct.cc:382): [agc] Initial GetMicVolume()=12
(webrtcvideoengine.cc:1000): SetVideoSend (ssrc= 939973652, options: VideoOptions {is_screencast : false, }, source = (source))
(video_stream_encoder.cc:487): ConfigureEncoder requested.
(rtp_sender_audio.cc:243): First audio RTP packet sent to pacer
(probe_controller.cc:195): Measured bitrate: 301000 Minimum to probe further: 1260000
(video_stream_encoder.cc:792): Video frame parameters changed: dimensions=640x480, texture=0.
(video_stream_encoder.cc:156): Set max framerate: 60
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=674a33476552502b4934314a
(quality_scaler.cc:72): Created CheckQpTask. Scheduling on queue...
(quality_scaler.cc:125): QP thresholds: low: 29, high: 95
(bitrate_allocator.cc:220): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=6b474e4c59475442316b7258
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=66784d787035626a4a525554
(rtp_sender_video.cc:486): Sent first RTP packet of the first video frame (pre-pacer)
(rtp_sender_video.cc:490): Sent last RTP packet of the first video frame (pre-pacer)
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=71756a506f6c58682b655a76
(probe_controller.cc:195): Measured bitrate: 303722 Minimum to probe further: 1260000
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920557, now_ms: 1603716920596)
(probe_controller.cc:195): Measured bitrate: 304722 Minimum to probe further: 1260000
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920561, now_ms: 1603716920598)
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 7856 bytes / 16 ms = 491 kb/s] [receive: 13456 bytes / 16 ms = 841 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 14168 bytes / 16 ms = 885.5 kb/s] [receive: 19768 bytes / 16 ms = 1235.5 kb/s]
(probe_controller.cc:195): Measured bitrate: 885500 Minimum to probe further: 1260000
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 20416 bytes / 65 ms = 314.092 kb/s] [receive: 20504 bytes / 66 ms = 310.667 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 21152 bytes / 67 ms = 315.702 kb/s] [receive: 23096 bytes / 67 ms = 344.716 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 22944 bytes / 67 ms = 342.448 kb/s] [receive: 24888 bytes / 67 ms = 371.463 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 0] [send: 25536 bytes / 68 ms = 375.529 kb/s] [receive: 26680 bytes / 67 ms = 398.209 kb/s]
(probe_controller.cc:195): Measured bitrate: 375529 Minimum to probe further: 1260000
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920656, now_ms: 1603716920669)
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920659, now_ms: 1603716920684)
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920661, now_ms: 1603716920700)
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920663, now_ms: 1603716920714)
(probe_controller.cc:195): Measured bitrate: 377005 Minimum to probe further: 1260000
(bitrate_prober.cc:129): Probe delay too high (next_ms:1603716920665, now_ms: 1603716920731)
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 1] [send: 21592 bytes / 80 ms = 269.9 kb/s] [receive: 19360 bytes / 79 ms = 245.063 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 1] [send: 22240 bytes / 81 ms = 274.568 kb/s] [receive: 21152 bytes / 80 ms = 264.4 kb/s]
(probe_controller.cc:195): Measured bitrate: 264400 Minimum to probe further: 1260000
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 1] [send: 24032 bytes / 96 ms = 250.333 kb/s] [receive: 21792 bytes / 95 ms = 229.389 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 1] [send: 24672 bytes / 97 ms = 254.351 kb/s] [receive: 23584 bytes / 95 ms = 248.253 kb/s]
(probe_bitrate_estimator.cc:149): Probing successful [cluster id: 1] [send: 26464 bytes / 97 ms = 272.825 kb/s] [receive: 25376 bytes / 96 ms = 264.333 kb/s]
(probe_controller.cc:195): Measured bitrate: 264333 Minimum to probe further: 1260000
The thread 0x70f8 has exited with code 0 (0x0).
(probe_controller.cc:195): Measured bitrate: 265333 Minimum to probe further: 1260000
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=674a33476552502b4934314a
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=6b474e4c59475442316b7258
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=66784d787035626a4a525554
(turnport.cc:1547): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN create permission request sent, id=71756a506f6c58682b655a76
(turnport.cc:1554): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=674a33476552502b4934314a, code=0, rtt=36
(turnport.cc:1729): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 192.168.0.171:45223 succeeded
(turnport.cc:1741): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(turnport.cc:1554): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=6b474e4c59475442316b7258, code=0, rtt=37
(turnport.cc:1729): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 49.207.133.244:37555 succeeded
(turnport.cc:1741): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(turnport.cc:1554): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=66784d787035626a4a525554, code=0, rtt=38
(turnport.cc:1729): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 119.81.194.156:47618 succeeded
(turnport.cc:1741): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(turnport.cc:1554): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN permission requested successfully, id=71756a506f6c58682b655a76, code=0, rtt=37
(probe_controller.cc:288): kWaitingForProbingResult: timeout
(turnport.cc:1729): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Create permission for 119.81.194.156:48765 succeeded
(turnport.cc:1741): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1462143277 in NTP clock: 3812685921942 estimated time in receiver clock: 1603716922584 converted to NTP clock: 3812685922584
(quality_scaler.cc:201): Checking average QP 29 (29).
(bitrate_allocator.cc:104): Current BWE 354273
The thread 0x7560 has exited with code 0 (0x0).
The thread 0x86e4 has exited with code 0 (0x0).
The thread 0xcc8 has exited with code 0 (0x0).
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\Windows.Globalization.dll'
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\Windows.Shell.ServiceHostBuilder.dll'
The thread 0xb140 has exited with code 0 (0x0).
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\msvcp110_win.dll'
'TestAppUwp.exe' (Win32): Unloaded 'C:\Windows\System32\Windows.System.Launcher.dll'
(remote_ntp_time_estimator.cc:74): RTP timestamp: 124105851 in NTP clock: 3812685927151 estimated time in receiver clock: 1603716927792 converted to NTP clock: 3812685927792
(quality_scaler.cc:201): Checking average QP 11 (11).
The thread 0xb484 has exited with code 0 (0x0).
(bitrate_allocator.cc:104): Current BWE 520348
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1462823497, sequence number: 16340, arrival time: 1603716930330
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c5abcb0 (11 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c33be90 (10 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c6db090 (9 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c5ab010 (8 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c57eea0 (7 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection fff75a90 (6 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c6f4d70 (5 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c3b5560 (4 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c57e200 (3 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c6f5a70 (2 remaining)
(port.cc:1773): Connection deleted with number of pings sent: 0
(p2ptransportchannel.cc:2237): Channel[audio|1|RW]: Removed connection 9c6dbd30 (1 remaining)
(quality_scaler.cc:201): Checking average QP 32 (32).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1463044177 in NTP clock: 3812685931952 estimated time in receiver clock: 1603716932593 converted to NTP clock: 3812685932593
(quality_scaler.cc:201): Checking average QP 30 (30).
(bitrate_allocator.cc:104): Current BWE 765314
(quality_scaler.cc:201): Checking average QP 30 (30).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 124584778 in NTP clock: 3812685937128 estimated time in receiver clock: 1603716937770 converted to NTP clock: 3812685937770
(quality_scaler.cc:201): Checking average QP 27 (27).
(audio_device_buffer.cc:414): [REC : 10045msec, 48kHz] callbacks: 1002, samples: 480960, rate: 47881, rate diff: 0%, level: 7587
(audio_device_buffer.cc:432): [PLAY: 10045msec, 48kHz] callbacks: 1004, samples: 481920, rate: 47976, rate diff: 0%, level: 27
(bitrate_allocator.cc:104): Current BWE 1122121
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1463733217, sequence number: 16641, arrival time: 1603716940336
(quality_scaler.cc:201): Checking average QP 8 (8).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1463949397 in NTP clock: 3812685942010 estimated time in receiver clock: 1603716942651 converted to NTP clock: 3812685942651
(bitrate_allocator.cc:104): Current BWE 1645319
(quality_scaler.cc:201): Checking average QP 5 (5).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 125062956 in NTP clock: 3812685947091 estimated time in receiver clock: 1603716947733 converted to NTP clock: 3812685947733
The thread 0x9b7c has exited with code 0 (0x0).
The thread 0x5c3c has exited with code 0 (0x0).
(audio_device_buffer.cc:414): [REC : 10052msec, 48kHz] callbacks: 1002, samples: 480960, rate: 47847, rate diff: 0%, level: 1865
(audio_device_buffer.cc:432): [PLAY: 10052msec, 48kHz] callbacks: 1003, samples: 481440, rate: 47895, rate diff: 0%, level: 27
The thread 0xa2c has exited with code 0 (0x0).
(bitrate_allocator.cc:104): Current BWE 2344670
The thread 0x6c6c has exited with code 0 (0x0).
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1464638437, sequence number: 16873, arrival time: 1603716950377
(quality_scaler.cc:201): Checking average QP 5 (5).
The thread 0x11bc has exited with code 0 (0x0).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1464850027 in NTP clock: 3812685952017 estimated time in receiver clock: 1603716952658 converted to NTP clock: 3812685952658
(quality_scaler.cc:201): Checking average QP 5 (5).
The thread 0xbde8 has exited with code 0 (0x0).
(bitrate_allocator.cc:104): Current BWE 2653397
(remote_ntp_time_estimator.cc:74): RTP timestamp: 125543299 in NTP clock: 3812685957098 estimated time in receiver clock: 1603716957740 converted to NTP clock: 3812685957740
(quality_scaler.cc:201): Checking average QP 5 (5).
(audio_device_buffer.cc:414): [REC : 10090msec, 48kHz] callbacks: 1005, samples: 482400, rate: 47810, rate diff: 0%, level: 1384
(audio_device_buffer.cc:432): [PLAY: 10090msec, 48kHz] callbacks: 1005, samples: 482400, rate: 47810, rate diff: 0%, level: 24
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1465543477, sequence number: 17090, arrival time: 1603716960405
(bitrate_allocator.cc:104): Current BWE 2718543
(quality_scaler.cc:201): Checking average QP 5 (5).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1465750657 in NTP clock: 3812685962024 estimated time in receiver clock: 1603716962665 converted to NTP clock: 3812685962665
(bitrate_allocator.cc:104): Current BWE 2792547
(quality_scaler.cc:201): Checking average QP 5 (5).
(remote_ntp_time_estimator.cc:74): RTP timestamp: 126023779 in NTP clock: 3812685967108 estimated time in receiver clock: 1603716967749 converted to NTP clock: 3812685967749
(audio_device_buffer.cc:414): [REC : 10007msec, 48kHz] callbacks: 1001, samples: 480480, rate: 48014, rate diff: 0%, level: 1490
(audio_device_buffer.cc:432): [PLAY: 10007msec, 48kHz] callbacks: 1001, samples: 480480, rate: 48014, rate diff: 0%, level: 30
(rtp_video_stream_receiver.cc:310): Packet received on SSRC: 2220358426 with payload type: 96, timestamp: 1466439607, sequence number: 17538, arrival time: 1603716970407
(quality_scaler.cc:201): Checking average QP 5 (5).
(bitrate_allocator.cc:104): Current BWE 3005052
(remote_ntp_time_estimator.cc:74): RTP timestamp: 1466633287 in NTP clock: 3812685971831 estimated time in receiver clock: 1603716972472 converted to NTP clock: 3812685972472
(quality_scaler.cc:201): Checking average QP 5 (5).
(bitrate_allocator.cc:104): Current BWE 3005052
(remote_ntp_time_estimator.cc:74): RTP timestamp: 126496799 in NTP clock: 3812685976963 estimated time in receiver clock: 1603716977604 converted to NTP clock: 3812685977604
(peerconnection.cc:5889): Usage signature is 3065
(openssladapter.cc:822): SSL3 alert read:warning:close notify
(opensslstreamadapter.cc:894): Cleanup
(openssladapter.cc:822): SSL3 alert write:warning:close notify
(dtlstransport.cc:657): DtlsTransport[audio|1|_W]: DTLS transport closed
(jseptransportcontroller.cc:1050): Transport audio writability changed to 0.
(peerconnection.cc:3464): Changing IceConnectionState 2 => 5
(channel.cc:573): Channel not writable (audio)
(channel.cc:573): Channel not writable (video)
(srtptransport.cc:349): The params in SRTP transport are reset.
(asyncudpsocket.cc:119): AsyncUDPSocket[192.168.0.194:59478] receive failed with error 10054
(channel.cc:779): Changing voice state, recv=1 send=1
(video_send_stream.cc:123): VideoSendStream::UpdateActiveSimulcastLayers
(video_send_stream_impl.cc:396): VideoSendStream::UpdateActiveSimulcastLayers
(channel.cc:909): Changing video state, send=1
(quality_scaler.cc:201): Checking average QP 5 (5).
(video_stream_encoder.cc:736): Number of frames: captured 1166, dropped (due to encoder blocked) 1, interval_ms 60000
(audio_device_buffer.cc:414): [REC : 10099msec, 48kHz] callbacks: 1002, samples: 480960, rate: 47625, rate diff: 1%, level: 1496
(audio_device_buffer.cc:432): [PLAY: 10099msec, 48kHz] callbacks: 1006, samples: 482880, rate: 47815, rate diff: 0%, level: 28
(webrtcvideoengine.cc:2052): VideoSendStream stats: 1603716979330, {input_fps: 20, encode_fps: 21, encode_ms: 12, encode_usage_perc: 30, target_bps: 1700000, media_bps: 1699712, suspended: false, bw_adapted: false} {ssrc: 939973652, width: 640, height: 480, key: 1, delta: 1174, total_bps: 1794568, retransmit_bps: 0, avg_delay_ms: 26, max_delay_ms: 86, cum_loss: 0, max_ext_seq: 10166, nack: 0, fir: 0, pli: 0}
(webrtcvideoengine.cc:2524): VideoReceiveStream stats: 1603716979331, {ssrc: 2220358426, total_bps: 0, width: 640, height: 480, key: 12, delta: 1148, network_fps: 0, decode_fps: 0, render_fps: 0, decode_ms: 2, max_decode_ms: 3, cur_delay_ms: 265, targ_delay_ms: 265, jb_delay_ms: 252, min_playout_delay_ms: 0, discarded: 0, sync_offset_ms: -92, cum_loss: 0, max_ext_seq: 18267, nack: 0, fir: 0, pli: 11}
(webrtcvideoengine.cc:1313): Call stats: 1603716979330, {send_bw_bps: 3005052, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
(peerconnection.cc:3512): Session: 8067918082610112391 Old state: kStable New state: kClosed
The thread 0x5ae4 has exited with code 0 (0x0).
The thread 0x58bc has exited with code 0 (0x0).
The thread 0xca84 has exited with code 0 (0x0).
(audio_device_buffer.cc:154): webrtc::AudioDeviceBuffer::StopRecording
(audio_device_buffer.cc:174): HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): 0
(audio_device_buffer.cc:177): total recording time: 59057
(webrtcvoiceengine.cc:1942): SetOutputVolume() to 0 for recv stream with ssrc 3953683229
(webrtcvideoengine.cc:1000): SetVideoSend (ssrc= 939973652, options: nullptr, source = nullptr)
(quality_scaler.cc:78): Stopping QP Check task.
(webrtcvideoengine.cc:1267): SetSink: ssrc:2220358426 nullptr
(channel.cc:540): Channel disabled
(video_send_stream.cc:150): VideoSendStream::Stop
(channel.cc:909): Changing video state, send=0
(video_send_stream_impl.cc:442): VideoSendStream::Stop
(rtp_rtcp_impl.cc:366): Failed to send RTCP BYE
(bitrate_allocator.cc:220): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps
(video_stream_encoder.cc:1007): Video suspend state changed to: suspended
(video_send_stream.cc:150): VideoSendStream::Stop
(video_send_stream_impl.cc:442): VideoSendStream::Stop
(video_send_stream_impl.cc:442): VideoSendStream::Stop
(video_send_stream_impl.cc:370): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [939973652], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {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}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: -1, red_payload_type: -1, red_rtx_payload_type: -1}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [], payload_type: -1}, c_name: uKW4QFepWmYz77ac}, rtcp: {video_report_interval_ms: 1000, audio_report_interval_ms: 5000}, pre_encode_callback: nullptr, post_encode_callback: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=up
(send_side_congestion_controller.cc:308): SignalNetworkState Up
The thread 0xac8c has exited with code 0 (0x0).
(send_statistics_proxy.cc:655): WebRTC.Video.InputWidthInPixels 640
WebRTC.Video.InputHeightInPixels 480
WebRTC.Video.InputFramesPerSecond periodic_samples:29, {min:20, avg:20, max:21}
WebRTC.Video.SentWidthInPixels 640
WebRTC.Video.SentHeightInPixels 480
WebRTC.Video.SentFramesPerSecond periodic_samples:29, {min:20, avg:20, max:21}
WebRTC.Video.SentToInputFpsRatioPercent 100
WebRTC.Video.EncodeTimeInMs 12
WebRTC.Video.KeyFramesSentInPermille 1
WebRTC.Video.QualityLimitedResolutionInPercent 0
WebRTC.Video.SentPacketsLostInPercent 0WebRTC.Video.NumberOfPauseEvents 0
WebRTC.Video.PausedTimeInPercent 0
WebRTC.Video.BitrateSentInBps periodic_samples:29, {min:194720, avg:1295672, max:1769016}
WebRTC.Video.MediaBitrateSentInBps periodic_samples:29, {min:189264, avg:1272096, max:1737624}
WebRTC.Video.PaddingBitrateSentInBps periodic_samples:29, {min:0, avg:0, max:0}
WebRTC.Video.RetransmittedBitrateSentInBps periodic_samples:29, {min:0, avg:0, max:0}
Frames encoded 1175
WebRTC.Video.DroppedFrames.Capturer 0
WebRTC.Video.DroppedFrames.EncoderQueue 1
WebRTC.Video.DroppedFrames.Encoder 0
WebRTC.Video.DroppedFrames.Ratelimiter 0
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=up
(send_side_congestion_controller.cc:308): SignalNetworkState Up
(video_receive_stream.cc:162): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, name: VP8, codec_params: {}}], rtp: {remote_ssrc: 2220358426, local_ssrc: 939973652, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: -1, red_type: -1, rtx_ssrc: 1637978611, rtx_payload_types: {-1 (pt) -> 96 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {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:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ANDROID_AV_STREAM_1, target_delay_ms: 0}
The thread 0x1190 has exited with code 0 (0x0).
The thread 0x3fcc has exited with code 0 (0x0).
(receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 60
Frames decoded 1160
WebRTC.Video.DroppedFrames.Receiver 1
WebRTC.Video.ReceivedPacketsLostInPercent 0
WebRTC.Video.AVSyncOffsetInMs 37
WebRTC.Video.RtpToNtpFreqOffsetInKhz periodic_samples:1, {min:0, avg:0, max:0}
WebRTC.Video.KeyFramesReceivedInPermille 10
WebRTC.Video.Decoded.Vp8.Qp 40
WebRTC.Video.DecodeTimeInMs 1
WebRTC.Video.JitterBufferDelayInMs 130
WebRTC.Video.TargetDelayInMs 142
WebRTC.Video.CurrentDelayInMs 142
WebRTC.Video.EndToEndDelayInMs 187
WebRTC.Video.EndToEndDelayMaxInMs 473
WebRTC.Video.InterframeDelayInMs 49
WebRTC.Video.InterframeDelayMaxInMs 199
WebRTC.Video.InterframeDelay95PercentileInMs 62
WebRTC.Video.ReceivedWidthInPixels 640
WebRTC.Video.ReceivedHeightInPixels 480
WebRTC.Video.EndToEndDelayInMs.S0 187
WebRTC.Video.EndToEndDelayMaxInMs.S0 473
WebRTC.Video.InterframeDelayInMs.S0 49
WebRTC.Video.InterframeDelayMaxInMs.S0 199
WebRTC.Video.InterframeDelay95PercentileInMs.S0 62
WebRTC.Video.ReceivedWidthInPixels.S0 640
WebRTC.Video.ReceivedHeightInPixels.S0 480
WebRTC.Video.MediaBitrateReceivedInKbps.S0 15
WebRTC.Video.Decoded.Vp8.Qp.S0 40
WebRTC.Video.MediaBitrateReceivedInKbps 283
(video_quality_observer.cc:109): WebRTC.Video.MeanTimeBetweenFreezesMs 57922
WebRTC.Video.TimeInHdPercentage 0
WebRTC.Video.TimeInBlockyVideoPercentage 0
WebRTC.Video.NumberResolutionDownswitchesPerMinute 0
(channel.cc:127): Destroyed channel: video
(channel.cc:540): Channel disabled
The thread 0x75a0 has exited with code 0 (0x0).
(audio_device_buffer.cc:140): webrtc::AudioDeviceBuffer::StopPlayout
(audio_device_buffer.cc:146): total playout time: 60796
(channel.cc:779): Changing voice state, recv=0 send=0
(webrtcvoiceengine.cc:1807): RemoveSendStream: 1392525940
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=up
(send_side_congestion_controller.cc:308): SignalNetworkState Up
(audio_send_stream.cc:158): ~AudioSendStream: 1392525940
(webrtcvoiceengine.cc:1880): RemoveRecvStream: 3953683229
(call.cc:1047): UpdateAggregateNetworkState: aggregate_state=down
(send_side_congestion_controller.cc:308): SignalNetworkState Down
(send_side_congestion_controller.cc:555): Bitrate estimate state changed, BWE: 0 bps.
(audio_receive_stream.cc:129): ~AudioReceiveStream: 3953683229
(channel.cc:127): Destroyed channel: audio
(messagequeue.cc:513): Message took 85ms to dispatch. Posted from: cricket::ChannelManager::DestroyVoiceChannel@../../pc/channelmanager.cc:203
(opensslstreamadapter.cc:894): Cleanup
(turnport.cc:1465): Port[9c3b2cc0:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN refresh request sent, id=716773555646356c54486832
(turnport.cc:1465): Port[9c53e790:audio:1:0:relay:Net[Intel(R):192.168.0.194/32:Wifi:id=3]]: TURN refresh request sent, id=457462772f32766132506747
The thread 0x63ec has exited with code 0 (0x0).
(call.cc:512): WebRTC.Call.EstimatedSendBitrateInKbps, periodic_samples:29, {min:303, avg:1833, max:3005}
(call.cc:520): WebRTC.Call.PacerBitrateInKbps, periodic_samples:29, {min:303, avg:1833, max:3005}
(call.cc:542): WebRTC.Call.VideoBitrateReceivedInBps, periodic_samples:28, {min:127936, avg:274824, max:876208}
(call.cc:550): WebRTC.Call.AudioBitrateReceivedInBps, periodic_samples:28, {min:34640, avg:35296, max:42960}
(call.cc:558): WebRTC.Call.RtcpBitrateReceivedInBps, periodic_samples:29, {min:10048, avg:12696, max:15264}
(call.cc:566): WebRTC.Call.BitrateReceivedInBps, periodic_samples:29, {min:176424, avg:344808, max:962232}
The thread 0x50e8 has exited with code 0 (0x0).
(paced_sender.cc:366): ProcessThreadAttached 0x0
(paced_sender.cc:366): ProcessThreadAttached 0x0
The thread 0x6b4c has exited with code 0 (0x0).
(send_delay_stats.cc:49): WebRTC.Video.SendDelayInMs, periodic_samples:28, {min:0, avg:1, max:1}
(rtc_event_log_impl.cc:198): Stopping WebRTC event log.
(rtc_event_log_impl.cc:215): WebRTC event log successfully stopped.
The thread 0x2c8c has exited with code 0 (0x0).
(peerconnection.cc:5889): Usage signature is 4089
(messagequeue.cc:513): Message took 212ms to dispatch. Posted from: webrtc::PeerConnectionProxyWithInternal::Close@D:\azp2\1\s\external\webrtc-uwp-sdk\webrtc\xplatform\webrtc\api/peerconnectionproxy.h:135
(peerconnection.cc:818): Session: 8067918082610112391 is destroyed.
(dtmfsender.cc:217): The Dtmf provider is deleted. Clear the sending queue.
Domonap
commented
3 years ago Hi I have exact same issue, no video on UWP from ios device, only Audio. On ios the video works fine from UWP. Ios to Ios and UWP to UWP everything works fine . Is there some fix to the issue? Thanks
Domonap
commented
3 years ago It did start working! By accident I turned my phone around to Landscape mode and it started to render the video on UWP. Maybe you guys have the same issue as I.
Hi,
First, a huge thanks for this library! It truly is a lifesaver.
On to the issue, I'm trying to get a video call work in UWP using MixedReality-WebRTC nuget v 1.0.3
Browser <-> UWP - all good Android Web (Chrome) <-> UWP - all good Android/iOS app <-> UWP - audio works fine, the Android/iOS video is not rendered in UWP
The 'OnTrack' method is called twice, once for audio and then for video but 'RemoteVideoFrameReady' is never called. So I guess the video track is received but something goes wrong in processing it.
iOS App SDPs
iOS app to UWP Offer
UWP to iOS app Answer
Android App SDPs
Android app to UWP Offer
UWP to Android app Answer
Android Web SDPs
Android web to UWP Offer
UWP to Android web Answer
I tried setting the PreferredVideoCodec to VP8, VP9 and H264 but it didn't work. Any help on this would be highly appreciated! Thanks much in advance.