Media/WebRTC/WebRTC Debugging

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Revision as of 06:57, 10 August 2023 by Ng (talk | contribs) (Adding profiling link)
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Reporting WebRTC Call Issues

The best way to report an issue is through Bugzilla using this link. Describing the issue you've run into, and include a URL along, with the details of the call setup.

For simple issues, the first place to look is to check the web developer console for error messages related to media format issues. If you see messages here related to WebRTC, getUserMedia, or getDisplayMedia, please add this information to your bug.

Share Your about:webrtc Contents

  1. While your call is still ongoing, open a tab and visit about:webrtc
  2. Click "Clear History" to clear the stats from other recent calls which are no longer ongoing.
  3. At the bottom of the page click 'Save Page', and save this file.
  4. Add this file as an attachment to your bug.

This file contains statistics about your call, the signalling that was used to setup your call, and information about the network transports.

Logging

Logging can be enabled through the "Enable WebRTC Log Preset" button at the bottom of [about:webrtc]. Alternatively one can set the following environment variable:

MOZ_LOG="jsep:5,sdp:5,signaling:5,mtransport:5,RTCRtpReceiver:5,RTCRtpSender:5,RTCDMTFSender:5,VideoFrameConverter:5,WebrtcTCPSocket:5,CamerasChild:5,CamerasParent:5,VideoEngine:5,ShmemPool:5,TabShare:5,MediaChild:5,MediaParent:5,MediaManager:5,MediaTrackGraph:5,cubeb:5,MediaStream:5,MediaStreamTrack:5,DriftCompensator:5,ForwardInputTrack:5,MediaRecorder:5,MediaEncoder:5,TrackEncoder:5,VP8TrackEncoder:5,Muxer:5,GetUserMedia:5,MediaPipeline:5,PeerConnectionImpl:5,WebAudioAPI:5,webrtc_trace:5,RTCRtpTransceiver:5,ForwardedInputTrack:5,HTMLMediaElement:5,HTMLMediaElementEvents:5"

Note that webrtc_trace will not be active until "Enable WebRTC Log Preset" is pressed.

Profiling

One can use the "WebRTC" preset on the [about:logging] page with the Firefox Performance Profiler.

Examining Call Performance Issues

Dumping Packet Headers

Running WebRTC Tests

Debugging Using 3rd Party Websites

Using RR And/Or Pernosco