Media/WebRTC/WebRTC Debugging: Difference between revisions
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==Examining Call Performance Issues== | ==Examining Call Performance Issues== | ||
==Dumping Packet Headers== | ==Dumping Packet Headers== | ||
==Running WebRTC Tests== | |||
==Debugging Using 3rd Party Websites== | ==Debugging Using 3rd Party Websites== | ||
==Using RR And/Or Pernosco== | ==Using RR And/Or Pernosco== | ||
Revision as of 03:37, 10 August 2023
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
- While your call is still ongoing, open a tab and visit about:webrtc
- Click "Clear History" to clear the stats from other recent calls which are no longer ongoing.
- At the bottom of the page click 'Save Page', and save this file.
- 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.