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Media/WebRTC/WebRTC Debugging: Difference between revisions
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[[File:About webrtc Jitter and RTT.png|800px|border|Location in about:webrtc of jitter and RTT stats]] | [[File:About webrtc Jitter and RTT.png|800px|border|Location in about:webrtc of jitter and RTT stats]] | ||
In this image we can see that there are 0 milliseconds of jitter, and 32 milliseconds of round trip delay. This call should not be experiencing any noticeable delay. For all intents and purposes jitter and RTT are additive in nature. If there was 25ms of jitter reported and a RTT of 272ms, one could estimate the expected delay from transmission at the send side to play out on receive side to be <syntaxhighlight inline>25ms + (272ms / 2) = 161ms</syntaxhighlight> | In this image we can see that there are 0 milliseconds of jitter, and 32 milliseconds of round trip delay. This call should not be experiencing any noticeable delay. For all intents and purposes jitter and RTT are additive in nature. If there was 25ms of jitter reported and a RTT of 272ms, one could estimate the expected delay from transmission at the send side to play out on receive side to be <syntaxhighlight inline>25ms + (272ms / 2) = 161ms</syntaxhighlight> If the perceived delay is larger than the estimated delay that could indicate a problem within Firefox that requires debugging. Under these circumstances it would be helpful to grab a JSON copy of the current stats by pressing the "Copy Report" button, pasting those stats into your Bugzilla bug report. | ||
[[File:About webrtc copy report.png|800px|border|Location in about:webrtc of Copy Report button]] | |||
==Logging== | ==Logging== | ||