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Media/WebRTC/WebRTC Debugging: Difference between revisions
Adding more info on delay
(Added debugging delay) |
(Adding more info on delay) |
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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. | 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> | ||
==Logging== | ==Logging== | ||