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Re: [webrtc-stats] jitterBufferDelay vs playoutDelay (#223)

From: Tilak via GitHub <sysbot+gh@w3.org>
Date: Tue, 03 Sep 2019 14:19:25 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-527479657-1567520364-sysbot+gh@w3.org>
I am a bit not clear about googJitterBufferMs. We have a freeviewpoint streaming webrtc framework, and I have tested it over experimental network, and found out that the link attributes (changing the RTT of the link) affects the googJitterBufferMs and the googTargetDelayMs to the RTT of the link. The minimum googJitterBufferMs observed in this case was 40msec and increases to 60~70msec, when the link RTT is changed to 40msec. 

The question is, is the jitterBuffer related to client Buffer (time when video frame enters the client buffer and leaves the client buffer) or the time when the video frame is sent from one peer and successfully decoded and played at the client?

Thanks,
Tilak 
[WebRTC Internals_new.pdf](https://github.com/w3c/webrtc-stats/files/3569887/WebRTC.Internals_new.pdf)
 

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Received on Tuesday, 3 September 2019 14:19:28 UTC

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