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Re: [webrtc-pc] Prefer variable ptime (audio frame lengths) (#2300)

From: Roman Shpount via GitHub <sysbot+gh@w3.org>
Date: Fri, 13 Sep 2019 18:43:09 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-531348392-1568400187-sysbot+gh@w3.org>
Using variable frame duration in RTP packets will cause serious interop issues. The problem is jitter buffer implementations that assume that all frames are the same length and estimate packet time based on the RTP sequence number and ptime. If frame size changes, then such estimates will no longer work. Furthermore, RTP packet duration cannot be calculated before decoding (or without understanding packet format). There is a decent chance that this will even break NetEQ jitter buffer, since, I think it is based on number of packets queued not their duration.

Your best option is to renegotiate with the different ptime without codec changes or ICE restart.

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Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2300#issuecomment-531348392 using your GitHub account
Received on Friday, 13 September 2019 18:43:10 UTC

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