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Re: [webrtc-pc] Add adaptivePtime to RTCConfiguration (#2309)

From: Roman Shpount via GitHub <sysbot+gh@w3.org>
Date: Wed, 02 Oct 2019 18:05:31 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-537612423-1570039530-sysbot+gh@w3.org>
Typically the discussion about API would happen in w3c and network protocols would be discussed in IETF, but the RTCWEB group was recently closed so I am not sure what is the right place to talk about this. 

My main issue is that in case of adaptive audio frame, actual valid frame sizes are an implementation specific issue. What I would prefer is some sort of limitation of what is allowed (like anything that is dividable by 20 or 10 ms up to maxptime). Without some limitation on allowed audio frame values interop would be a nightmare due to very high number of use cases. This would effectively force me to disable adaptive frame size when using WebRTC with my media server, since I would not be able to effectively test it.

An additional API to set the valid audio frame size would be nice but might not be necessary. Some sort of API which informs about frame size used or when it is switched would definitely be useful.

GitHub Notification of comment by rshpount
Please view or discuss this issue at https://github.com/w3c/webrtc-pc/pull/2309#issuecomment-537612423 using your GitHub account
Received on Wednesday, 2 October 2019 18:05:32 UTC

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