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Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender?

From: Taylor Brandstetter via GitHub <sysbot+gh@w3.org>
Date: Thu, 12 Oct 2017 21:01:35 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-336265996-1507842079-sysbot+gh@w3.org>
This is a use case we've run into as well. Also, sometimes the application may have more knowledge about the network than the browser does, and has a better idea of what bandwidth it's capable of supporting.

We added a [native-only API](https://cs.chromium.org/chromium/src/third_party/webrtc/api/peerconnectioninterface.h?q=peerconnectioninterface&dr=CSs&l=786) for this kind of use case. I think the only question is whether this is in scope for WebRTC 1.0.

This also may be slightly related: https://github.com/w3c/ortc/issues/603

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Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1635#issuecomment-336265996 using your GitHub account
Received on Thursday, 12 October 2017 21:01:22 UTC

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