- From: Ruslan Burakov via GitHub <sysbot+gh@w3.org>
- Date: Fri, 01 Mar 2019 11:15:04 +0000
- To: public-webrtc-logs@w3.org
As for the use case I can share how we want to use it. In the real time communication if you have more than two participants then quite often some of the participants doesn't speak. During this time we want to increase latency on MediaStreamTracks sourced by an RTCPeerConnection. Increased delay means, depending on your implementation of course, that you buffer can accumulate more packets, and you have more media to play in case of problems which results in much better media quality. Increased delay is undesirable during conversation but that is why we want to increase only for participants who don't speak, for example muted their microphone. -- GitHub Notification of comment by kuddai Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2109#issuecomment-468631961 using your GitHub account
Received on Friday, 1 March 2019 11:15:05 UTC