- From: henbos via GitHub <sysbot+gh@w3.org>
- Date: Wed, 12 Sep 2018 15:26:15 +0000
- To: public-webrtc-logs@w3.org
The receiver already provides this information for audio RTP streams. This would allow you to check stream activity (inspect timestamps) or for mixer cases which SSRC or CSRC is of interest (why would this be audio-only? and even if it was then that argument is only against the CSRC case not SSRC). We already have an API that does exactly this, why tread audio and video different? Why add a new API that does the same thing that we already have when it is specific to receiving RTP packets? I don't think adding this to MediaStreamTrack is the right thing to do because this is specific to WebRTC and the referenced link says only to mute based on BYE or timeout; this does not work for inspecting the state of network, it only mutes when things are really bad and if you use mixing of SSRCs or mixing of CSRCs you do not get any onmute and you don't want to either. -- GitHub Notification of comment by henbos Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1983#issuecomment-420690658 using your GitHub account
Received on Wednesday, 12 September 2018 15:26:17 UTC