> but while unmuting happens as a result of an RTP packet arriving, muting only happens as a result of negotiation (setRemoteDescription's muteTracks list). I do not think that is true, see https://w3c.github.io/webrtc-pc/#mediastreamtrack-network-use And I am not happy with muted flipflopping for both network and api events. getSynchronizationSources() and getContributingSources() was built for audio-levels mainly (which is also the reason it does not make much sense for video -- i do not think any video rtp mixer uses csrcs...). If we want a timestamp for last-packet-received we should not use this API but expose something on the receiver. -- GitHub Notification of comment by fippo Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1983#issuecomment-420676417 using your GitHub accountReceived on Wednesday, 12 September 2018 14:47:48 UTC
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