Would be great to have remote-inbound-rtp packetsLost implemented to calculate send-side packet loss calculations for media tracks. Chrome now implements this (from https://webrtc.github.io/samples/src/content/peerconnection/constraints/) Report type=remote-inbound-rtp id RTCRemoteInboundRtpAudioStream_1121966634 time 1646339188209.999 ssrc: 1121966634 kind: audio transportId: RTCTransport_0_1 codecId: RTCCodec_0_Outbound_111 jitter: 0.00014583333333333335 packetsLost: 0 <-- here localId: RTCOutboundRTPAudioStream_1121966634 roundTripTime: 0.001 fractionLost: 0 totalRoundTripTime: 0.003 roundTripTimeMeasurements: 3 Would be a great way for us to design UI/handling around packet loss depending on this heuristic, as well as for our own analytical purposes and tagging calls good/bad. Since it's currently unsupported in Safari which amounts to about 60% of our traffic, we lose one important signal of quantitative call quality.
Created attachment 453822 [details] Patch
Amazing turnaround time! Thank you, youenn!
Committed r290865 (248096@main): <https://commits.webkit.org/248096@main> All reviewed patches have been landed. Closing bug and clearing flags on attachment 453822 [details].
<rdar://problem/89854282>