[Webkit-unassigned] [Bug 237443] New: Implement remote-inbound-rtp packetsLost

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=237443

            Bug ID: 237443
           Summary: Implement remote-inbound-rtp packetsLost
           Product: WebKit
           Version: Safari 15
          Hardware: All
                OS: All
            Status: NEW
          Severity: Normal
          Priority: P2
         Component: WebRTC
          Assignee: webkit-unassigned at lists.webkit.org
          Reporter: daginge at confrere.com
                CC: youennf at gmail.com

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.

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