Bug 237443 – Implement remote-inbound-rtp packetsLost

• Status: RESOLVED
• Resolution: FIXED
• Component: WebRTC

[Dag-Inge Aas]

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.

[youenn fablet]

Created attachment 453822 [details]
Patch

[Dag-Inge Aas]

Amazing turnaround time! Thank you, youenn!

[EWS]

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].

[Radar WebKit Bug Importer]

<rdar://problem/89854282>