Bug 237443 - Implement remote-inbound-rtp packetsLost
Summary: Implement remote-inbound-rtp packetsLost
Status: RESOLVED FIXED
Alias: None
Product: WebKit
Classification: Unclassified
Component: WebRTC (show other bugs)
Version: Safari 15
Hardware: All All
: P2 Normal
Assignee: youenn fablet
URL:
Keywords: InRadar
Depends on:
Blocks:
 
Reported: 2022-03-03 12:30 PST by daginge
Modified: 2022-03-05 03:35 PST (History)
10 users (show)

See Also:


Attachments
Patch (6.68 KB, patch)
2022-03-04 01:33 PST, youenn fablet
no flags Details | Formatted Diff | Diff

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Description daginge 2022-03-03 12:30:21 PST
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.
Comment 1 youenn fablet 2022-03-04 01:33:19 PST
Created attachment 453822 [details]
Patch
Comment 2 daginge 2022-03-04 02:47:58 PST
Amazing turnaround time! Thank you, youenn!
Comment 3 EWS 2022-03-05 03:34:30 PST
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].
Comment 4 Radar WebKit Bug Importer 2022-03-05 03:35:17 PST
<rdar://problem/89854282>