Closed Bug 970682 Opened 11 years ago Closed 11 years ago

Add loss, jitter, and RTT to about:webrtc

Categories

(Core :: WebRTC: Audio/Video, defect)

defect
Not set
normal

Tracking

()

RESOLVED FIXED
mozilla30

People

(Reporter: abr, Assigned: jib)

References

(Blocks 1 open bug)

Details

Attachments

(1 file)

No description provided.
Blocks: 964161
Once these are added to the stats API, we need to display them on about:webrtc.
Depends on: 970686
Loss and jitter are covered by Bug 970686, so this is the last patch (i.e. no need to leave this open)
Assignee: nobody → jib
Comment on attachment 8391431 [details] [diff] [review] Add RTT to about:webrtc Review of attachment 8391431 [details] [diff] [review]: ----------------------------------------------------------------- Looks pretty much good, with just a minor concern. ::: toolkit/content/aboutWebrtc.xhtml @@ +157,5 @@ > if (stat.bytesReceived !== undefined) { > statsString += " (" + round00(stat.bytesReceived/1024) + " Kb)"; > } > + statsString += " Lost: " + stat.packetsLost + " Jitter: " + stat.jitter; > + if (stat.mozRtt !== undefined) { Is this expected to be undefined in normal operation?
Attachment #8391431 - Flags: review?(docfaraday) → review+
(In reply to Byron Campen [:bwc] from comment #4) > > + if (stat.mozRtt !== undefined) { > > Is this expected to be undefined in normal operation? Yes, glad you asked. RTT is only included in the RTCP case on the outbound side.
Status: NEW → RESOLVED
Closed: 11 years ago
Resolution: --- → FIXED
Target Milestone: --- → mozilla30
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