1339270 - Upstream webrtc code doesn't add padding packets to retransmission history (Issue 7143)

Status: RESOLVED FIXED

(Core :: WebRTC: Networking, defect, P1)

Description

[Randell Jesup [:jesup] (needinfo me)]

Upstream code in webrtc.org branch 49 and 57 (and 43) don't add padding packets (RTPSender::SendPadData()) to the rtp packet history.  If one of the padding packets is lost, it can't be retransmitted, and the other side's jitter buffer will still until the next keyframe, or until it times out waiting for a retransmit and requests a keyframe.  This causes freezes and lower video quality.

Webrtc.org issue 7143: https://bugs.chromium.org/p/webrtc/issues/detail?id=7143

Comment 1

[Randell Jesup [:jesup] (needinfo me)]

Attachment: Add rtp 'padding' packets into rtp history for handling NACKs

this also causes the simulcast tests to be flaky under rr, for example

Comment 2

[Pulsebot]

Pushed by [e-mail]:
https://hg.mozilla.org/integration/mozilla-inbound/rev/b6bdf4cf8348
Add rtp 'padding' packets into rtp history for handling NACKs r=ng

Comment 3

[Carsten Book [:Tomcat]]

https://hg.mozilla.org/mozilla-central/rev/b6bdf4cf8348