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[webrtc-stats] Define reception time for jitterBufferDelay stat (#549)

From: Sergio Garcia Murillo via GitHub <sysbot+gh@w3.org>
Date: Fri, 07 Feb 2020 09:21:42 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issues.opened-561515576-1581067301-sysbot+gh@w3.org>
murillo128 has just created a new issue for https://github.com/w3c/webrtc-stats:

== Define reception time for jitterBufferDelay stat ==
Currently the `jitterBufferDelay` stat is defined as:
It is the sum of the time, in seconds, each audio sample or video frame takes from the time 
it is received and to the time it exits the jitter buffer.
When tried to land a patch for libwebrtc to correct the current implementation (https://webrtc-review.googlesource.com/c/src/+/168042), it has been risen the concern that it is not clear what the reception time really mean:

1. The reception time of the RTP packet as close to the network layer as possible
2. The reception time of the RTP packet by the jitter buffer code
3. The time when the audio samples have been already decoded and ready to be passed to the jitter buffer internal queues

In case of (2) or (3), we would need a new stat defined for (1)  (playoutDelay?)

Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/549 using your GitHub account
Received on Friday, 7 February 2020 09:21:44 UTC

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