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Re: [webrtc-stats] Audio/Video sync follow-up

From: ilnik via GitHub <sysbot+gh@w3.org>
Date: Wed, 12 Jul 2017 08:38:29 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-314695811-1499848707-sysbot+gh@w3.org>
@henbos that rtp header extension and timing frames are needed for debugging purposes. The idea is to have detailed history of delays for a certain frame for all components of webrtc. Theoretically, it could be done without the extension - send-side webrtc should report its half, receive-side should report it's side, then peers exchange information via out of band signaling. However, having the extension simplifies implementation and gives more flexibility (e.g. having delay of in-network rtp processors, high granularity reporting). The focus here is not on the E2E delay, but on the full details of all the delays.

Considering E2E delay - the requirement is to have RTT estimated via rtcp messages. I don't know all the details, but looks like it was done via RR rtcp messages. Also, if rtrr/dlrr is enabled it also works for sending-only clients. 

Also, we a currently investigating issue, where many calls have no estimations for E2E delay. What exactly went wrong there, we don't know yet.

GitHub Notification of comment by ilyanikolaevskiy
Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/222#issuecomment-314695811 using your GitHub account
Received on Wednesday, 12 July 2017 08:38:35 UTC

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