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

From: henbos via GitHub <sysbot+gh@w3.org>
Date: Wed, 12 Jul 2017 08:24:07 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-314692310-1499847846-sysbot+gh@w3.org>
It seems that non-standards stats calculated end-to-end delay with estimates based on RTT, https://codereview.webrtc.org/2946413002/. It uses an RTP header extension to signal timestamps from all over the pipeline. Exposing all of these timestamps would seem more suitable for @alvestrand's WebRTC Frame Event Logging API than getStats since it's per-frame, but the fact that E2E is calculated based on RTT might mean that we can do something like we've previously discussed here? Is the RTP header extension for timing frames needed?

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Received on Wednesday, 12 July 2017 08:24:13 UTC

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