- From: Harald Alvestrand via GitHub <sysbot+gh@w3.org>
- Date: Mon, 26 Feb 2024 15:06:44 +0000
- To: public-webrtc-logs@w3.org
So if the problem is that you have two laptops A and B in a room, and A plays out the signal at T+10 while B receives the signal at T+30, fixing the jitter buffer of both to min=max=20 ms will still give 20 ms difference between the signals. if you want to synchronize playout, it seems to me that you need the capture timestamp + synchronized clocks between the PCs + an agreement between the PCs on the offset between the capture timestamp and the playout time. I don't think just controlling the jitter buffer is going to give you that. -- GitHub Notification of comment by alvestrand Please view or discuss this issue at https://github.com/w3c/webrtc-extensions/issues/199#issuecomment-1964365100 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Monday, 26 February 2024 15:06:45 UTC