Re:Starlink review, four months in
by chromakode on Thursday July 22, 2021 @02:59PM
(#27922823)
WebRTC adjusts playback speed to accommodate variable network latency. Here's a paper which describes it a bit.
https://www.isca-speech.org/archive/PQS_2016/abstracts/19.ht...
When you lose a connection for a few seconds, maybe even 10 seconds, the video will pause, but when your connection reconnects, it continues where it paused. So you don't miss anything. I believe at some point when the speaker stops talking, like waiting for a response, it will jump cut their video to a more live feed again. So that you don't get too far behind.
I'm curious if anyone here knows about how this works and if it is common practice in live video chat?