This article is incorrect about WebRTC. I don’t know about other protocols and what they offer. * Clock Recovery I have had no problems with measuring NTP drift. As the clocks change I would measure. * Common Clock for Audio and Video Sender Reports contain a mapping of Sequence Number to RTP Sequence Numbers. This is respected by every player I have used. My guess is author put their media in different MediaStreams.…
> I have had no problems with measuring NTP drift. Yeah, their claim is just weird. RTP does not impose an accuracy requirement on its timestamps (despite the name "NTP timestamp" in the Sender Reports, they are not actually expected to be synchronized with an NTP source), but I am skeptical such requirements would be met in practice if they did exist. The author only talks about video, but audio is a much bigger pro…
In practice this clock is generated via the PC clock so it isn't the same clock at all: https://chromium.googlesource.com/external/webrtc/+/lkgr/mod...
RTCP SRs are sent quite rarely (defaulting to 1s for video, 5s for audio) so quite poor for precise clock recovery required in professional applications.
Probably practical implementations just use buffer fullness to drive their resampler.