Comment by derf_ | Hacker News Reader

Comment by derf_ | Hacker News Reader

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> 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 problem: if you do not send the right number of samples to the sound card at the right rate, you are going to get annoying clicks and pops, and "dropping or duplicating" audio frames will only cause more problems. You do not need to send media for a day to have these issues. Oscillators are bad enough they show up in minutes, and WebRTC obviously has strategies for dealing with them.

> If you want all your tracks to be synced you need to mark them as one MediaStream.

More specifically, the underlying mechanism of giving tracks that need to be synchronized the same CNAME has existed in RTP since the original RFC 1889 from the year 1996. The timestamp wrapping does require some care to get right, but is basically a non-issue.

That said, a lot of WebRTC applications will not ask for synchronization because it necessarily introduces latency, and for interactive use cases you are often better served by sacrificing exact sync for lower latency (as long as latency is low enough, sync is never going to get too bad, anyway). But that is very different from saying the standards do not support it if you want it.

The RFC does not mandate that the RTCP timestamp (which you need to handle wraparound if you join a stream halfway through) needs to be the same as the video/audio clock.

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.

> In practice this clock is generated via the PC clock so it isn't the same clock at all...

Yes, that is the point. WebRTC has to work with cheap / commodity hardware running a general purpose OS with clocks that are not synchronized. You get a bunch of clocks in different units running at different rates, and periodically are told the mappings between them. It is your job not to "run out of memory, or advance too quickly and run out of data to process," and indeed, WebRTC implementations have methods of solving those problems.

> Probably practical implementations just use buffer fullness to drive their resampler.

You are correct that in practice this gets driven by the jitter buffer in libwebrtc, but there is no resampler at all. Small changes in sampling rate are hard / computationally expensive to do well (as you certainly know). That is an implementation detail, though. You can use a different strategy if you have different requirements.

> ...precise clock recovery required in professional applications.

What are your actual precision requirements?

I also do not understand your concerns with wrap-around. Even SRs once every 5 seconds are more than enough to resolve ambiguity. You would have to go over half a day without seeing one before you could actually get confused, by which point you would have bigger issues.

Keep in mind that even at the start of the stream, the media timestamps are not absolute: each chooses an initial offset randomly.

Don't get myopic here - the timing constraints are critical for systems like DVB-T/C/S (and whatever the US equivalent is), not as much web. The things you're talking about might be dismissable when you're sending things from your blog app, but TS is primarily used in broadcasting.

The DVB machines I've worked with were very sensitive to any kind of jitter and clock skew.

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