Is buffering possible in WebRTC?

Question

Is it possible to buffer the video/audio in WebRTC (of course, having then a delay on the other side) to improve the quality?

score 1 · Answer 1 · answered Oct 14 '15 at 21:18

1

WebRtc does buffering automatically when it is necessary. You don't have to think about it.

answered Oct 14 '15 at 21:18

Andrey Kon

I know that WebRTC does buffering, but how can you control it? Usually there's a trade-off between delay and quality. How can I control this? I suspect that a longer buffer will improve quality, but then a longer delay will be introduced. – Adrian Ber Oct 16 '15 at 08:27
WebRTC makes the best decision for you. – Andrey Kon Oct 16 '15 at 09:39
@AdrianBer how exactly would a longer buffer improve quality? – mido Oct 26 '15 at 07:49
Buffer will improve as any cache will. Of course a very long buffer will not improve things. – Adrian Ber Oct 27 '15 at 09:03
@AndreyKon, there is actually a way to do this. I'll post an answer on this. – Adrian Ber Oct 27 '15 at 09:07

score 1 · Accepted Answer · edited Oct 07 '21 at 07:24

1

There's no exact way to fully control this, but there are some settings in the Opus codec that can influence it, like, minptime, ptime, maxptime. Please check https://datatracker.ietf.org/doc/html/draft-spittka-payload-rtp-opus-03#page-12 for more info.

edited Oct 07 '21 at 07:24

Community

answered Oct 27 '15 at 09:09

Adrian Ber

2 Answers2