Re:Video Live Streaming: Notes on RTMP, HLS, and WebRTC
by Karrot_Kream on Monday June 06, 2022 @03:06PM
(#31644490)
None of these problems are specific to WebRTC. You'll run into them in a WebRTC implementation, you'll run into them with QUIC, even with ffmpeg on the CLI you'll need to specify buffer sizes. As you mention these are both problems with livestreaming and the more you buffer, the less "live" your stream becomes. If you're interested in transmitting static videos, then why not go with HLS or even just making the static file available for direct download through HTTP instead of a live technology?
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Re:Video Live Streaming: Notes on RTMP, HLS, and WebRTC
by sumy23 on Monday June 06, 2022 @03:11PM
(#31644585)
The buffer sizes in ffmpeg are more about ensuring that the calculated bitrate is accurate iirc than ensuring smooth streaming (although you need your bitrate enforced to guarantee smooth streaming).
Re:Video Live Streaming: Notes on RTMP, HLS, and WebRTC
by Karrot_Kream on Monday June 06, 2022 @03:31PM
(#31644938)
IIRC (it's been a bit since I've configured this), you can specify both codec buffers and buffers for streaming to smooth out issues reading from the codec output. I could be wrong though.