[Karrot_Kream]

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?

[sumy23]

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).

[Karrot_Kream]

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.