[vr000m]

Webrtc protocol doesn’t dictate 1 or 2. Although browsers do implement some of their own assumptions for this. By default the client side buffer can be orders of 100s of milliseconds. this is as you pointed out tuned for real-time or live applications.

If you’re doing something like YouTube/Netflix and want to avoid going to a lower definition of the stream, that too can be tuned, albeit you’d want to use simulcast and implement your own player (to feed the video and audio frames for decoding at the pace you dictate).