On 11/09/2020 14:16, Sergio Garcia Murillo wrote: > On 11/09/2020 13:49, Lynne wrote: > > WebRTC uses rtp muxing to send all data in the same udp port, and lip > sync based on rtcp (also muxed on same udp port). It has been working > for years.
Its still a bit of a mess, and it took many years to get to this point. And yet its still having to be hacked around to support encryption in multi-person calls. RTP is ultimately a limiting factor here, it wasn't designed to be extensible, its just a simple header after all, with a complete pain of a timestamp because it wasn't built up for the internet (which hardly existed at the time) but for more local transmissions not based on UDP. And I haven't even said anything about SIP at all. Nothing prevents you from having a low, sub-frame latency Matroska stream, its just a lot nicer, well defined, future-proofed design rather than an amalgamation of multiple existing protocols not designed to work together (or arguably at all). _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
