3 ms·
To get it into the browser check out rtp-to-webrtc[0] Another big piece missing here is congestion control. It isn’t just about keeping bitrate low, but figuri
by Sean-Der 6y ago
To get it into the browser check out rtp-to-webrtc[0]
Another big piece missing here is congestion control. It isn’t just about keeping bitrate low, but figuring out what you can use. It is a really interesting topic to measure RTT/Loss to figure out what is available. You don’t get that in ffmpeg or GStreamer yet. The best intro to this is the BBR IETF doc IMO [1]
[0] https://github.com/pion/webrtc/tree/master/examples/rtp-to-webrtc https://github.com/pion/webrtc/tree/master/examples/rtp-to-w...
[1] https://tools.ietf.org/html/draft-cardwell-iccrg-bbr-congestion-control-00 https://tools.ietf.org/html/draft-cardwell-iccrg-bbr-congest...