galene

ore than 1 video stream; for 1 video stream it probably works fine). Either way, it appears that each server->client video stream is independently running a loss-based congestion controller, all of which will be battling each other (like N TCP streams do). That can work, I guess, but it's better to run one congestion controller and then divide that bitrate among the various video streams, which is what I meant by "bitrate allocation". In other words, selecting video layers to send is exactly what I mean by bitrate allocation. Sorry for being unclear about that. The code you linked to is estimating the client->server bitrate for a given video stream. What I was looking for is code that will take a bitrate (from one congestion control mechanism, whatever it may be) and then divides that between the various video streams that flow server->client by selecting which layers to forward. I couldn't find that, and now I see why: because each video stream has its own congestion controller, and they apparently compete with each other where they are likely all loss-based in practice. Loss-based congestion control for video conferencing isn't as good as latency-based congestion control because it will cause more latency. Thanks for pointing out what I was missing. Now that I understand it better, I can see that it would work fine for 1 video stream (when there are only 2 clients in the call), but then likely falls back to loss-based congestion control for more than 2 clients, which will work, but not as well. If this is accurate, then I'd make a few suggestions for Galene: 1. Use one "maxBitrate" calculation per-"rtpConn" instead of per-"downTrack" and then divide that bitrate between those downTracks rather than doing N such independent calculations. This will avoid the problems from having congestion controllers competing with each other. 2. Feed the REMB value from the receiving client into the unified calculation. Then you'll get the benefit of latency-based congestion control (assuming the client is doing latency-based congestion control). 3. Switch the loss-based mechanism in the server to a latency-based system using something like transport-cc (which I think Pion supports).

Jitsi (Videobridge) [...] the SFU we use for Signal group calls is also open source Jitsi VideoBridge is a great piece of software, no doubt about it. I'm sure that Signal's SFU is competently done too.

nt back for all the video streams by WebRTC rather than one per stream (see, for example, here: https://source.chromium.org/chromium/chromium/src/+/main:thi...). Further, I find the spatial layer switching code to be strange. For examples, it doesn't go down a layer unless it's 150% over the estimated allowable bitrate, which gives a lot of opportunity for inducing latency. In short, I think Galene has a ways to go before it works as well as Jitsi (Videobridge), and thus Pyrite group calls are unlikely to work as well as Jitsi group calls (for 1:1 calls, I don't know; I didn't look into that). Oh, and just a reminder, the SFU we use for Signal group calls is also open source: https://github.com/signalapp/Signal-Calling-Service.