Signal-Calling-Service VS Jitsi Video Bridge

I am npt sure if this is related but Signal built Signal Calling Service and according to them it worked great.

I was curious and looked through the code of Galene briefly and found the following, which may answer your question. For context, I am familiar with the Jitsi code and have written my own calling server (and written about it: https://signal.org/blog/how-to-build-encrypted-group-calls/).

Galene appears to be less mature than Jitsi. For example, it uses REMB feedback messages from the client to calculate allowable bitrates rather than calculating the bitrates itself (as Jitsi and Signal's SFU do). Worse, it appears that what it does with that information is erroneous. I could be wrong, but it looks like the bitrate allocation code (see https://github.com/jech/galene/blob/e8fbfcb9ba532f733405b1c5...) only allocates the bitrate for one of the video streams, not all of them. Perhaps the author did not realize that there is one REMB sent 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.

And yeah, it uses Signal-Calling-Service written on Rust.

That's a well written article covering the basics of simulcast.

If you're interested in seeing an implementation of an SFU doing simulcast forwarding written in Rust, we (at Signal) recently open sourced our SFU:

https://github.com/signalapp/Signal-Calling-Service/blob/mai...

I've been looking at open source video conferencing software options, specifically Jitsi. When reading their deployment docs the phrase "real time" comes up occasionally, for example:

It stops being peer-to-peer as soon as there are more than two participants (i.e more than one destination for the stream). Jitsi uses their Videobridge https://jitsi.org/jitsi-videobridge/ as a central server to do the multiplexing.