| Oh, that's interesting. That's not as bad as I thought, although I am surprised that's the approach that was taken (unbundled). In that case, if you're relying on REMB from WebRTC with multiple PeerConnections, then, yes, each will be handled separately and you'll likely not end up with loss-base congestion control but will be using the latency-based congestion control WebRTC is going. Some potential problems I can think of with this approach: - It won't scale to a large number of streams. I'm not sure at what point it would cause problems, but I'm guessing it would work OK for... 20? 40? Most of the time, the extra ICE traffic and per-connection RTCP is probably small. A long time ago, there were issues with per-PeerConnection memory usage, but I think we (back when I worked on the WebRTC team) fixed them to a reasonable degree, so it should work for 20-40. You have to do a bunch of DTLS handshakes, but you could get around that by using SDES. You'll opening a lot of NAT bindings, potentially at the same time. Again, with a smaller number of streams, this might be fine. But at some point, you may cause issues with consumer-grade NATs. - The WebRTC receive-side congestion controllers (sending back REMB) will effectively be competing with one another. I'm not sure how well they work with a large number doing so (or even a small number.) I'd be interested to hear how many you can have running at the same time before you notice problems. - You can't easily prioritize one stream over the others if you want an "active speaker" view. I suppose you could something where you "steal" bits from one PeerConnection and give it to another, but that will probably contradict what you're trying to do with playing nicely with traffic shapers and AQM. I'd be very interested to hear your findings with how traffic shapers and AQM respond to non-bundled traffic vs bundled traffic. What characteristics do you seek work better or worse? Higher rates? Lower latencies? Lower jitter? Less loss? And how do you test (given that network behavior can vary so widely one network and the next)? If you'd like to talk more directly I'm "peter at signal.org". |
> It won't scale to a large number of streams.
Galene was designed for lectures and conferences, where a small number (1-5) of streams are sent to hundreds or thousands of receivers (and the budget is virtually nonexistent, because teaching and public research are eternally underfnded). It works beautifully for that particular application. It also happens to work well for medium-size meetings (25 senders, 50 receivers), which is a nice bonus, but not what the software was designed for.
> I'd be interested to hear how many you can have running at the same time before you notice problems.
We've been doing staff meetings with 25 senders and 50 receivers (50 people attending, half of which have their camera switched off). However, our main goal is supporting large lectures (2 flows distributed to hundreds of students, with students only switching their camera on in order to ask a question or show their cat): the commercial offerings work reasonably well for meetings, but are completely inadapted to lecturing.
> You can't easily prioritize one stream over the others if you want an "active speaker" view.
Oh, that. We simply forcibly switch the background streams to the lowest spatial layer. It works very well, but relies on the senders implementing either simulcast or SVC. Which is something we may safely assume (all desktop browsers implement at least simulcast, and lecturing is done with laptops).