What client-side tricks do you use to scale WebRTC applications to larger call sizes?
I've worked with teams building video calling applications that run in a browser for quite a while now. If I've learned anything, it's that there's always a new trick to learn or limitation to work around.
For example, in joining Daily, I learned that to balance CPU and network traffic, it's best to keep the overall download bitrate under 3Mbps (and upload bitrate under 1Mbps). This helps to: 1) keep the UDP network traffic within a range most routers can handle and 2) keep CPU decoding in a range most modern PCs can handle. A nice benefit of establishing this limit is that it helps scope the UI/UX design for your team.
What types of tricks have you learned or limitations have you hit in building and scaling WebRTC applications?