Unless you're capping the video bitrate, the browser will try to use whatever the browser's default target is, for each connection. On Chrome that's 3mb/s, which is a lot of network bandwidth, and turns out to be a lot of cpu as well just shuffling those packets through the encoding->sending->bandwidth-estimation and receiving->decoding->rendering pipelines.
Capping the video bitrate is more complicated and confusing than it should be. It's better now that the browser implementations are all more or less closing in on "WebRTC 1.0" compliance. But you still need to reach into either the raw SDP you are exchanging during signaling, or the RTCPeerConnection objects, and set the encoding bitrate target.
The SaaS platforms that offer WebRTC APIs and infrastructure all do a lot of work under the covers to set bitrate caps, track constraints (resolution, for example), and other bits and pieces of WebRTC config that work well on a wide variety of networks, devices, and browsers.
There is a little more nuance then just paternalistic networks though. In same cases like NAT Mapping exhaustion you just can't give an individual user multiple long lived mappings. Address Dependendent filtering/mapping also makes sense in some cases. It makes P2P harder, but does give you the ability to provide your users more sessions at least!
https://medium.com/the-making-of-whereby/what-kind-of-turn-s...
Way more for customers that are mostly serving corporate users, of course (firewalls). And more for mobile-heavy user populations.
Actually, that's a good reminder that it would be nice to understand the mobile data networks breakdown in more detail. Most of the US mobile data networks require TURN, as far as I remember when I last looked at this. But I don't know if that's true everywhere in the world.
I wonder if we come back to this in 10 years what this number will be. Linux on the desktop and IPv6 is just around the corner...
If I were to guess, the problem GP is facing is bandwidth, a mesh network uses exponentially more bandwidth. For each user, the bandwidth is linear, N more people requires N more bandwidth. This is fine for downloads, but uploading N more can be much more challenging for certain networks.
And uPNP (Universal Plug and Play) sounds like its for device discovery in the same local network, so again, it doesn't sound related to webRTC, we can connect directly with each other on the same local network anyway.
UPNP has a number of functions, including forwarding of WAN packets to a specific LAN device.
Which, as it turns out, is a lot of users. I've seen estimates in the range of 10 to 20% of users. Which means, for a random selection of 7 users, you pretty much have a 50/50 chance of not being able to peer everyone using just STUN.