Limited bandwidtch for audio/video, only so many can join at once.

I have gone through your support guide on webrtc, and understand it is peer-to-peer.  We have 5 players, we can only get three to work at any given time. If one disconnects another person will immediately become viewable and so forth.  I as the gm (and a 300Mb fiber connection) can always see and hear people, but others rarely see and hear everyone else as one connections video/audio drops and another player will appear in their place.  Any ideas on what I can do? 

Trying to figure this out it has been an issue for weeks. 

Same overhere, when using Roll20 to share webcams i - as DM - hardly ever see all my players and get the feedback from the players they only see at most two other players.  Is webRTC using the same firebase shard for signalling as the campaign is on? I started using video and voice via discord and have the disctinct feeling that the vtt is less laggy.  Maybe the roll20 staff can explain something about webRTC and as a user i would gladly help relieve the peak loads and at the same time increase my own user experience.

I know webrtc is peer 2 peer not server based, so it does require everyone to broadcast to everyone, and receive individually from everyone. 

If only it was that simple. <a href="http://www.w3.org/TR/webrtc/" rel="nofollow">http://www.w3.org/TR/webrtc/</a> See the section: 10.5 Call Flow Browser to Browser It is a workflow diagram called Simple Call Flow ( so production code is way more complex) and there are quite some steps performed on the server. A key part of that is letting all clients know there is a new participant to connect to. And&nbsp; <a href="https://www.html5rocks.com/en/tutorials/webrtc/infrastructure/" rel="nofollow">https://www.html5rocks.com/en/tutorials/webrtc/infrastructure/</a> &nbsp;is also a good read.