I would like to create an audio (mic) and video (camera) chat room with 12 people using webRTC. I understand signalling and the need for external services like ICE & STUN to help peers connect with each other.
But I don't want to use a full mesh architecture where everyone connects to everyone else because it is less efficient. I don't want to use expensive TURN relay services. I want the swarm to propagate the streams automatically so that if a direct connection isn't possible, the network routes the stream packets via peers automatically using encapsulation.
I don't want to use star architecture because I don't want a bottleneck peer.
I would like a peer to connect to maybe 2 or 3 other peers (max) and broadcast their media across the network without worrying about the relaying between peers. The routing would obviously need to be controlled by some service but I can't see that it's possible to do the stream encapsulation with RTCPeerConnection. Since WebRTC only allows for a RTCPeerConnection object (1 peer to 1 peer) and a peer would need to distinguish where the incoming stream is coming from and whether it needs to be relayed to another peer.
Is there a technology that extends WebRTC to allow for this more bandwidth efficient architecture?