That's an interesting idea. I wonder if the audio compression codec wouldn't obliterate it, though. Typically, they are supposed to be designed for human speech-level audio.

Regardless, there is nothing within the WebRTC spec for dealing with failed connections. I suppose you could close the WebRTCPeerConnection object, drop it on the floor for the GC to find, and make a new one. But the entire point of providing a list of STUN servers is that it's supposed to find the one that works for you.

I also just found out that, even though Tokbox (the 3rd party framework I'm using for WebRTC now that I've admitted I have more important things to do than screw around with this) is communicating a P2P video or audio connection, it cannot also let you negotiate a P2P data channel.