Checking for WebRTC connectivity - reliable methods

Question

I have a live video chat application and I use a TURN server which supports STUN/TURN and both UPD/TCP transmission.

Sometimes users can be connected to the network which blocks that much ports and protocols that WebRTC connection just cannot happen (usually those are corporate networks). I would like to check if a WebRTC connection is possible before users try to connect to each other (actually, perform a technical check).

How can I do it? Ideas I have in my head:

1. Try to download a hosted chunk of data (audio file, for example) via WebRTC - is it possible and would this be enough to make sure both inbound and outbound connections are open?
2. Use a TURN server as a host to make a connection to and see if it fails (have no idea if I can do it or not)
3. Use Flash to try to download/upload a chunk of data over specific ports and protocols. May be even using Cirrus. However, I am not sure this test will be accurate from WebRTC prospective.
4. Any other ideas?

Additional requirement: the checking technique must support Chrome, Opera and Firefox. Preferably also IE/Safari via Temasys plugin.

Edition 1 - gathering ICE candidates is a good idea, however, it is not 100% reliable. Once I checked logs in my application and it actually gathered relay ICE candidates, but video/audio transmission failed. Tested on Apprtc as well and got same results.

Answer 1

the typical WebRTC approach to this is to create a peerconnection with STUN and TURN servers, call createOffer and setLocalDescription and watch the candidates gathered. See e.g. http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/

If you get srflx candidates, your stun server works (i.e. UDP is not blocked). More interesting is whether you get relay candidates. If you do, using TURN as a fallback will work. Quality might suffer if TURN/TCP is used. If you don't get relay candidates... calls are very unlikely to work.