WebRTC existing session

Question

WebRTC comes essentially in 3 flavors depending on the network situation:

• No NAT - session is peer to peer

• NAT - session uses STUN but is eventually peer to peer for the media

• NAT+Firewall: in which case session is not peer to peer and media goes through a TURN server.

Is there a way/tool to figure out for an existing live session if it is using a TURN server or if it's peer to peer?

I am using chrome. If a tool exists for other browsers I am happy to know about it as well.

Thx

Answer 1

this pull request for the webrtc samples makes the constraints/getStats sample show the ip address. Instead of the local and remote ip around here you want the candidateType property. If either of them is "relay" you are using a turn server. If both are "host" this is a direct connection without NAT. For all other cases NAT is used.

Note that this sample is currently a bit of a mess since the getStats API changed a bit.