ICE connectivity in a WebRTC call

Question

In a Webrtc call, I am using sip signalling and sdp for media parameter negotiation.

Before call start, I do a stun-bind transaction and get reflexive candidates. I have put those reflexive candidates in sdp in addition to base and host candidates.

As soon as we get 200 OK for Invite, we need to start media. For media start, I need to know which candidate pair I need to use.

I hope to determine which candidate pair I need to use, we need to do connectivity check. I am not sure how to do connectivity check (like which message to send.. etc).

Can somebody help me in this to understand.

Also is there an open source (c, linux based), that gives ice/stun/turn support.

Answer 1

This information is given on RFC 5245. You need to read this RFC for implementing ICE. For your query about doing ICE connectivity check, read this section of the RFC.

Also is there an open source (c, linux based), that gives ice/stun/turn support.

Search google for this and you will get your answer.