Hi Alon,
I tried PhoneRTC 2.0, and was able start a audio call between two peers, using default Google stun server and turn server.
Actually I want to use own TURN server and Stun Server [ not Google's], and I given the TURN server detail in the session config and updated stun server detail also.
After this when we run the application, we are getting host ICE Candidates, but we are not getting any relay candidates in the ice candidate list[Which is need for our turn server].
Is there any way we can specify the ICE transport in phoneRTC, so we will able to get relay candidates.
One more question,
Is we can enable logging in libjingle_peerconnection_so.so
Thanks and Regards
Shyam