Hi All,
I am currently working with following setup.
Scenario 1: TURN server not forced
SipML5 on FireFox -> Asterisk WebRTC gateway
Audio call happens successfully.

Scenario 2: TURN server forced with iceTransportPolicy: "relay"
SipML5 on FireFox -> coTurn Turn Server -> Asterisk WebRTC gateway
Call fails.

Only change here is TURN server is forced. I checked the Wireshark packet capture in both scenarios. Attached the snapshots.
Asterisk server is 172.16.4.210
TURN is 172.16.4.205
FireFox 172.16.1.26
In second scenario, FireFox keep sending "Server Hello" repeatedly. I am not able to figure out the reason. How to proceed further is figuring out the issue? Any pointers will be helpful.
Same setup works fine if Chrome is used instead of Firefox.
Thanks in advance.
Thanks,
Sachin