Hi Everyone, I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The Flow is:
PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISK
Regular call (no srtp) works fine. However when I setup SRTP the asterisk replies with 488 Not Acceptable Here. I followed the Secure Calling Tutorial, but nothing seems to solve the issue.

Below you can see the endpoint configuration + debug output. Any help would be appreciated.

[acme]
type=endpoint transport=transport-udp context=app-router disallow=all allow=alaw allow=ulaw aors