I'd like to make a SIP call into a conference for broadcasting to an RTMP destination. However the conference service prefers SIP over TLS to port 5061. (I can fallback to SIP over TCP, if necessary.)
I don't see any mention of the transport used by SIP in Web Call Server. Is it UDP only? If I can configure SIP over TCP or TLS, I need some guidance.
Thanks!
I don't see any mention of the transport used by SIP in Web Call Server. Is it UDP only? If I can configure SIP over TCP or TLS, I need some guidance.
Thanks!