Hi friends,
I am just starting use asterisk for our VoIP server. It works fine in LAN. But
when it is deployed in public network(with a public IP), the SIP clients in
different NAT fails to communicate with each other. I have set 'icesupport' to
'yes' in sip.conf and set STURN and TURN server
Some of us do allow sip from the internet, but just like for smtp email
protections are in order.
I point my SRV records at dedicated sip proxies (I use kamailio) which
check the INVITEd sip uri the same way my MXs check the SMTP Evelope-To
addresses, and only allow INVITEs through to authorized
