Hi All,
I installed Kamailio server on my ubuntu machine. And installed rtpproxy
also in the same machine. I didm't make many changes to default
configuration files, kamctlrc and kamailio.cfg. Now I am able to
communicate between two SIP clients, if both sip clients are in the same
LAN. But I am not able to communicate between one sip client in one LAN and
the other SIP client in other LAN. Sip Call handling and ringing etc. is ok
but there is no voice for this case.
I don't have much experience in routing for NAT traversal. I tried to
experiment with rtpproxy_manage function in kamailio.cfg file. But I was
not successful. Please help me by providing any hints or pointers to
proceed further. I am using only Kamailio and Rtpproxy. No other software
like Asterisk or FreeSwitch.
*************
These are defines I used in kamailio.cfg file:
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ALIASDB
#!define WITH_USRLOCDB
#!define WITH_ANTIFLOOD
#!define WITH_NAT
#!define WITH_PRESENCE
listen ip address changed
*******************
Kamailio is compiled with following modules:
include_modules= db_mysql dialplan presence presence_xml
********
Regards,
Sateesh