[SR-Users] RTPPROXY issue and sip to sip calling

Dear All I m trying to setup kamailio and asterisk in load balancing with a2billing . Currently all of my VMs, one Kamailio and two asterisks are on same subnet . I have started the RTPproxy like below ./rtpproxy -s udp:127.0.0.1:7722 -l X.X.X.153 -m 10000 -M 50000 -u root root -F -d INFO LOG_LOCAL0 My question is if all the VMs are on same subnet with same gateway what should be written in the private IP X.X.X.153/<private - ip> Secondly i m authenticating and registering the SIP on kamailio using the A2B DB . all the dialplan for a2b is being run on asterisk . Now if i want to call SIP peer to Peer like in case of followme case , How should i route the calls in Kamailio ? will it be using usr loc module? if so any help will be appreciated -- Regards Ahmed Rehan