I am bit new to asterisk and trying to bypass the RTP traffic away from my asterisk server. Below is my current setup
VOIP Provider <—INTERNET—> MY ASTERISK SERVER <—INTERNET—>NAT ROUTER—>END USER
Currently in wireshark capture when VOIP Provider passes the SIP 183/200 they use different media servers around the world (which is different from providers SIP proxy address). I want to pass whatever media IP VOIP Provider using to END user so media directly flows between two endpoints.
Anyone can please guide me what to achieve this? It would be really helpful.