I have a confusion about asterisk’s “re-invite” concept . What I have come to know is that RTP traffic is redirected to the callee without passing through asterisk. This can be done by writing in the configuration files canreinvite=yes. I have also seen directrtpsetup=yes has a similar kind of result. Plz clarify the difference b/w the two.
I have experimented with both of the above settings. Both gave the same result. On capturing packets by a packet sniffer, I observed that when a call is established, some of RTP packets reach asterisk, then conversation goes on, bypassing the asterisk. What is the function of these initial RTP packets which reach asterisk?