[asterisk-dev] RTP- INVITE

0617260639 at alu.uma.es 0617260639 at alu.uma.es
Tue May 13 05:02:45 CDT 2008


Im making calls between 2 nokia N80 in the same subnet with Asterisk. I
want direct RTP traffic between the terminals but I've noticed that before
that happens there is a short RTP flow between the terminals and Asterisk.
Why is that?
I've also notices that there are several SIP INVITE messages during the
RTP flow between terminal. Why does it send more than the initial INVITE?




More information about the asterisk-dev mailing list