This document pointing out the Direct RTP media or peer to peer communication of RTP.
I have managed to get Asterisk not to proxy media. I am running Freepbx 2.10.1.9 and Asterisk 1.8.12.0 on CentOS Linux 5.7 (Linux 2.6.18-274.3.1.el15.i686 - 32-bit) in Virtual machine. Directly connected to static IP.
It can be done under below conditions.
--> NAT should be disabled in the FreePBX ( sip.conf, Extension)
--> Network Devices and Phones should support for peer to peer communication (NO NAT)
--> In the extensions recording features should be turned off.
--> Should havedirect internet connection with static IP address
Setting changes in the SIP server, this is should be done via freepbx GUI
1) Application -> Extensions -> 'canreinvite=yes' and 'nat=no'
2) Settings -> Asterix SIP settings -> 'NAT=no' and 'IPconfiguratoin=static IP' and 'Reinvite Behavior=yes'
3) Add below entries to Other SIP Settings
--> 'directrtpsetup=yes' and
--> 'keepalive=yes'
4) Settings -> Advanced Settings -> "SIP canrenivite (directmedia)=yes" and "SIP nat=no"
5) Settings -> General Settings -> "Asterisk Dial command options:" should be empty
I have used tcpdump tool to monitor the communicatoin between server and SIP phones. Then I were albe to recognized the peer to peer communication.
Reference : http://www.dslreports.com/forum/r27852319-Can-I-get-Asterisk-to-not-proxy-media-
