[OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[Nick Khamis]

Hello Everyone,

After a week of tinkering with opensips rtpproxy functions, I have a
quite messy config file. Was wondering if anyone would
be kind enough to share or walkthrough a configuration that will get
two way audio working. Presently I have single
outgoing audio. Seems like I am not able to pick up the callee's RTP.

INFO:remove_session: RTP stats: 0 in from callee, 872 in from caller,
872 relayed, 0 dropped
INFO:remove_session: RTCP stats: 8 in from callee, 2 in from caller,
10 relayed, 0 dropped

Basic layout of the network

router 192.168.2.1
opensips 192.168.2.102 (bridged virutal box, ports forwarded)
asterisk 192.168.2.110 (bridged virutal box)
Polycom 192.168.2.11

[router]-----[opensips]-------[asterisk]--------[SIP Trunk]

Please bare with the virtual box setup, I am just trying to get all
the configs together before deploying onto the servers. I know i'm
really
close, and would love to be able to move on to the other parts (i.e.,
dialplan, routing etc...)

I pasted an ngrep trace at http://pastebin.com/A39vBG3t.

Thank you Kindly,

Nick.