[OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[qasimakhan at gmail.com]

My friend has this good walkthrough's for Opensips configuration and RTP.
and example is

http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html

You can also find other posts there. Just go through them and you will be
good to go.

PS: I also learned using rtpproxy using above mentioned page.

Regards,
Qasim

On Wed, Aug 29, 2012 at 7:58 AM, Nick Khamis <symack at gmail.com> wrote:

> 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.
>
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>
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