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 <[email protected]> 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. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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