[OpenSIPS-Users] NAT - Unable to solve RTP Problem

[Jens Sauer]

Hi Lazlo and Flavio,

today i started from a fresh git install of Opensips 1.8 and use the rtpproxy delivered by opensips. ( http://opensips.org/pub/rtpproxy/) Last time i used the original from http://www.rtpproxy.org/ with the high load of CPU. Now it runs like it should be.

I´m happy to say - thanks! It works now.


Jens



________________________________
 Von: Laszlo <voipfreak.net at gmail.com>
An: Jens Sauer <sauer.jens at yahoo.de>; OpenSIPS users mailling list <users at lists.opensips.org> 
Gesendet: 22:41 Samstag, 22.Juni 2013
Betreff: Re: [OpenSIPS-Users] NAT - Unable to solve RTP Problem
 


Hi Jens,
Can you make a ngrep trace?
That way we can see more things about what is going on...
Do you see anything wrong in your syslog during the call?

-Laszlo

2013.06.22. 19:39, "Jens Sauer" <sauer.jens at yahoo.de> ezt írta:

Hello,
>
>
>since a month i try to solve my rtp problem. I´m quite new to OpenSips but i want it to be running on my Server.
>
>
>Both Phones (phonerLite) are connected and each one could call the other one, the only Problem is - no voice!
>Could someone please give me a hint, on using the 
right module (and parameter) to solve the problem. The Screenshot shows 
my topology. Both Clients use a public stun Server. I tried to use the 
rtpproxy but this piece of code use 99% of my CPU - there must be a 
different way.  
>
>
>
>Thanks for any help.
>
>
>Jens
>_______________________________________________
>Users mailing list
>Users at lists.opensips.org
>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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