Hi Sammy,
Yeah I was sure that this problem can be resolved via NAT traversal or
something but I've not used it before so got to study it a bit first. What
about the media relaying thing, you are talking about Media Proxy aren't you?
Regards,
Faisal Rehman
________________________________
From: SamyGo <[email protected]>
To: Faisal Rehman <[email protected]>
Cc: OpenSIPS users mailling list <[email protected]>
Sent: Tuesday, September 11, 2012 11:01 PM
Subject: Re: [OpenSIPS-Users] Audio Issue
Hi,
Use NAT handling for clients behind NAT and may use media relaying for those
clients.
Clients will pubic IP may send/received RTP directly or go through with the
media relay tool if the other end needs NAT handling.
Thanks
Sammy
On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[email protected]> wrote:
Hi Sammy,
>
>
>I have neither changed the configuration file for any media-relay nor engaged
>any media proxy yet, I mean to say that configuration is totally fresh with no
>changes except configurations with database. The network is the simplest one
>as my server is in UK with a public IP running OpenSIPS on it & I just want to
>make pc to pc calls through it. There is no issue in call connectivity but
>without RTP & also running the command for RTP as tethereal -i any -R rtpevent
>does not show any thing or any codec flow. Yeah you are correct about the fact
>that changing networks can not send the RTP directly to the subscribers so is
>there any possible available solution for it?
>
>
>
>Warmest Regards,
>
>
>Faisal Rehman
>
>
>________________________________
> From: SamyGo <[email protected]>
>To: OpenSIPS users mailling list <[email protected]>; Faisal Rehman
><[email protected]>
>Sent: Tuesday, September 11, 2012 9:40 PM
>Subject: Re: [OpenSIPS-Users] Audio Issue
>
>
>Hi Faisal,
>What are your opensips config related to any media-relay ? Have you engaged
>any mediaproxy in your dialplan ?
>What is your network topology.?
>I can only imagine media between two endpoints getting connected directly when
>on same network. But when you change network the two endpoints cant possibly
>send RTPs directly to each other.
>Regards,
>Sammy
>On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[email protected]> wrote:
>
>Hello Everyone!
>>
>>
>>I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine.
>>In the initial phase I am just testing PC to PC calls but facing a little
>>issue with audio that is fine if we test on the same network but disappears
>>as soon we change the network, inspite of disabling firewall on both local PC
>>& server the issue still persists. So may I get any ideas what could be the
>>possible reasons for this?
>>
>>
>>
>>
>>
>>Regards,
>>
>>
>>Faisal Rehman
>>_______________________________________________
>>Users mailing list
>>[email protected]
>>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
