Hi Sammy,
Thank you for your quickest response, yeah it is the chapter 9 of the OpenSIPS
book & I'll study it but if I rely on Mediaproxy only, can that problem would
be resolved?
Regards,
Faisal Rehman
________________________________
From: SamyGo <[email protected]>
To: Faisal Rehman <[email protected]>
Cc: OpenSIPS users mailling list <[email protected]>
Sent: Wednesday, September 12, 2012 6:58 PM
Subject: Re: [OpenSIPS-Users] Audio Issue
Hi Faisal,
Do study Ch:9 of the opensips book. That will clear almost everything.
By media relaying I mean anything from rtpproxy or mediaproxy.
--
BR
Sammy
On Sep 12, 2012 6:51 PM, "Faisal Rehman" <[email protected]> wrote:
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
>>>
>>>
>>
>>
>
>
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