[OpenSIPS-Users] Audio Issue

SamyGo govoiper at gmail.com
Wed Sep 12 16:05:32 CEST 2012


Hi again,

In any case you've to intelligently detect NAT [ hint:
nat_uac_test()<http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id271494>]
Once NAT is detected apply rtpproxy or mediaproxy to traverse audio
successfully in between the two legs.

--
BR
Sammy


On Wed, Sep 12, 2012 at 7:00 PM, Faisal Rehman <faisal.rehman22 at yahoo.com>wrote:

> 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 <govoiper at gmail.com>
> *To:* Faisal Rehman <faisal.rehman22 at yahoo.com>
> *Cc:* OpenSIPS users mailling list <users at lists.opensips.org>
> *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" <faisal.rehman22 at yahoo.com>
> 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 <govoiper at gmail.com>
> *To:* Faisal Rehman <faisal.rehman22 at yahoo.com>
> *Cc:* OpenSIPS users mailling list <users at lists.opensips.org>
> *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" <faisal.rehman22 at yahoo.com>
> 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 <govoiper at gmail.com>
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; Faisal
> Rehman <faisal.rehman22 at yahoo.com>
> *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" <faisal.rehman22 at yahoo.com>
> 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
>
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