[OpenSIPS-Devel] Sip stereo B2BUA conference server

[Dave Singer]

MatzeMuc86,

Opensips handles just the SIP signaling which contains the information about
where the RTP should connect to. The modules media_proxy and nat_helper can
be used to communicate to the external applications media_proxy and
rtp_proxy respectively that setup proxying of the RTP and return to the
opensips module the connection setup it has prepared for the RTP. The module
then alters the SDP appropriately and the opensips script continues on
deciding where to send and sending the SIP message to the next server.
I believe rtp_proxy can be setup to stream audio from a file and it might be
a starting point for you to mix the stereo audio.
However I'm not sure if opensips can be the endpoint of a call (SIP) without
writing or extending a module for that purpose. The B2B module is something
you might look at to see how much tweaking it would take to make it do what
you want.

Dave

On Fri, Feb 25, 2011 at 1:15 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:

> Hi,
>
> as I already invested a lot of time, I try to be sure to check out all
> possibilities. I already know about PJSIP but that does not mean that
> OpenSIPS could not be the right project for me - I wanted to be sure.
> Anyway: Thanks A LOT for your very nice support and searching time!!!
>
> -----Ursprüngliche Nachricht-----
> Von: devel-bounces at lists.opensips.org
> [mailto:devel-bounces at lists.opensips.org] Im Auftrag von Saúl Ibarra
> Corretgé
> Gesendet: Freitag, 25. Februar 2011 10:08
> An: OpenSIPS devel mailling list
> Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
>
> Hi,
>
> On 02/25/2011 10:02 AM, MatzeMuc86 wrote:
> > Hello Adrian,
> >
> > I know that I need some RTP part which receives, mixes and sends the
> > media stream. I thought I can do all these things with OpenSIPS? I saw
> > that SDP is implemented, but, of course, this is transported by SIP.
> > Maybe I am wrong about RTP and OpenSIPS - sorry.
> >
>
> SDP is just signaling, OpenSIPS doesn't deal with RTP at all.
>
> > Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. -
> > but no stereo. TO implement this, the project is that big that - after
> > talking with the freeswitch developers - this seems to be a very big
> > project. As it is only a Bachelor Thesis I thought about finding an
> > easier way to implement my idea.
> >
>
> IIRC, PJSIP does have stereo support to some degree. A quick search
> returned
> this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
> _______________________________________________
> Devel mailing list
> Devel at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>
>
> _______________________________________________
> Devel mailing list
> Devel at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/devel/attachments/20110225/85896f88/attachment.htm>