[Kamailio-Users] Kamailio Newb questions
carl Lougher
c_lougher at yahoo.co.uk
Thu Feb 26 23:07:24 CET 2009
So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy?
What are you using rtpproxy for that is different than stun?
--- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
> From: Daniel-Constantin Mierla <miconda at gmail.com>
> Subject: Re: [Kamailio-Users] Kamailio Newb questions
> To: c_lougher at yahoo.co.uk
> Cc: users at lists.kamailio.org
> Date: Thursday, 26 February, 2009, 12:27 PM
> On 02/26/2009 01:19 PM, carl Lougher wrote:
> > Thanks for that. So does it mean by using rtpproxy you
> will therefore carry all the rtp streams through that server
> yes, that is the role of RTPProxy - to proxy the RTP
> streams, therefore those go via the server.
>
> If you want end-to-end RTP stream, then look at STUN, if
> the phones are not behind symmetric nat, it can help.
>
> > or can it be redirected to the sip provider from the
> endpoint?
> >
> > Also how do you put the kamailio server in the
> equation? Do you set it up as an external proxy for the
> clients or do you register the clients to it then just use
> asterisk for the media/vmail etc?
> >
> I do everything in kamailio but the media services which i
> do with asterisk (vmail, ivr, ...) - authentication,
> registration, call routing is done in kamailio.
>
> Cheers,
> Daniel
>
> >
> > --- On Thu, 26/2/09, Daniel-Constantin Mierla
> <miconda at gmail.com> wrote:
> >
> >
> >> From: Daniel-Constantin Mierla
> <miconda at gmail.com>
> >> Subject: Re: [Kamailio-Users] Kamailio Newb
> questions
> >> To: c_lougher at yahoo.co.uk
> >> Cc: users at lists.kamailio.org
> >> Date: Thursday, 26 February, 2009, 9:16 AM
> >> Hello,
> >>
> >> On 02/26/2009 12:59 AM, carl Lougher wrote:
> >>
> >>> Howdy,
> >>> I'm trying to remove the media/rtp streams
> from an
> >>>
> >> asterisk server for natted users so would like to
> know if
> >> this is possible with kamailio.
> >>
> >>>
> >> yes it is possible. nathelper+rtpproxy is the
> option I use
> >> and prefer because of flexibility and
> performances. You can see an
> >> example at:
> >>
> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
> >>
> >>
> >>
> >>> Qu's:
> >>> What is the best option?
> >>> rtpproxy/mediaproxy?
> >>> nathelper?
> >>>
> >>> If i use kamailio to achieve this does it mean
> that i
> >>>
> >> still have to carry the rtp streams through the
> kamailio
> >> server instead?
> >>
> >>>
> >> through the rtpproxy server, which can be located
> on same
> >> or different machine than kamailio.
> >>
> >>
> >>> Also will i need to change the logon info for
> the
> >>>
> >> clients so they now logon to kamailio then i just
> point
> >> registrar to asterisk?
> >>
> >>> Can i use kamailio for sip trunks to asterisk
> and
> >>>
> >> carry rtp and natted clients media streams rather
> than
> >> register to asterisk?
> >>
> >>>
> >> Yes, you can register to kamailio, see registrar
> and usrloc
> >> modules.
> >>
> >> Cheers,
> >> Daniel
> >>
> >>
> >>> Many thanks,
> >>> Taff..
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> _______________________________________________
> >>> Kamailio (OpenSER) - Users mailing list
> >>> Users at lists.kamailio.org
> >>>
> >>>
> >>
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> >>
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
> >>
> >>>
> >> -- Daniel-Constantin Mierla
> >> http://www.asipto.com
> >>
> >
> >
> >
> > _______________________________________________
> > Kamailio (OpenSER) - Users mailing list
> > Users at lists.kamailio.org
> >
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> >
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
> >
> >
>
> -- Daniel-Constantin Mierla
> http://www.asipto.com
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