[Kamailio-Users] Openser-Asterisk Codec conversion..

Neill Wilkinson neill.wilkinson at btinternet.com
Fri Jan 23 13:09:35 CET 2009


Or Put another Way Asterisk acts in SIP terms as a Back2Back User Agent, to
terminate one side of the call let and originate a new call leg with a
different codec profile in the SIP/SDP. Asterisk then terminates the inbound
media, transcodes it an originates a new media stream on a completely
different call leg.

Neill....;o)

2009/1/23 Iñaki Baz Castillo <ibc at aliax.net>

> 2009/1/23 Rawshan Iajdani <iajdani at provati.com>:
> >
> > UA----->OpenSer(Outbound Proxy)---------Register Server
> > |
>  |
> >                       |
> >          Asterisk(codec converion)----------------------
> >
> > The UA will register to Register server through outbound proxy OpenSer.
> When
> > UA makes call it first comes to Openser, OpenSer should route the media
> to
> > Register server through Asterisk for codec conversion. OpenSer will not
> hold
> > any User account rather it will act as a proxy.
>
> Asterisk cannot receive *just* the media, it needs to receive the SIP
> signalling so then it can handle the media (and do the codec
> conversion).
>
> --
> Iñaki Baz Castillo
> <ibc at aliax.net>
>
> _______________________________________________
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20090123/78092599/attachment.htm>


More information about the sr-users mailing list