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

[Neill Wilkinson]

Asterisk would need credentials configured in SIP.CONF for the user to do
this. Otherwise you would really need to be using a proper SBC or B2BUA with
Media conversion capability.


Neill...;o)

2009/1/23 Rawshan Iajdani <iajdani at provati.com>

>   Well that is what I am trying to do.. To originate the 2nd leg.. I need
> the username/password for authentication to the terminating server. Can I
> get that from OpenSer??? Because UA already logged into OpenSer..
>
>
>
>
>
> *From:* users-bounces at lists.kamailio.org [mailto:
> users-bounces at lists.kamailio.org] *On Behalf Of *Neill Wilkinson
> *Sent:* Friday, January 23, 2009 6:10 PM
> *To:* users at lists.kamailio.org
> *Subject:* Re: [Kamailio-Users] Openser-Asterisk Codec conversion..
>
>
>
> 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>
>
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