[OpenSIPS-Users] SIP Trunking

[Matthew S. Crocker]

I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket).  Will it perform the functions to proxy the SIP & RTP streams (via mediaproxy) between my end users and my internal gateway?

At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc.

-Matt

----- "Alex Balashov" <abalashov at evaristesys.com> wrote:

> From: "Alex Balashov" <abalashov at evaristesys.com>
> To: "OpenSIPS users mailling list" <users at lists.opensips.org>
> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [OpenSIPS-Users] SIP Trunking
>
> Matthew,
> 
> Look for the mediaproxy module.
> 
> That said, do be aware that a proxy is, by definition, not like an
> SBC. 
>   SBCs have many other capabilities a proxy does not;  a proxy is a 
> relatively "thin" interoperation layer.
> 
> Perhaps the recently introduced b2bua module is brought to bear on
> that 
> somewhat, but classically, OpenSIPS is a proxy.
> 
> -- Alex
> 
> Matthew S. Crocker wrote:
> 
> > Hello,
> > 
> >  I'm brand new to OpenSIPS, just going through the make process now.
>  
> > 
> >  I need to configure OpenSIPS to act like a SBC for some SIP trunks
> coming off a VoIP switch.  Where should I look for
> Documentation/Examples of a working config?
> > 
> > Here is my scenario:
> > 
> > OpenSIPS has two interfaces,  private & public.  
> > VoIP Gateway is on private LAN with no gateway configured (it can
> only talk to local machines, no routing)
> > 
> > End user has an Asterisk server on a private lan behind their
> firewall (NAT)
> > 
> > I need to configure OpenSIPS to listen for SIP messages on :5060
> from the end user firewall.  It then need to rewrite the SIP message
> and send it to the Gateway.  The Gateway would see the messages coming
> from the internal IP of the OpenSIPS server.  Once all of the SIP
> messages get processed I then need the OpenSIPS server to proxy the
> RTP streams (plan on using mediaproxy) between the Asterisk server and
> VoIP Gateway.
> > 
> > Any helpful hints on where to look?
> > 
> > -Matt
> > 
> > 
> 
> 
> -- 
> Alex Balashov - Principal
> Evariste Systems
> Web     : http://www.evaristesys.com/
> Tel     : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
> 
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-- 
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760