[OpenSIPS-Users] SIP Trunking

[Ghaith ALKAYYEM]

Hello,

I tried to use mediaproxy, it includes two softwares (dispatcher &
relay), I tried a lot to run more than one relay on the same server in
order to bind them to different interfaces. But unfortunately this
didn't work and I think it's not possible.
I recommend using RTPProxy which is designed to work in bridging mode
between two networks and you can run multiple instance of RTPProxy on
the same server.

Regards.


On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote:
> Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
> Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 & eth1) ?
> 
> If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1).
> If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues.
> 
> EndUser <-> (eth0) MediaProxy (eth1) <-> SIP Gateway
> 
> 
> ----- "Jeff Pyle" <jpyle at fidelityvoice.com> wrote:
> 
> > From: "Jeff Pyle" <jpyle at fidelityvoice.com>
> > To: "OpenSIPS users mailling list" <users at lists.opensips.org>
> > Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern
> > Subject: Re: [OpenSIPS-Users] SIP Trunking
> >
> > Matthew,
> > 
> > While I'm no Mediaproxy expert, I have seen many conversations on this
> > list
> > where Mediaproxy is described as a part of a far-end NAT solution.  It
> > was
> > not designed to have a private IP attached to it.  For that, you most
> > likely
> > will want to look at the rtpproxy application.
> > 
> > It sounds like you are constructing a local ALG to connect private
> > and
> > public networks.  You don't necessarily need a full-blown Acme for
> > that.
> > I've had great luck with Edgewater Networks' "Edgemarc" devices, for
> > example.  That's just one.  There are many.
> > 
> > 
> > - Jeff
> > 
> > 
> > 
> > On 8/20/09 2:49 PM, "Matthew S. Crocker" <matthew at corp.crocker.com>
> > wrote:
> > 
> > > 
> > > 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
> > >> 
> > >> _______________________________________________
> > >> Users mailing list
> > >> Users at lists.opensips.org
> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> > 
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