[OpenSIPS-Users] SIP Trunking

[Jeff Pyle]

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
>> 
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