[OpenSIPS-Users] OpenSIPS ALG

Fri May 22 12:12:35 CEST 2009

Hi Robert,

Interesting points of view :).

But there is a huge difference between a proxy acting as a mid-registrar 
(with no auth knowledge, no user knowledge, but simply following the 
master registrar decisions in a blind way) and proxy doing actually 
account and registrar management (like you described in the scenario 
with hiding registrations).

I'm not saying is not possible, but the question is where a proxy stops 
and where an ALG starts :).

Anyhow, I will add on the TODO list the possibility to do a register at 
reply time, so that we can use opensips as mid-registrar.

Regards,
Bogdan

Robert Dyck wrote:
> I like the idea that we could maintain a local registrar that accurately 
> reflects the remote registration. I would go so far as to say that it would 
> be useful to be able to optimize opensips as an ALG. Some people could use a 
> proxy as an edge device on a LAN. You could have several phones with the same 
> user name register with a service provider. Some service providers limit the 
> number of contacts per AOR. This would have the side effect of keeping the 
> signaling associated with forking on the LAN. By detecting local calls, media 
> could also be kept on the LAN. Such an edge device might have a dynamic IP 
> address. It would be helpful if opensips could conveniently be setup to fix 
> the contacts. Using the received address does not work when the phones are on 
> the same LAN as the proxy.
>
> Opensips has a bias toward service providers. This is quite understandable. 
> However I feel with a bit of tweaking it could serve as an ALG for the home 
> owner or a small business.
>
> On Wednesday 13 May 2009, Bogdan-Andrei Iancu wrote:
>   
>> Hi Jeff,
>>
>> theoretically yes, because you have all the needed information
>> (hmm...maybe except the NAT status from request, but you can store it
>> via transaction)....Practically, the save() function does expect to
>> receive a request only, so it must be changed to work with a reply also.
>>
>> Regards,
>> Bogdan
>>
>> Jeff Pyle wrote:
>>     
>>> I've thought a lot about this as well, although I haven't taken it nearly
>>> as far as John has.
>>>
>>> A thought:  is it possible to do a save() in the reply route, only upon a
>>> 200 OK from the end registrar?
>>>
>>>
>>> - Jeff
>>>
>>> On 5/12/09 5:09 AM, "Bogdan-Andrei Iancu" <bogdan at voice-system.ro> wrote:
>>>       
>>>> Hi John,
>>>>
>>>> This mid-registrar approach may work but it is not 100% correct as
>>>> OpenSIPS (as mid-registrar) does not obey the actions of the final
>>>> registrar (Asterisk). Ex:
>>>>     - Asterisk may forbid the registration and you already saved the
>>>> registration on OpenSIPS
>>>>     - Asterisk may change the Expire time while to saved the
>>>> registration with the expire sent by client.
>>>>
>>>> Anyhow, ignoring this aspects, lets go further :) :
>>>>
>>>> 1) is the registration scenario working ok? if not what is the exact
>>>> problem (some trace will help).
>>>>
>>>> I will wait for you answer before moving further with the calling stuff.
>>>>
>>>> Regards,
>>>> Bogdan
>>>>
>>>> John Morris wrote:
>>>>         
>>>>> After several days of playing with OpenSIPS 1.5.0 and RTPProxy 1.2.0, I
>>>>> have a partially working SIP+RTP ALG configuration, and have gotten
>>>>> stuck. I could use some general advice from the list.
>>>>>
>>>>> The company has an Asterisk/FreePBX server on an internal network, and
>>>>> the CEO wants to use a SIP phone from outside.  Because the sip alg
>>>>> iptables module isn't working, and in preparation for another project,
>>>>> I started investigating OpenSIPS for use as a border proxy to connect
>>>>> phones across NAT (and, the next project, to route a SIP trunk over a
>>>>> VPN from the network of a DSL+phone company that intermittently blocks
>>>>> SIP traffic in hopes of plugging revenue leaks).
>>>>>
>>>>> The network looks like this:
>>>>>
>>>>> SIP UA <-> home NAT gateway <-> Internet <-> OpenSIPS server/NAT router
>>>>> <-> Asterisk
>>>>>
>>>>> The standard opensips.cfg file doesn't work as is.  The SIP phone needs
>>>>> to register to the Asterisk server directly.  In addition, it seems
>>>>> there is extra logic needed to support multiple network interfaces
>>>>> (mhomed=1 only partially solves the problem).
>>>>>
>>>>> The way I've gone with this in testing is to relay REGISTERs to
>>>>> Asterisk, but after a save("location","0x02") to enable a
>>>>> lookup("location") on messages originating from the PBX.  The phone is
>>>>> configured with an outbound proxy, and all packets to the proxy
>>>>> matching "uri==myself" are thrown away.  This worked great on the
>>>>> single-interfaced, internal test installation.  Now that there are
>>>>> multiple interfaces involved, things are breaking again; ACKs and BYEs
>>>>> are sent out the wrong interface, and RTPProxy is behaving strangely in
>>>>> bridged mode.
>>>>>
>>>>> There seem to be no good configuration examples for either multi-homed
>>>>> proxies or for proxies that relay REGISTERs.  This makes me think that
>>>>> I'm going about this the wrong way.
>>>>>
>>>>> Also, I have looked at other software, like siproxd, opensbc and uh,
>>>>> that other b2bua that functions as an SBC, but none of those seem to
>>>>> allow this REGISTER pass-through function.
>>>>>
>>>>> What is the best approach for this scenario?  The above approach of
>>>>> relaying REGISTERs to Asterisk?  Is there maybe another approach where
>>>>> phones register to OpenSIPS directly, and OpenSIPS in turn somehow
>>>>> sends another REGISTER to Asterisk?  Or am I missing the idea
>>>>> completely?
>>>>>
>>>>> I'd appreciate general pointers about how to proceed.  I've been
>>>>> putting some Asterisk and FreePBX tutorials and CentOS RPMs on
>>>>> http://www.zultron.com, mostly aimed at small office-like environments.
>>>>> Looking through various lists, this seems a highly sought-after
>>>>> configuration.  If I succeed, I'll document it in hopes of filling the
>>>>> gap in this sort of example.
>>>>>
>>>>> Thanks
>>>>>
>>>>>     John
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>           
>>>> _______________________________________________
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>>>>         
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