[OpenSIPS-Users] OpenSIPS ALG

[Jeff Pyle]

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
>> 
>> 
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>> 
>>   
> 
> 
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