[OpenSIPS-Users] OpenSIPS ALG

[Bogdan-Andrei Iancu]

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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