[asterisk-users] Do I need a sip proxy?

[Bruce B]

Thanks a lot for the great input Pan.

I think you are right on point with this one. I have STATIC PORT enabled in
my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it
is there for a reason.

So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it
though. If I have the Siproxd enabled, does it act as a one single server
that connects multiple times to my provider or providers and then I connect
to the Siproxd in return? Or, I can still register from Asterisk directly
with the provider(s) and Siproxd will take care of the SIP packets to be
handled nicely?

If it's the latter then it sounds fine to use otherwise it would not only be
complicated but also a downtime to Siproxd mean downtime to all Asterisk
servers.

***In addition I have setup Siproxd according to pfsense guide online but
once I save the configurations and return to it there are no configs left. I
know this question is for pfsense forum but maybe someone else experienced
this?

***And to return to my original question, do I need a SIP proxy and which
one would be suit my needs? I still like to get an input on my previous
e-mail. I have to stay with pfsense for now as it has proven to be a good
router in all other aspect.

Thanks,

On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen <pan at ibidium.no> wrote:

>   Hello Bruce,
>
> Your understanding of NAT is correct, and your setup should work.
>
> I’m not familiar with Pfsense, but I suspected that your problem was due to
> a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of
> VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably
> rewriting the SIP packets in addition to the IP packets. Try reconfiguring
> Pfsense or swapping it for something else. A good way to troubleshoot your
> scenario is to compare the traffic in your end to the traffic on your
> providers end (or on either side of pfsense). Pay attention to the source
> and destination IP and ports in addition to the contents of the SIP
> messages.
>
> http://doc.pfsense.org/index.php/VoIP_Configuration
> http://en.wikipedia.org/wiki/Application-level_gateway
>
> With kind regards,
> Pan
>
>  *From:* Bruce B <bruceb444 at gmail.com>
> *Sent:* Tuesday, January 11, 2011 8:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com>
> *Subject:* [asterisk-users] Do I need a sip proxy?
>
> Hi Everyone,
>
> I am running multiple instances of Asterisk in Proxmox and so far I had one
> central Asterisk feeding all others with trunks from one provider. Now, I
> want to connect each Asterisk server directly to the provider. Based on my
> understanding, each connection made to the provider port 5060 would be on a
> port that is unique to that server. And so other connections made to the
> same provider will go out through a different port and should receive
> responses through that different port. At least that is my understanding of
> NAT. The provider should see me trying to register from the same IP with
> multiple different ports (high number ports; not talking about 5060 as this
> is outbound and not inbound) and should be able to differentiate between SIP
> packets coming from various servers. However, it seems to not happen.
>
> There is some sort of clash and only one of the servers shows registered
> with the provider and other's trunks go down. I have noticed that keeping
> one server works. It could also be that my Fail2ban kicks in on all servers
> if the SIP packets received are broadcasted to all servers which shouldn't
> really happen and router should take of this by sending it to the server
> that has the established connection through that port.
>
> *My equipment:*
> Asterisk 1.6x
> Pfsense 1.2.3
> Dumb Switch
>
> *My questions:*
> A- What is the rational behind this?
> B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or
> Kamailio?
> C- Which one of the above is the easiest to get running given I never tried
> any of those.
> D- If I am doing an SIP proxy server then it might have to also be
> redundant. What options do I have in that and which of above or any other
> suggested package might be great for future expansions.
>
> Clarification on how NAT would work in situations like this would be much
> appreciated.
>
> Thanks
>
> ------------------------------
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110111/134d4786/attachment.htm>