[Asterisk-Users] A solution for SIP and NAT

[John Todd]

I'm uncertain why you're not able to get SIP working for your user 
agents (SIP clients.)  With Cisco equipment, as an example, it works 
quite well and almost every 79xx or ATA-186 I have is behind a NAT, 
and this configuration is duplicated across a dozen or more systems 
now running behind almost every conceivable NAT/PAT situation*

Known working config:

UA -> (NAT) -> Internet -> Asterisk

Can you be more specific about your problems with SIP?  Perhaps you 
have done so in the past, but re-state and maybe someone can see what 
the problem is.

JT


*Note: the Cisco PIX, while supposedly SIP-friendly, has been the one 
box that has not worked with NAT/PAT SIP sessions.  I have not been 
the admin on that system, but a fairly clueful Cisco wrangler has 
been unable to make it work for originating calls in both directions 
- only one-way origination works.)


>Hi all.
>
>I have come to the conclusion that there just isn't anything out there
>for allowing SIP and NAT to work together nicely. This is rather amazing
>considering that as far back as March 2000 there are documents
>describing how to do it.
>
>So I've started a really simple SIP and RTP proxy project, SaRP, on
>sourceforge.net. Yesterday we uploaded 0.2 of the perl based release.
>This is the first general release and should work for most people. We
>are using it quite successfully for standard calls between all sorts of
>NATed clients. All you need to do is forward UDP/5060 from your
>firewall/router to the box running SaRP if you want incoming calls to
>work and also allow UDP traffic from the ports listed in the config file
>out.
>
>The project can be found at http://sarp.sourceforge.net/
>
>I would be very interested in any feedback you may have.
>
>Regards
>
>Andrew Radke.
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