[Asterisk-Users] SIP and NAT - best practices?

[Michaël Gaudette]

Thanks Moises.  I was kind of hoping that, at least if I hosted my Asterisk
server somewhere where there was no NAT for the * box that the SIP phones
wouldn't create any issues.  

How do you people with Hosted PBX handle the deployment of SIP phones behind
NAT firewalls? Is it just elbow grease and configuring every single phone
for the customer, or is there a way?

Mike



you can redirect the ports of the router as well. Or you can configure
your SIP phone to use a STUN server. Please read in voip-info.org
about SIP NAT, there are good suggestions.

regards

On 1/20/06, Michakl Gaudette <michael.gaudette at virtutel.ca> wrote:
> Hello,
>
> I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my
> wholesale provider.  That worked, fine.  I ahd to open up the ports on my
> router, forward them to the correct box, again fine.
>
> Now, if I get one of my customers to connect his SIP phone to my Asterisk
> box, and HE'S behind a NAT firewall, does he have to go through the same
> process, or is it just the Asterisk box that needs to translate the SIP
and
> RTP port?
>
> In other words: if my SIP phone is behind a Linksys router, do I need to
> configure the Router for any reason?
>
> Mike