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

[Mark Phillips]

Most often the simple addition of nat=yes in the relevant sip.conf 
stanza is all that's required to make a remote SIP phone work from 
behind a firewall.

for example

[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes

I've been running 4 remote SIP phones across the internet from my 
families houses all over the world in this manner. The only issues I get 
are those of bandwidth availability or rather occasional lack of it.

Hosted PBX's are no different. The hosting service should be providing a 
similar mechanism (although it might not be Asterisk based).

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Michaël Gaudette wrote:
> 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
> 
> 
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