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

[Krystian Filiks]

Apart of what everyone writes with the NAT=YES I would suggest using 
canreinvite=no as well as normally asterisk cans the reinvite and this 
might cause the audio not to get through the NAT  and cause dead air for 
the users specially if the users are behind 2 seperate NAT servers eg. 
different private networks.

By using canreinvite=no and nat=yes most of the NAT problems go away.

In this scenario the example would look like this:

[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
*nat=yes
canreinvite=no*



Mark Phillips wrote:

> 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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