[Asterisk-Users] SIP, Asterisk and NAT

[Zoa]

Im trying to write some tutorial for these ever recurring SIP + NAT
questions.
Its far from ready, and its without layout, but the draft can be found
at: http://www.asteriskguru.com/natut.php it has all most of the
situations explained, and explains all the options you need to look at
in the asterisk config files.

/Z

Olle E. Johansson wrote:

> Irakli Natsvlishvili wrote:
>
>> 100k question - does asterisk correctly handle following situations:
>
> There are plenty of good documents on Asterisk, SIP and NAT on the
> voip-info.org wiki. Please look them up. There are also information
> within the configs/sip.conf.sample file within Asterisk.
>
>> 1. Asterisk is on a public IP
>>    Two SIP clients on separate networks, each of them are behind
>> dynamic NAT
>> gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go
>> thought
>> asterisk.
>
> If the media stream SHOULD NOT go through Asterisk, then it's up to
> the phones
> to support NAT traversal properly and handle this, it's not an
> Asterisk problem. From Asterisk's point of view, we should not see
> that they are in fact behind NAT. Modern phones in combination with
> STUN and a decent NAT device supports this.
>
>> 2. Even worst case -  three clients, two of them on one site, second
>> is on
>> another site. For example extensions 500 and 600 are on the same site
>> and in
>> the same subnet and extension 1000 is on another site/network. There
>> are PAT
>> FW/gateways with dynamic public IP in front of clients and those are
>> symmetric NAT/FW.
>>
>> The task - clients registering on Asterisk server, calling each other
>> and
>> RTP should not go via asterisk. So, media stream should go directly
>> from one
>> client to another.
>
> If Asterisk is on a public IP, again: it's up to the phones. It's
> still not
> an Asterisk problem.
>
>> I want to know:
>>
>> 1. Is it possible? - yes/no. Implementation should involve asterisk
>> and SIP
>> clients and not involving third party hardware products - ALG, session
>> border controllers or so on.
>
> Yes, but you need to pick the right phone, the right NAT/FW and have a
> lot of patience :-)
>
>> 2. If it is possible, what are requirements for SIP clients.
>
> Good NAT traversal support.
>
>> 3. What configuration changes should be done on Asterisk server and
>> on a sip
>> clients.
>
> From Asterisk's point of view, all of these phones are on a public IP
> and we
> do not give them any NAT traversal support. If you want detailed
> configurations, there are several consultants available that can help
> you with that (including my company).
>
>> And final question - if it is NOT possible with Asterisk, do you know an
>> open source product which works in above stated scenarios and you've
>> actually tested it.
>
> It is possible with Asterisk and every other SIP server. With your
> requirements, it's completely a client-side problem.
>
> Best regards,
> /Olle
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