[Asterisk-Users] SIP, Asterisk and NAT

Michael D Schelin mike at shelcomm.com
Tue Apr 26 18:30:25 MST 2005


You are talking about a sip proxie server. I don't like ser. I use a 
full commercial proxie that works great but it's expensive. I believe 
asterisk can do what you want but I'm not sure. I use Sipquest for my 
services. I'm a provider.

Irakli Natsvlishvili wrote:

>100k question - does asterisk correctly handle following situations:
>
>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.
>
>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.
>
>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.
>2. If it is possible, what are requirements for SIP clients.
>3. What configuration changes should be done on Asterisk server and on a sip
>clients.
>
>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. 
>
>Thanks for your help.
>
>Irakli
>
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