[Asterisk-Users] SIP, Asterisk and NAT

[Irakli Natsvlishvili]

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