[Asterisk-Users] A solution for SIP and NAT

[Patrick]

On Wed, 2003-07-02 at 04:26, John Todd wrote:
> You may be correct about the Via: header, but you're incorrect in the 
> concept as to how it relates to Asterisk, notably in your reversal of 
> what side of the transaction is putting data in the Via: header to 
> make SIP work correctly.
> 
> This is cluttering up the list.  Talk to me off line if you want a 
> better understanding of how NAT and SIP work with Cisco devices.
> 
> Again, for those of you who might be trying to figure out what the 
> result of this conversation is:  SIP clients behind NAT works fine in 
> both directions (incoming and outgoing calls), Asterisk makes it 
> work, it's not using STUN.  Cisco devices work especially well.
> 
> JT

Hi John,

Thanks for the very helpful info so far. I concur with Richard
Alexander's request to keep this discussion on list. 

How about Asterisk and NAT? Can you please comment if the examples below
also work.

1x SIP phone <-> NAT box <-> Internet <-> NAT box <-> Asterisk
10x SIP phone <-> NAT box <-> Internet <-> NAT box <-> Asterisk

The SIP phone(s) and Asterisk server are on private IP addresses. The
NAT boxes (e.g. adsl router) have a public IP address. Any requirements
for the NAT boxes like being a SIP proxy?

Thanks,
Patrick