[Asterisk-Users] Asterisk behind NAT to SIP provider

[Olle E. Johansson]

Rich Adamson wrote:

>>>I experimented a little bit and Asterisk behind NAT with SIP works. I created 
>>>an account at iptel.org and use that account for outbound SIP traffic from
>>>Asterisk.
>>
>>Great! I copied your information for other users to the Wiki.
>>
>>http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER
>>
>>Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind
>>console keyboard, but anyway...
> 
> 
> There has been a fair amount of discussion on the list as to whether nat
> works with various/different configurations of sip phones with *. The
> "exact" configuration required is highly dependent on a number of technical
> factors that must be well understood before anyone can make a generic
> statement relative to whether it works or doesn't work. Without that
> understanding, practically every statement made on the list has been 
> based on opinion and/or some trial & error methodology that has resulted
> in a working example. (Nothing wrong with that, but the majority of the
> postings leave out critical info that causes the next person to attempt
> the same implementation but fails, and additional questions are generated.)
Rich,
Thank you for your additional information on the NAT/VoIP issue. Is it ok
with you if I add it to the Wiki?

As you say, we need to collect information and compose a data base of
what works and what's not working in certain circumstances.

Jan got * -> SER working, I can't. We have different NAT:s. To try to
solve my problem I made sure his solution was documented so far.

There's no silver bullet here. With NATs, we've built a network without
end-to-end connectivity and we need to  patch it up to get VoIP working
on an IPv4 network with NATs in every corner.

I just hope that IPv6 will make life easier for the next generation of
VoIP users. Right now, we need to understand all variables.

/O