[asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities

[Wouter Schoot]

Dear list,

I'm trying to get Asterisk to work dual-stack on Linux and I'm left with 
a question.

Imagine that a user (on the road) connects to Asterisk from various 
places. Many of them probably don't have IPv6 support yet. However, his 
house and office do have IPv6 connectivity. I would like to make sure 
that whenever IPv6 is available, the connection will be made over IPv6, 
but offer IPv4 as a "fallback" option.

The pitfall, in my opinion, is to create one sip.conf entry for that 
user which supports the voicecalls over IPv4 and IPv6. However, settings 
like nat=, directmedia= and/or canreinvite= seem to be addressfamily 
unrelated. I want to configure it in a way that when I connect using 
IPv6, no NAT options should be set and the mediapath (almost) always 
should be directly between the peers and not over the Asterisk server 
(so, "nat=no" and "canreinvite=yes").

But, when a user comes via IPv4, changes are that he's on NAT. When that 
happens obviously the connections should traverse the NAT using options 
like "nat=yes" and "canreinvite=no".

There's little to no documentation available as far as my google-skills 
go. There's some in sip.conf, and I couldn't find anything on the website.

Does anyone have some pointers for me, either for the configuration of 
the sip.conf entry or for more documentation on this?

Best regards,

Wouter Schoot