[asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities
wouter at schoot.org
Wed Aug 4 09:44:44 CDT 2010
I'm trying to get Asterisk to work dual-stack on Linux and I'm left with
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?
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