[Asterisk-Users] SIP and NAT - best practices?

[Pavel Jezek]

I thing, that configuring nat device/firewall at consumer site isn't 
always possible, thus simplest (but not optimal) way is to configure 
phone in sip.conf as nat=yes & canreinvite=no, this should work in most 
cases even if multiple phones are behind same nat, like adsl router.
disadvatage is, that rtp stream will go always through asterisk server 
(even for calls between phones in same location-behind same nat/fw).
so, as I ask before, if is planned in asterisk development to make 
"canreinvite" function more flexible, e.g. possibility specify that for 
call, e.g. inside one context, to do reinvite and for other calls 
(between different contexts)  don't do reinvite ...
PJ