[Asterisk-Dev] STUN for asterisk as SIP client

[Goldenear]

Hi,

my * box is inside a private network behind a "restricted port cone" 
NAT. I'm successfully using it to connect to some others * servers or 
clients on the internet: I only need to forward port 4569 on the 
nat/router to get it working, very simple :)
The issue is that I need to connect to some SIP *only* providers.
I also would like my * box to directly make (media/rtp) connection to 
other SIP endpoints (for less delay).
Forward port 5060 on the nat/router to the * box only solve one half of 
the problem: rtp streams won't be managed properly without the help of 
STUN...
So I'm wondering if somebody is still working on a STUN support for 
asterisk, so an * box can act as a SIP client behind a NAT.
Any news/information about this ?