[Asterisk-Dev] STUN for asterisk as SIP client

[Eric Wieling aka ManxPower]

Goldenear wrote:
> 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 ?

RTP streams are properly managed using localnet=, externip= and 
forwarding the RTP ports.