[Asterisk-Dev] STUN for asterisk as SIP client

[Goldenear]

Eric Wieling aka ManxPower a écrit :

> 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.
> __

IMHO this is only a work around and not a true solution:

externip= need a fixed IP address and that is not always the case.
forwarding the RTP ports need a lot of work on the router, can have a 
security issue and may be a problem if the asterisk box is not the only 
sip client behind the NAT.

STUN would be a really better and cleaner solution. Every modern SIP UA 
has STUN support, so why not asterisk ?
STUN support for asterisk has been discussed since a long time.. why 
isn't it implemented yet ? is it so difficult to add STUN support in 
chan_sip ?