[Asterisk-Dev] STUN for asterisk as SIP client

[Goldenear]

Chih-Wei Huang a écrit :

> Olle E. Johansson wrote:
>
>> Kevin P. Fleming wrote:
>>
>>> Goldenear wrote:
>>>
>>>> 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 ?
>>>
>>>
>>> You are aware that Asterisk is an open source project, right? Things 
>>> don't get implemented just because they've been 'discussed for a 
>>> long time', they get implemented because someone with the desire to 
>>> do so does the dirty work and writes the code. (or, someone who 
>>> wants it bad enough pays someone else to write it for them)
>>
>> I fully agree that it would be helpful to have it and fully agree with
>> Kevin - a lot of people has asked for it but no one seems to want to 
>> fund the development work for it...
>
>
> Agree + 1
>
> By the way, IMO, you don't need to wait STUN support for Asterisk.
> Because
>
> * STUN is not the right solution for NAT traversal. It won't work
> for many situations.

It won't work for symmetric NAT only. any other NAT type should work.

> * The best solution without modifying your router/FW/CPE, etc
> is using an RTP proxy. Install SER with RTP proxy and have Asterisk
> outbound route calls to SER. It works like a magic!
>
Installing an RTP proxy on the SIP server/proxy (e.g. with SER) increase 
the need of bandwidth on the server and will add delay. Direct/P2P 
connection between UA is a far better solution IMHO. RTP proxy should 
only be used when STUN fails (e.g. with symmetric NAT).