[Asterisk-Dev] STUN for asterisk as SIP client
Goldenear
goldenear at free.fr
Wed Apr 27 13:22:46 MST 2005
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).
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