Fwd: [Serusers] Asterisk inside a NAT, client inside ANOTHER NAT

[C.K]

Hello all,

Additional infor to below is I could run the sipsak 
successfully. but just no audio could pass through the
NAT.

[root at detone stund]# sipsak -T -s
sip:1008 at 202.129.171.223
warning: IP extract from warning activated to be more
informational
0: 10.38.38.14 (3.749 ms) SIP/2.0 483 Too Many Hops
1: 219.95.43.92 "detected NAT type is full cone"
Contact (102.951 ms) SIP/2.0 200 OK
        Contact:
<sip:1008 at 219.95.43.92:5060;user=phone>
[root at detone stund]# 

--- "C.K" <ckng128 at yahoo.com> wrote:

> Date: Sun, 15 Aug 2004 21:51:44 -0700 (PDT)
> From: "C.K" <ckng128 at yahoo.com>
> To: serusers at lists.iptel.org
> Subject: [Serusers] Asterisk inside a NAT, client
> inside ANOTHER NAT
> 
> Hello,
> 
> By looking at this section from the link
>
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
> 
> 9. Asterisk inside a NAT, client inside ANOTHER NAT
> 
> In this case, we need a middle man to even find each
> other, an outbound SIP proxy that handles the SIP
> transaction and is reachable by all parties. To get
> media streams from point to point we need another
> middle man, a media server. Asterisk could be that
> media server, that could add media codec conversion.
> Portaone's rtpproxy works together with SIP Express
> router as a media server in this situation. 
> 
> Could anyone share the configuration on how to do
> this
> ? I could only succeed if I put on port forwarding
> on
> the UA's end.
> 
> Regards, C.K
> 
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