16 Aug
2004
16 Aug
'04
3 p.m.
Hello all,
Additional infor to below is I could run the sipsak
successfully. but just no audio could pass through the
NAT.
[root@detone stund]# sipsak -T -s
sip:1008@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@219.95.43.92:5060;user=phone>
[root@detone stund]#
--- "C.K" <ckng128(a)yahoo.com> wrote:
Date: Sun, 15 Aug 2004 21:51:44 -0700 (PDT)
From: "C.K" <ckng128(a)yahoo.com>
To: serusers(a)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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