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@yahoo.com wrote:
Date: Sun, 15 Aug 2004 21:51:44 -0700 (PDT) From: "C.K" ckng128@yahoo.com To: serusers@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
- 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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
C.K