[asterisk-users] SIP tunnel

[mosbah.abdelkader]

> Problem is that the port 80 you are talking about is a TCP port.  Voip
(iax and rtp) use UDP


< Yes true. HTTP uses 80 TCP port.

I mentioned port 80 as example (even if it can be used for SIP signaling:
SIP supports also TCP). For RTP, UDP must be used. We can use another well
known UDP  port.

But, from other replies from the asterisk community, the use of well known
ports does not solve thye problem in all cases. Because in some scenarios
the firewall inspects the traffic and cuts it off if it discovers that it is
corresponding to a voip traffic.

Some users have recommended to me the use of the VPn technology through the
use of openvpn open source tool.


I will try to use it and give the results of the work to the asterisk
community.

I thank a lot all the community for its very good and professional help.

I am really pleased by that.

-- 
*Please discover scientific miracles of CORAN*

http://www.55a.net/
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