a little newbie question:
I've just installed asterisk and played a little with it. the server has a pubblic address while the clients (sjphone, msn messenger, sipset) are behind a firewall/NAT. sip part always works, while rtp part sometimes works, sometimes not. the question is: does asterisk decapsulates data coming from client 1 and re-encapsulates them in the call-leg from asterisk to client 2 so the problem could be caused by work overload in the server? or can it make data pass through?
the second question (that is related) is: I would like to send video and I set sip.conf with video. I saw that asterisk has plugins for h.261, h.263: what should I do if I want to send h.264 or MPEG4 for example? if asterisk decodes and re-encodes I MUST have a specific plugin, but if it only takes the data from the first call-leg to send them on the second one I don't need it...
thanks for any hint,
Alberto Forchino
On Mon, 13 Oct 2003 17:32:46 +0100, WipeOut <[EMAIL PROTECTED]> wrote:
Chris Hariga wrote:
There is a solution.. buy a SIP aware router with a built in SIP proxy.. But even then you will probably still have issues..This is bull... I can't believe that... Must be a solution...
Chris HARIGA
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Monday, October 13, 2003 9:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No sound with SIP Phones on the Internet
Chris Hariga wrote:
Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to my Linux box.There is your problem.. Asterisk does not like playing behind NAT.. The UA's can be made to work behind NAT but the server must have a public IP address..
Search the archives and you will see that this issue has come up time and time again and I have not heard of anyone who has managed to get Asterisk to work correctly when the Asterisk server is behind NAT..
If the SIP UA is also behind NAT then there is even less chance of it working..
Believe it, Don't believe it its your choice..
Later..
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