[Asterisk-Users] No sound with SIP Phones on the Internet

[Olle E. Johansson]

Chris Hariga wrote:

> This is bull... I can't believe that...
Read more on NAT and Voip here:
http://www.voip-info.org/wiki-NAT+and+VOIP

It's not simple. The server must be reached in order to allow registration for
clients. A server inside a NAT can't be reached unless you use port forwarding.
A client inside can reach a server outside and there are ways to keep a connection
open in the NAT. Asterisk does not support this very well.

The other side of the coin is the contstruction of SIP and SDP. That's a long
story, but it ends in something like:
Asterisk help clients on the inside of a NAT with NAT=yes in SIP.CONF, but
can't work the other way as of today.

 > Must be a solution...
Do some research, add som code and we will all be happy!
IPv6 is a solution, if NAT is avoided.

/O


> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of WipeOut
> Sent: Monday, October 13, 2003 9:57 AM
> To: asterisk-users at lists.digium.com
> 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..
> 
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-- 
*** Olle E. Johansson, oej at edvina.net

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