[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[John Todd]

Keywords: SIP, NAT, proxy

The solution can be found if someone wants to take up these requests:

  - http://bugs.digium.com/bug_view_page.php?bug_id=0000359
  - http://bugs.digium.com/bug_view_page.php?bug_id=0000104

JT



>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..
>
>--
>*** Olle E. Johansson, oej at edvina.net
>
>Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
>Runbovägen 10, 192 48 Sollentuna, Sweden
>Phone: +46 8 594 78 810, Fax: +46 8 594 78 820