[Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

In your second option using a STUN server would I need to setup my own STUN
server?

Thanks

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 12:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal

> Thank you for your reply.  There is a wealth of information on the
> wiki, etc.   I turned on RTP debug and the SPA is not sending it's
> public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP
> packets are going nowhere... 

Do I understand your question correctly:

You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both
devices register, but calls between the devices result in no audio?

If that is the case, you can do one of two things:

- set canreinvite=no for the devices' sip.conf entries, or
- teach both devices to *stop* using their internal IPs for all
communications and remove nat=yes from the entry for the SIP device inside
NAT2.

To set the SPA to give the correct IP, enable STUN, add a STUN server, and
say Yes to "Substitue VIA Addr".

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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