[Asterisk-Users] grandstream asterisk configuration

[Mike Machado]

On my handytone, if I did not enable STUN, the * box would send the RTP
data to my 192.168 address, even though I had nat=yes in sip.conf and
the SIP handshake happened with my public IP. It seemed * was not
properly translating (ignoring) the IP that was in the SDP into the IP
that the SIP handshake happened on.

When I enabled STUN, * then started to send the RTP to the correct IP
and port, but I still did not have audio. That was as far as I got and
have not had time to run additional traces to see what the holdup is
now.

On Wed, 2004-01-14 at 08:45, SW wrote:
> Hi,
> 
> In my experience with GS phones, you need STUN support to make it work
> properly (behind NAT), otherwise you would need lot of trial end error to
> figure out how to do port forwarding. If you have STUN you wouldn't need to
> touch the Netgear (except for firewalls).
> 
> If you can't run your own stun server (need two public IPs) then use one of
> many STUN servers out there on public internet.
> 
> For an example enable NAT traversal on your GS phone and point the STUN
> server to one of these STUN servers
> 
> larry.gloo.net or stun01.newkinetics.com.
> 
> Then reboot the GS and see how it discover the NAT (top of the gs web GUI).
> If it is not a full cone or UDP blocked then you should be fine (Netgear is
> restricted cone).
> 
> Cheers
> 
> SW
> 
> 
> From: "Chandra" <chandra at digital.com.np>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] grandstream asterisk configuration
> Date: Wed, 14 Jan 2004 19:35:48 +0545
> Reply-To: asterisk-users at lists.digium.com
> 
> i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to
> grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i
> have also opened all 5060, 5000-5008 ports in my firewall configuration.
> grandstream uses 5004 port for rtp.
> 
> what am i missing here? please tell me.
> 
> chandra
> 
> 
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