[Asterisk-Users] SIP / GrandStream Configuration

[Anthony Wood]

On Thu, Sep 25, 2003 at 12:33:02AM -0400, Uriel Carrasquilla wrote:
> Adam:
> I believe you.  I assume that the RTP is creating a symetric configuration
> between * and the SIP phone.  The situation we are left to live with is that
> * (won't be the Sip phone) can only live in the Internet brave world (and
> not behind a firewall).  is this acceptable?
> Uriel

You could set up a tunnel between both NATed networks so they didn't need
to use NAT.  Assuming there are no IP conflicts.  This will add
some latency, I'm not sure how much.

cheers,
Woody

> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Stephen Varga
> Sent: Wednesday, September 24, 2003 11:02 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration
> 
> 
> On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote:
> > Adam:
> > in reference to my first message, the NAT on the SIP/GS (a D-Link router)
> > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
> > forwarded to the Sip/GS.
> > The Asterisk server, also behind another NAT (Linksys), has the same ports
> > opened and forwarded.
> > is it still impossible?
> > URiel
> 
> Nope, it is not currently possible. * behind a NAT for SIP does not work
> because the * real IP address is placed in the SDP information,
> therefore the 'outside' phone can not send the media stream to *. See my
> answers over the last week for the more details and possible work
> arounds.
> 
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-- 
Woody