[Asterisk-Users] SIP / GrandStream Configuration

[Uriel Carrasquilla]

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

-----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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