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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga > Sent: Wednesday, September 24, 2003 11:02 PM > To: [EMAIL PROTECTED] > 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. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
