> > Yes. Exactly, That is why incoming call has a problem. > However if there are a lot of endpoints behind same NAT then there will be > a problem > configuring NAT in such way. Too difficult to configure NAT on each > customer side. IMO, if you need lots of users behind the same NAT, you should use another gnugk behind the NAT, or as the NAT. Or, if you don't need IP phones, you can use a gateway that allows, say, 4 or 8 analog phones (or analog PBX trunk ports) to be connected, but still looks like one endpoint on the H.323 side. Or, as you say, use SIP. Sorry, I don't know of software that can proxy H.323 from behind a NAT, that is simpler than gnugk. However, others on this list may have a recommendation. > How can SIP phones work with Asterisk/SER/sipX without any NAT configuration? SIP uses a single UDP port for registration and signaling. The repeated registrations keep the NAT association open, so signaling for incoming calls is not a problem. The SIP proxy tells the endpoint to send RTP to it, and sends RTP back to whatever port it sees it coming from. This is the same behavior as gnugk in ProxyForNAT mode. However, it means that the caller can't hear anything until some RTP has been sent out. Many H.323 endpoints won't send RTP until Connect has been received, so you don't get to hear call progress tones or announcements. Most SIP clients are smart enough to send RTP as soon as a 183 Progress message comes in. Regards, Stewart ------------------------------------------------------- SF email is sponsored by - The IT Product Guide Read honest & candid reviews on hundreds of IT Products from real users. Discover which products truly live up to the hype. Start reading now. http://ads.osdn.com/?ad_id=6595&alloc_id=14396&op=click _______________________________________________________ List: Openh323gk-users@xxxxxxxxxxxxxxxxxxxxx Archive: http://sourceforge.net/mailarchive/forum.php?forum_id=8549 Homepage: http://www.gnugk.org/
