SIP normally doesn't use TCP, it uses UDP, and is sessionless in that context. The exact mechanics of a registration can get deeply involved, so I'm going to give a very cursory overview. The endpoint tells the server (Asterisk, or whatever) that it would like to register, with a username and password, and what its IP address and port are. The server puts this in a list, and when it has a call for that endpoint, sends UDP packets to the known IP and port. There it typically encounters a NAT rounter, which had opened the port during the original registration and hopefully still has it open.
You can enable a feature called NAT keep-alive on most endpoints to overcome bad NAT in some routers. On Mon, Nov 14, 2011 at 2:51 PM, Douglas Mortensen <[email protected]>wrote: > I know this is probably a very basic question for many on this list. But > in troubleshooting an issue, I wanted to take a step back & ask the > question. In Asterisk (or maybe all SIP), how do extensions stay registered > with the SIP server?**** > > ** ** > > Do the extensions simply register repeatedly as a means of telling > asterisk “I’m still here”, or are there actual keepalive packets that are > transmitted to actually keep a TCP session alive? My guess is the former.* > *** > > ** ** > > But am I oversimplifying it? Is there more to the process?**** > > ** ** > > Thanks,**** > > -**** > > Doug Mortensen**** > > Network Consultant**** > > *Impala Networks Inc* > > CCNA, MCSA, Security+, A+**** > > Linux+, Network+, Server+**** > > .**** > > www.impalanetworks.com**** > > P: (505) 327-7300**** > > F: (505) 327-7545**** > > .**** > > ** ** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Carlos Alvarez TelEvolve 602-889-3003
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
