On Mon, Jan 22, 2007 at 09:59:06AM +0000, Tim Panton wrote: > In the meanwhile, use IAX, which understands about NAT pretty well. > If you have multiple SIP phones on a LAN behind a NATing router, just > put a small asterisk box on the LAN. It can manage your hairpin > calls internally, save you bandwidth by trunking the IAX traffic > to the central asterisk and avoid all the NAT hassle by using > a single port (outgoing) and refreshing it often enough for the > router to hold it open. > > > Tim Panton > > www.mexuar.net > www.westhawk.co.uk/
IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. For a remote phone, not on the same network as the Asterisk box (in which event the NAT worries are different) you definitely want to use the same protocol for the phone as for your term/orig provider. Otherwise you will be forced to hairpin your audio through your asterisk server, adding latency and wasting bandwidth and cpu for little reason. In addition, many people just want to do things like give family or employees a phone they can take home, or take to a remote location and use on the PBX. They probably can't "just" put up an Asterisk server to make this happen, and nor should they want to. An additional server is not only more work and requires an always-on server computer, it's another thing that can go wrong. No thanks. Even if you can run Asterisk on a WRT54G, and thus don't have the $200/year power expense of a server, it's still not what you really want. IAX is great but SIP is also a reality, and putting Asterisk into the "just works" category is a really important milestone. One I think that is intended to be improved a lot for 1.6. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
