Alan Lord (News) wrote: > On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote: >> Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID >> calls, originating and transferring. >> >> A provider offers both SIP and IAX trunking. Cateris paribus, what is >> the preferred solution to choose? What points to consider? > > We use IAX trunks from our provider primarily as they are so much easier > to configure and you only need one port open on your firewall/nat gateway. > > SIP needs hundreds, if not thousands of open ports IIUC.
Yes, SIP requires very good, precise and complex stateful inspection from a firewall. You need 2 ports for each RTP stream, and these should be maintained by the firewall as media circuits are established and torn down. Do you know about any voice quality issues with IAX, especially with a big call volime? -kkm _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users