SIP is a XML-like control channel and is used to negotiate a separate RTP channel which carries the audio. It is complicated to set-up in cases of firewalls and NAT, but is an open standard.
IAX2 is a candidate open standard and merges all traffic onto a single UDP stream - control and audio data. It has two modes, trunk and non-trunk. Trunk mode is highly efficient for transmitting multiple calls on a single UDP bearer and has minimal overhead. Standard IAX2 is easier to set-up than SIP. In terms of user experience, there should be little difference in call handling and audio quality - in general all of the same codecs and features are supported. IAX2 is a native protocol of Digium's Asterisk switch and I believe stands for Inter-Asterisk-eXchange version 2. To answer the query below, IAX (ie IAX11) was the precursor of IAX2. It is obsolete and should no longer be used. I use IAX when referring to IAX2, but obviously not all do! HTH Peter -----Original Message----- From: Serge Schumacher [mailto:[EMAIL PROTECTED] Sent: 31 December 2004 15:41 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX users Sorry ? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: vendredi 31 décembre 2004 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX users IAX2 ----- Original Message ----- From: "Serge Schumacher" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Sent: Friday, December 31, 2004 8:00 AM Subject: [Asterisk-Users] IAX users > Hi, > > I do not understand the difference between SIP and IAX, is it only two > different protocols or something more special. > > The problem I have is that I've created two users > > > Aix.conf > > register => users1:passwd1 > register => user2:passwd2 > > [user1] > type=user > context=default > secret=passwd1 > host=dynamic > > > [user2] > type=user > context=default > secret=passwd2 > host=dynamic > > extensions.conf > > exten => 550,1(Dial,IAX/user1); > exten => 551,1(Dial,IAX/user2); > > and the error I get : > > > Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No > application 'IAX/user1)' for extension (default, 550, 1) > == Spawn extension (default, 550, 1) exited non-zero on > 'IAX2/[EMAIL PROTECTED]:1059/1' > -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' > > Can someone help me how to get both users connected ? > > Thank you, > > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users