[Asterisk-doc] docs introduction.xml,1.6,1.7
blitzrage
asterisk-doc@lists.digium.com
Tue, 1 Jun 2004 12:02:12 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv30164/docs
Modified Files:
introduction.xml
Log Message:
Changes contributed by Martin List-Petersen to the mailing list on 31 May 2004.
Index: introduction.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/introduction.xml,v
retrieving revision 1.6
retrieving revision 1.7
diff -C2 -d -r1.6 -r1.7
*** introduction.xml 1 Jun 2004 00:32:08 -0000 1.6
--- introduction.xml 1 Jun 2004 12:02:04 -0000 1.7
***************
*** 260,275 ****
</sect2>
<sect2>
! <title>Channels</title>
<sect3>
! <title>Zaptel Devices/Channels</title>
! <para/>
</sect3>
<sect3>
! <title>The IAX/IAX2 Protocol</title>
! <para/>
</sect3>
<sect3>
! <title>SIP</title>
! <para/>
</sect3>
<sect3>
--- 260,304 ----
</sect2>
<sect2>
! <title>Protocols/Channels</title>
<sect3>
! <title>Zap Devices/Channels</title>
! <para>
! Zap channels (Zapata/Zaptel) are the channel-type, that is used for FXO, FXS
! and PRI-cards. There is also a third-party module, that implements zap
! channels for certain BRI ISDN cards.
! </para>
! <para>
! The Zap channels were originally given the name by the Zapata Telephony Project,
! which is an effort to bring affordable computer telephony to the public domain.
! This is happening because the commercial marked is ridiculous expensive and often
! offers poor support. Besides that, it might be worth to mention that Zapata
! was a Mexican revolutionary.
! </para>
</sect3>
<sect3>
! <title>The IAX Protocol</title>
! <para>
! IAX stands for InterAsterisk eXchange and was developed as an alternative to SIP
! and H.323. Currently there exists 2 versions of IAX, where IAX2 is the most common
! used. IAX is not submitted by any standards group, but is currently being adopted
! by different manufacturers for both soft- and hard-phones.
! </para>
! <para>
! The biggest advantage for IAX is that it only uses 1 UDP port and thus works
! very well behind NAT firewalls. It allocates only the the minimum of bandwidth,
! that is used at any time.
! </para>
</sect3>
<sect3>
! <title>The SIP Protocol</title>
! <para>
! SIP, or the Session Initiation Protocol, is specified by the IETF. It allows text,
! voice and multimedia sessions and uses port 5060 udp and tcp, but may use other
! ports. Most VoIP devices on the marked currently support this protocol.
! </para>
! <para>
! This protocol is not always easy to deploy in a firewalled environment, but with
! the help of a STUN server not impossible.
! </para>
</sect3>
<sect3>
***************
*** 278,283 ****
</sect3>
<sect3>
! <title>H323</title>
! <para/>
</sect3>
<sect3>
--- 307,338 ----
</sect3>
<sect3>
! <title>The H.323 Protocol</title>
! <para>
! H.323 is specified by the ITU-T (International Telecommunication Union Stanardization
! Sector) and was meant for teleconferencing (Speech and Video). It basically should
! enable you to terminate voice, video, fax and much more over IP, depending on what
! features your client offers.
! </para>
! <para>
! There are 2 implementations for H.323 that can be used with asterisk:
! <orderedlist>
! <listitem>
! <para>
! asterisk-oh323 - This was the first channel module available to asterisk, that
! implemented H.323. It simulates a pseudo soundcard implementation to pass audio
! from asterisk to the open h.323 stack.
! </para>
! </listitem>
! <listitem>
! <para>
! chan_h323 - This channel module is part of asterisk now, it uses asterisk RTP stack and
! implements H.323 in one shared library.
! </para>
! </listitem>
! </orderedlist>
! You might ask yourself now, what module you should choose and quite frankly, there is
! no all-round answer to that. Implement whichever you have in hands and test it, if you
! not are happy with that, try the other one.
! </para>
</sect3>
<sect3>