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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines Merge changes from svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc directory into a single LaTeX formatted document so that we can generate a PDF, HTML, or other formats from this information. * Add a CLI command to dump the application documentation into LaTeX format which will only be include if the configure script is run with --enable-dev-mode. * The PDF turned out to be close to 1 MB, so it is not included. However, you can simply run "make asterisk.pdf" to generate it yourself. We may include it in release tarballs or have automatically generated ones on the web site, but that has yet to be decided. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
85 lines
3.3 KiB
TeX
85 lines
3.3 KiB
TeX
\subsection{Introduction}
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This section is intended as an introduction to the Inter-Asterisk
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eXchange v2 (or simply IAX2) protocol. It provides both a theoretical
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background and practical information on its use.
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\subsection{Why IAX2?}
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The first question most people are thinking at this point is "Why do you
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need another VoIP protocol? Why didn't you just use SIP or H.323?"
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Well, the answer is a fairly complicated one, but in a nutshell it's like
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this... Asterisk is intended as a very flexible and powerful
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communications tool. As such, the primary feature we need from a VoIP
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protocol is the ability to meet our own goals with Asterisk, and one with
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enough flexibility that we could use it as a kind of laboratory for
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inventing and implementing new concepts in the field. Neither H.323 or
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SIP fit the roles we needed, so we developed our own protocol, which,
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while not standards based, provides a number of advantages over both SIP
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and H.323, some of which are:
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\begin{itemize}
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\item Interoperability with NAT/PAT/Masquerade firewalls
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\begin{itemize}
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\item IAX seamlessly interoperates through all sorts of NAT and PAT
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and other firewalls, including the ability to place and
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receive calls, and transfer calls to other stations.
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\end{itemize}
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\item High performance, low overhead protocol
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\begin{itemize}
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\item When running on low-bandwidth connections, or when running
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large numbers of calls, optimized bandwidth utilization is
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imperative. IAX uses only 4 bytes of overhead
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\end{itemize}
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\item Internationalization support
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\begin{itemize}
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\item IAX transmits language information, so that remote PBX
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content can be delivered in the native language of the
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calling party.
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\end{itemize}
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\item Remote dialplan polling
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\begin{itemize}
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\item IAX allows a PBX or IP phone to poll the availability of a
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number from a remote server. This allows PBX dialplans to
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be centralized.
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\end{itemize}
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\item Flexible authentication
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\begin{itemize}
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\item IAX supports cleartext, md5, and RSA authentication,
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providing flexible security models for outgoing calls and
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registration services.
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\end{itemize}
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\item Multimedia protocol
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\begin{itemize}
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\item IAX supports the transmission of voice, video, images, text,
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HTML, DTMF, and URL's. Voice menus can be presented in both
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audibly and visually.
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\end{itemize}
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\item Call statistic gathering
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\begin{itemize}
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\item IAX gathers statistics about network performance (including
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latency and jitter, as well as providing end-to-end latency
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measurement.
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\end{itemize}
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\item Call parameter communication
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\begin{itemize}
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\item Caller*ID, requested extension, requested context, etc are
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all communicated through the call.
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\end{itemize}
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\item Single socket design
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\begin{itemize}
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\item IAX's single socket design allows up to 32768 calls to be
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multiplexed.
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\end{itemize}
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\end{itemize}
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While we value the importance of standards based (i.e. SIP) call handling,
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hopefully this will provide a reasonable explanation of why we developed
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IAX rather than starting with SIP.
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\subsection{Configuration}
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For examples of a configuration, please see the iax.conf.sample in
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your the /configs directory of you source code distribution.
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