Open source VOIP/telephony

Oliver Rist, InfoWorld

11/08/2005 09:51:52

One of the first open source VoIP projects -- and one of the earliest
VoIP PBXes, period -- is Digium-sponsored Asterisk. A highly mature
platform licensed under the GPL, Asterisk supports almost everything
that even larger enterprises would desire of a VoIP gateway solution,
including voice mail, call forwarding, conferencing, and even IVR
(Interactive Voice Response). It also has call-detail records -- the
golden goose of VoIP -- as well as advanced features suitable for use
in virtual classroom or virtual conference room applications. Its large
developer community contributes still more add-ons for the platform,
both commercial and open source.

But while Asterisk may have been a pioneer, it's certainly no longer
alone. A number of new, competitive open source VoIP platforms based on
the SIP protocol have emerged. Pingtel has released the code to its
commercial SIPxchange PBX, which is currently managed by a nonprofit
organization called SIPFoundry under the name sipX. Although not as
mature as Asterisk, sipX adheres much more closely to the open SIP
standard, giving it greater hardware and software compatibility -- at
least for the moment. The InfoWorld Test Center reviewed both Digium
Asterisk and Pingtel SIPxchange in January.

SER (SIP Express Router) is a close adherent to the SIP standard.
Written in C and issued under the GPL (General Public License), it has
been ported to Linux and Solaris. In addition to acting as an SIP
server, it features gateways for SMS (short messaging service) and IM,
RADIUS accounting and authorization, and Web-based user provisioning.
Commercial products based on SER are available from iptelorg. A
bootable LiveCD version of the software is also available, which has
extended SER to include a much easier Web-based administration tool and
support for general VoIP hardware from vendors such as Cisco Systems
and Mitel.

Yate (Yet Another Telephony Engine) is published under the GPL and is a
surprisingly flexible platform. Fully mature, it includes support for
SIP, H.323, and other protocols, and it runs on either Linux or
Windows. It has all the usual PBX enhancements -- voice mail, call
forwarding, and so on -- but also functions as an IVR server.

Those interested in more robust IVR applications, however, would do
well to seek out Bayonne, the script-driven telephony server of the GNU
Project. Bayonne has a long history and is designed for a wide range of
carrier-grade telephony applications. Commercial support is available
from a number of sources. Bayonne has recently been brought under a
larger GNU Telephony umbrella, which encompasses a number of other free
software projects. There can be little doubt that open source efforts
in this area will continue to progress as interest in VoIP and digital
telephony continues to grow.


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