Open source VoIP/Telephony

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.