Internet Telephony show focuses on open source

Expect open source VoIP gear to make a splash as the Internet Telephony East Conference and Expo opens its doors this week in Fort Lauderdale, Fla.

Vendors including Signate and Ranch Networks are announcing products that rely on open source VoIP code, and the show is hosting a certification program for open source PBX project Asterisk.

With about 8,500 attendees expected and 120 exhibitors signed up, Internet Telephony, now in its fifth year, is one of the largest trade shows dedicated to VoIP.

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Expect open source VoIP gear to make a splash as the Internet Telephony East Conference and Expo opens its doors this week in Fort Lauderdale, Fla.

Vendors including Signate and Ranch Networks are announcing products that rely on open source VoIP code, and the show is hosting a certification program for open source PBX project Asterisk.

With about 8,500 attendees expected and 120 exhibitors signed up, Internet Telephony, now in its fifth year, is one of the largest trade shows dedicated to VoIP.

The focus on open source PBXs is reflected in the chosen keynote speakers as well, with the leaders of the two major IP PBX projects - Asterisk and SIPfoundry - slated to talk alongside executives from established voice vendors AT&T, Avaya, Cisco, Level 3 Communications, Nortel and Time Warner Telecom.

Mark Spencer, author of Asterisk and president and CEO of its commercial counterpart Digium, will deliver the leadoff keynote about how a company using VoIP on its private network can expand that to include VoIP services that take calls to the traditional public phone network.

William Rich, president and CEO of PingTel, promoter of SIPfoundry, will talk about how IP communications is broadening to more than just VoIP and how open source projects can contribute to that expansion.

The show is sponsoring a program in which attendees take a one-day course on Asterisk software and receive certification that they are proficient with the platform. The certification is designed for corporate IT professionals looking to make informed decisions about IP PBX technology.

During the week, Signate will launch SigPro, its open source-based IP telephony platform designed for carriers. The product consists of software on off-the-shelf, Intel-based mid-range computers and supports as many as 500,000 customers.

SigPro utilizes Asterisk, the open source PBX, and the SIP Express router, an open source Session Initiation Protocol (SIP) server, says William Boehlke, president and CEO of Signate. The product also employs Apache, MySQL and Linux, as well as Signate's own code, which provides graphical interfaces for adding direct inward-dialing numbers and directing calls to call servers.

SigPro can support commercial VoIP services, including hosted PBX with an auto attendant, call-detail records, call forwarding and call parking.

The attraction of the platform is that the code is accessible, says Jason Cohen, president and CEO of FutureLink, an IP phone provider in New York. "Our CTO wants to make sure he can get in and change the source code himself if necessary rather than rely on a manufacturer," Cohen says. Before deciding on SigPro, FutureLink considered gear from BroadSoft, NexTone, Sylantro and VocalData, he says.

The cost of new features from vendors of VoIP gear based on proprietary code can be high, says Will Stofega, research manager for VoIP services at IDC. With open source, carriers can write code themselves to add functionality. "They can develop new and interesting features without incurring the costs," he says.