Source: snapvoip.blogspot.com

The Philadelphia-based Ivy League university, University of Pennsylvania, currently has over 1,250 Session Initiation Protocol (SIP) IP phones on desktops, tied to a back end based on SIP Express Router — an open source VoIP call-control and routing stack, and Asterisk for voice mail messaging. Don’t go away! This is just the start. The University has extending the VoIP network in to a 15000 seats, in it’s plans!

Deke Kassabian, the university’s senior technology director for information systems and computing, plans to grow that installed base by a factor of more than 10 over the next five years. Driving the project is the desire to get off costly Centrex monthly fees and infrastructure, and the promise of an open source, standards-based VOIP infrastructure that provides superior integration and control.

"If we can run one modern IP network for voice, video and data …. there’s a clear win," Kassabian says. "If we provide business telephony internally, less money leaves the university."

The Linux-based SER call control and Asterisk messaging servers were a better fit with UPenn’s standard back ends for authentication (Kerberos and RADIUS), its OpenLDAP directory structure, and e mail. While commercial IP PBXs are adaptable to these platforms, "they don’t work that way out of the box" typically, he adds.

With open source running extensively throughout the university — from directories, to e-mail, DHCP and DNS — the level of expertise in open source troubleshooting and development was there to support the Asterisk plans, Kassabian says.

"For years UPenn has had a strong open source talent pool. As a result, we have the staff and expertise to develop and roll out open source VOIP."

UPenn’s work with the Asterisk community is also paying off by improving the product itself. University programmers have already contributed to two additions to the code base, which is now supported in the main release. One change integrates IMAP-based voice mail and messaging stores, and another involves improvements in SMDI signaling between IP phones and voice-mail system back end.

"Instead of going off and making changes ourselves," Kassabian says, "we get our changes built into [the code base] and don’t have to maintain them ourselves. They’re part of the next distribution."
The infrastructure Kassabian and his team built is designed for high-availability VOIP, with redundant connections to IP call and feature servers, PSTN and IP telephony service provider (ITSP) point-of-presence links. Two data centers on campus host redundant clusters of Asterisk boxes, SIP proxy servers, and media/messaging feature servers. Phones on the network can register to and access any set of servers. "In this way, there’s no single failure, and no single site failure that would take out the servers," Kassabian says.
For outbound calling, UPenn is using a mix of VoIP and PSTN services. For long distance service and other calls, UPenn is plugging its campus VoIP network directly into a SIP trunking service from Level 3. A pair of dedicated Cisco 3600 routers also support PSTN links for local calls, and as a back for ITSP service.

Kassabian and his staff, mostly with IP networking backgrounds, also had to get up to speed with voice system jargon and terminology before being able to understand user needs. "We had to learn about how people use their phones," Kassabian says. "I had to learn what a bridge line appearance actually was."

Like many large organizations converging voice and data networks, consolidating the school’s telecom and data network teams helped tremendously.

"Having people from our traditional telecom organization learning IP technology has been great," he says. "And our networking staff has been learning telephony. That’s all been part of it. As time goes on, more of us are more well cross-trained and the two technologies come together very well."

More information at UPenn voice