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Open source tools such as Asterisk and SIP Enterprise Router are facing their biggest test yet at the University of Pennsylvania, where deployment of 15,000-seat VoIP network based on these open source IP telephony servers is rolling along.
The Philadelphia-based Ivy League university 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.
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 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.
Optical fiber supporting multiple wavelengths of Gbps Ethernet over dense wavelength division multiplexing links UPenn's campus to a carrier hotel in Philadelphia, which provides ISP and SIP trunking services. A Session Border Controller sits at the edge of UPenn's VoIP network and peers with SBCs from Level 3 to connect campus SIP endpoints to IP and PSTN endpoints beyond the university.
The new method of tying a campus VoIP network to an ITSP’s VoIP service was a more complex and challenging process than plugging into Centrex services, but worth it, Kassabian says.
"There's quite a bit of testing and certification that goes on in both directions with an ITSP," Kassabian says. "There's a whole series of certifications to make sure we can handle the set of situations that come up in the range of calls that will happen" between the ITSP’s SIP SBCs, and the ones in UPenn's data centers. "There's several dozen inbound and outbound tests, and we work with their engineers."
However, the new service offers more flexibility, gives more control to the school's IT staff and lowers costs.
Comments (4)
RE: Open source VoIP hits the Ivy LeagueBy meatpieandtatters on July 18, 2007, 6:22 pmVery nice article!!! congs! Re: Open source VoIP hits the Ivy League.
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Open source VoIP hits Ivy LeagueBy DogLoverInPhilly on August 1, 2007, 3:14 pm(UPDATED on August 1, 2007 to clarify a few things that didn't strike me as exactly what I wanted to say and how) Meatpieandtatters says the article was "very...
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Open Source Early Adopters and Next PhaseBy Bob Murphy on October 18, 2007, 11:00 amThis is another real world example of the viability of Open Source Telephony for practical application. I think what worked out so well here was the bench depth...
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Very nice article,By josueconti on November 22, 2008, 11:13 pmVery nice article, Congratulations Regards!
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