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By Phil Hochmuth
Network World, 12/24/01

While Ethernet and IP have helped simplify data networking to the point of near-universal plug-and-play, getting different vendors' IP telephony wares to work together has been a trickier enterprise. But the IETF's Session Initiation Protocol holds great promise for changing that.

Users and industry watchers say SIP could empower telephony the way HTTP did the Web - make interoperability a given, while adding new features and applications. Unified voice, video and instant-messaging applications are in the offing, as are outsourced voice-over-IP services. Simply put, "voice should be an application on your network, just the same as e-mail," says Patrick Olson, CIO of Menlo College, an Atherton, Calif., school that runs a voice IP network. Just as e-mail clients from different vendors can communicate is how voice over IP should work, he adds.

A SIP at a time

Today, much of SIP's promise for enterprise use resides in carrier voice-over-IP services. Through such services, network executives can forego installing PBX-based voice infrastructures for outsourced voice services delivered over IP.

Users get SIP-based IP phones and, in some cases, a voice-over-IP-enabled router or gateway from the carrier. All calls are processed on switches in the carrier's central office, as they are for traditional Centrex service.


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WorldCom is preparing an outsourced voice-over-IP service - what it calls IP Centrex - for availability in January. It will use SIP-based phones and gateways from 3Com, Cisco and other vendors.

With a service such as the one planned by WorldCom, users not only could find contact phone numbers in a company directory, but also place the calls. They'd get unified messaging with a single in-box for voice, e-mail and fax. Calendar integration features would remind users of a phone meeting, then contact and set up all the parties, says Theresa Hastings, multimedia product engineering director for WorldCom.

With WorldCom's IP Centrex, users also will get more control over their voice network than they have with traditional Centrex. Through a Web interface, they can add and remove users or change features and applications. In the old Centrex world, these tasks could take weeks, Hastings says.

This self-provisioning will be the biggest plus over traditional Centrex, says Steve Blair, senior network engineer at the University of Pennsylvania, a beta-test site for WorldCom's IP Centrex service.

"We're hoping WorldCom's [IP Centrex] will provide the level of service our customers are asking for while improving our process for activating, changing and deactivating [phone] services on campus," Blair says. Of course, he adds, the university also is excited about the potential of not having to buy a new PBX or phone switch.

Besides WorldCom, carriers with SIP-based service plans include Broadwing, GoBeam, Net2Phone and TalkingNets.

SIP on the rise

For now, user organizations that prefer building their own voice IP networks are limited in their choices of SIP gear. Most major voice-over-IP vendors, including 3Com, Avaya, Cisco, Nortel and Polycom, rely on the ITU H.323 standard because it's been tried and proven in ISDN videoconference applications.

"[H.323] works, and it's deployed," says Tom Valovic, an analyst at IDC.

Some vendors also run proprietary protocols in their voice-over-IP gear. Cisco voice-over-IP phones and servers come with a default protocol called "Skinny." Users can choose among Skinny, H.323 and Media Gateway Control Protocol. 

SIP can also co-exist in a network with Cisco CallManagers by adding a SIP proxy server, available from Cisco, 3Com and others. Open source options are also available for organizations that want to build their own SIP servers. Columbia University in New York has developed H.323-SIP gateway software that can be used to register SIP phones directly to a Cisco CallManager.

Menlo College uses Skinny and SIP. The school uses the default Skinny protocol on some Cisco 7960 IP phones because it's easier to deploy and manage on a LAN than SIP, and is less taxing on a CallManager server in terms of processing power, Olson says. But a majority of handsets rely on SIP via a server running Columbia's H.323-SIP software, which ties into CallManager. In a nutshell, "You can do more with SIP," he says.

In cases where voice-over-IP phone makers support or plan to support Java and XML, that means the option of developing special applications. Java applets running on SIP phones could deliver stock reports, the weather conditions or corporate announcements on the LCD. Or, using a SIP phone's touch screen, a teacher could enter daily class-attendance records into a database. The ability to use applications such as these on a voice IP network will eventually give SIP a leg up over H.323, experts say.

Another reason SIP has not yet broken the enterprise voice-over-IP market wide open is that the early voice-over-IP landscape has mirrored the traditional telecom model of closed systems, says Brian Strachman, a senior analyst with Cahners In-Stat.

"You can't have a system with a Nortel [IP PBX] and plug in Cisco or 3Com phones," Strachman says. "Frankly, most vendors don't want to be interoperable. No one wants to say, 'Go ahead and buy our IP PBX phone system, and oh, you can use Cisco or 3Com phones with it too.' "

Standard telecom practice in large companies has been to buy PBXs and phones from the same vendor, mostly for interoperability reasons.

"It's the telecom mindset, but it's been changing over the past five years," says Strachman, adding that SIP could become a driving force for making enterprise voice-over-IP networks as interoperable as Ethernet-based client/server LANs. "Eventually they will be open. You'll just have to give it a few years before you can go out and build a LAN telephony system piecemeal with phones and switches from different vendors."

Microsoft takes a big SIP

Vendor commitments to SIP are growing. Besides Cisco, Pingtel and Siemens already support the IETF protocol. Avaya says it plans to have SIP integrated into its IP600 IP PBX and phones by mid-2002. 3Com, which already sells SIP-based phones and SIP servers for service providers such as WorldCom, says it will support SIP in its enterprise NBX in 2002.

And Microsoft has thrown its considerable clout behind SIP by including the protocol as a piece of its new Windows XP operating system. The protocol carries IP voice and video traffic in the new Windows Messenger, an XP application for real-time voice, video and instant-messaging communications.

Windows Messenger not only will integrate multiple real-time communications applications, but also add "presence" to the software. Presence is an instant-messaging feature that notifies users who on their "buddy lists" is available to chat. Windows Messenger will use the SIP for Instant Messaging and Presence Leveraging (SIMPLE) extension to bring buddy lists to voice and video users registered on a service provider's network or in an enterprise environment.

"One of the biggest benefits of SIP is the addition of the SIMPLE extension and presence," says Imad Yanni, product manager for XP at Microsoft. "In NetMeeting [Windows Communicator's H.323-based predecessor], it was difficult to find someone to start a session . . . most people would end up calling each other first, then starting the [videoconference] meeting. And asking for someone's IP address is not practical."

Going one step beyond the "SIP on every desktop" future that it anticipates with XP, Microsoft also has outlined plans for Windows Real Time Collaboration, a SIP-based server application for letting many kinds of devices communicate. The SIP server could be used in a company as a voice, video and text messaging hub for myriad devices. Microsoft has not announced when it plans to release the product.

Menlo College's Olson likes the fact that his SIP-enabled network will be ready to handle new SIP applications, such as Windows Messenger in XP. Olson says he could see the school offering students the choice of an IP phone in their dorms or a phone application based on Windows Messenger and SIP.

"Ultimately what we want is interoperability so we can have anyone's phone, anyone's [IP voice] switch on our network," Olson says, noting that while Menlo College's voice-over-IP network is Cisco-centric now, he's looking to add Pingtel SIP phones soon.

What he wants, Olson says, is the ability going forward to pick and choose products from multiple vendors, whether he's looking for new features or lower prices. "SIP is what will let us do that."

Related links

Contact Senior Writer Phil Hochmuth

Other recent articles by Hochmuth

SIP Forum
Industry group.

VoIP vendors pass SIP test
Devices based on Session Initiation Protocol offer basic interoperability, but challenges remain for advanced features.
Network World, 08/27/01.

SIMPLE, but more complex
The instant-messaging interoperability wars are far from over, but the smoke is beginning to clear from some of the nastier standards battles.
Network World, 10/08/01.

That future was a thing of the past
Was SIP the cause of Sprint's expensive ION failure?
Network World, 11/05/01.

Videoconferencing research page

VoIP Security Products Buyer's Guide
Jul. 09, 2009

VoIP Analysis and Management Tools Buyer's Guide
Jul. 08, 2009

Logitech Vid Offers Free & Easy Video Calling
Jul. 07, 2009

Outremer Telecom's Mauritius plans on hold
Jul. 03, 2009

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