Convergence on tap for apps

SAN JOSE – News out of last week’s Voice on the Net show might signal that long-promised converged voice/data applications are just around the corner.

Siemens, Microsoft and others laid out application plans, while a host of speakers discussed the role that Session Initiation Protocol (SIP) would play in enterprise integration efforts. Conference sessions – many of them well-attended – dealt with everything from the current state of voice over IP (VoIP) to where the technology is going.

For its part, Siemens displayed OpenScape, a new suite of SIP-based software tools that make it possible to deliver a consolidated view of various back-end communications and collaboration systems. End users employ what Siemens calls a Personal Communications Portal to access Microsoft Exchange e-mail, calendars, instant messaging, presence-based telephony features, and voice and videoconferencing.

Users can list their status (for example, “in the office,” “working remotely,” “unavailable”) and input their preferred phone number (desk, cell, remote office, home), and see similar details for colleagues. Buddy lists – similar to AOL’s Instant Messenger – show who is available via what media.

The XP-based portal includes click-to-contact features, letting phone or video calls, chat sessions or e-mails be placed by clicking on names in a Microsoft Active Directory-based listing. Conferences can be established by clicking on multiple names. OpenScape has integrated support for WebEx’s Internet-based whiteboard and document sharing service.

At the heart of the suite is unified messaging middleware that runs on Microsoft’s soon to be released Greenwich Real Time Communications (RTC) server above Windows 2003 Server. An integrated SIP gateway makes it possible to interface OpenScape to IP and traditional telephony systems from Siemens and other vendors, even if they are not SIP-based, says Mark Straton, senior vice president of global marketing at Siemens Information and Communication Networks.

OpenScape is in alpha testing now, with a beta-test version due next month. It will be generally available in the third quarter for about $250 per seat.

With the release of OpenScape, “Siemens is trying to recraft itself into more of a software company, as opposed to a traditional PBX maker,” says Brian Riggs, a senior analyst at Current Analysis.

This approach is a trend among other PBX vendors, such as Avaya, Nortel and Alcatel, he says, as they position themselves to compete with Cisco, whose strategy relies entirely on IP and convergence applications.

Basing OpenScape on SIP was a good move by Siemens, Riggs says, but it raises the question of when, or if, the company will fully integrate the protocol into its HiPath IP PBX line, which uses a mix of proprietary protocols and H.323 for call control.

He adds that OpenScape is a good start, because it separates Siemens from its rivals, who have been slow to release SIP-based gear: “Implementing SIP halfway is more than what a lot of other vendors have done.”

Although Microsoft wasn’t talking applications, it did showcase development tools for the forthcoming Windows CE .Net 4.2 that will let manufacturers integrate voice support into IP-based clients.

Ultimately Windows CE-based IP phones and other devices might be able to place calls over wireless LANs, according to Scott Horn, director of the Embedded and Appliance Platforms Group at Microsoft.

Microsoft will provide in Win CE 4.2 a telephony user interface that can be used to build features such as custom telephony application interfaces.

The software also includes a VoIP Application Interface Layer, an application suite and a set of APIs that support SIP. This will allow devices based on Win CE 4.2 to serve as clients to Microsoft’s Greenwich RTC server.

The Win CE 4.2 software, which is scheduled to ship this summer, also will include Enterprise Infrastructure Integration services intended to simplify the integration of computer telephony software and enterprise applications.

Microsoft hopes Win CE 4.2 will foster the integration of data and voice, and provide capabilities such as managing voice mail and e-mail from a single device or creating voice-driven interfaces to back-end services such as databases.

This recent movement to integrate SIP support in applications is a positive development, Mark Katsouros, communications automation engineer at the University of Maryland in College Park, said after the show.

“SIP seems to be the most efficient way of implementing VoIP,” Katsouros said. “It has a lot of advantages over proprietary protocols” and older VoIP standards, promising improved interoperability and capabilities that extend to other types of messaging beyond voice.

“As the technology develops, you’ll see [VoIP] creeping out more to the edge in the form of IP phones and applications on desktops,” he adds.

Other announcements at the show relating to SIP and VoIP technology included:

• A SIP-based multipoint control unit from RadVision – the ViaIP MCU – for controlling videoconferencing sessions among SIP-based Windows Messenger clients.

• Polycom’s SIP-based SoundStation IP telephone, with support for multiple lines, and enhanced sound quality through embedded quality-of-service mechanisms.

• Citel’s CitelLink IP Handset Gateway, which could be used to let digital Nortel phone handsets receive SIP-based IP-based IP Centrex services from carriers. The product was announced with support from softswitch maker Sylantro Systems.

SIP, today and tomorrow

One expert speaking at VON said the increase of SIP applications is a good start but we’ve only scratched the surface.

In a VON keynote presentation, Henning Schulzrinne, associate professor of computer science and electrical engineering at Columbia University and co-author of the IETF RFC outlining SIP, said SIP’s contribution would be its ubiquity. It will be integrated into all kinds of software and become as much a part of enterprise applications as TCP/IP. “If you are waiting for a killer app for SIP, you will be waiting until you are dead.”

Most interpretations of SIP are too telephony-centric, Schulzrinne said. It’s possible, for example, to add SIP-based real-time communications and voice features to Web browsers or build in presence or chat interfaces to ERP applications.

Adding such services to network applications will become easier as SIP adoption progresses. “We need to get away from the notion that services are created by specialists,” Schulzrinne says. Tools are needed that can allow SIP services and applications to be deployed on a model similar to creating personal home pages.

As for the gains SIP has already made, Schulzrinne said the development of low-cost SIP devices – such as phones and PC software – have made the technology more accessible. “We are getting to the point where you don’t have to put on desktops a PC-equivalent device – in terms of cost – in order to deploy SIP.”

He also discussed improvements in SIP voice quality, mentioning tests done at Columbia in which SIP-to-SIP performance was better than cell phone-to-TDM phone links in terms of milliseconds of delay.

Proprietary? Not me!

While the promise of SIP was touted widely at VON, the practical adoption of standards-based protocols and other VoIP interoperability issues were hotly debated at the Network World IP PBX Showdown, where leading convergence vendors took shots at each other.

The VON Showdown was hosted by Network World Editor-in-Chief John Dix and Mike Hommer, consulting manager for Miercom, and included representatives from Alcatel, Avaya, Cisco, Mitel and Nortel (read more about the Showdown).

As panelists took turns grilling each other over their IP PBX products and strategies, the issue of open protocols – who was using SIP and how much, and the use of proprietary technology – came up frequently. And Cisco took the brunt of the heat.

Jeanne Bayerl, director of product market for Alcatel, brought up Cisco’s use of the proprietary Skinny Call Control Protocol (SCCP), and asked why the vendor did not embrace open standards more.

“Cisco fully supports open standards,” said Bill King, technical marketing manager at Cisco. He said Cisco phones support H.323, Media Gateway Control Protocol and SIP, and said the company supports SIP-based gateways and proxy servers, adding “we’ve shipped more SIP-based products than any other vendor up here.” He also said Cisco’s SCCP is licensed to partners – such as Polycom and Spectralink – and is not as proprietary as critics make it out to be.

Each vendor endorsed SIP as a strategy. Alcatel and Mitel said their IP PBXs could run SIP natively, while Avaya, Cisco and Nortel all said broader SIP support for their gear was in the works.

Avaya and Mitel brought up the closed protocol. Mack Leathurby, Avaya’s director of converged system and unified communication applications, mentioned Cisco’s use of its proprietary Cisco Discovery Protocol as an integral management function, such as an initializing inline power, for its CallManager IP telephony system.

“All vendors use some proprietary protocols,” Cisco’s King said, adding that closed protocols are used to add features that standards such as SIP and H.323 fall short of. And firing back at Avaya’s Leathurby, King added, “I haven’t seen Avaya working to make available its [Digital Communications Protocol],” which Avaya uses in addition to H.323 for adding features to its IP phones.

– Senior Editor John Fontana contributed to this story.