December 1997

BY DAVID MISUNAS, MICOM COMMUNICATIONS CORPORATION

Voice over Internet Protocol, or VoIP, technologies unite the telephony and data worlds, permitting phone calls, fax calls, and voice traffic to be transmitted over corporate enterprise networks, Intranets, and the Internet. Designed to operate in a wide variety of network environments, the common thread between VoIP products is their focus on using IP as a voice transport mechanism, providing transparent, business-quality communication in corporate Intranets and lower-cost, reduced-quality operation in the Internet.

The key motivation for VoIP technology in all network environments is toll-call cost savings. These savings are realized by a company calling its stores in the next town over its internal data network or by a family calling relatives on the other side of the world. Once the data connection is paid for — in the case of the company, it’s already in place to service the organization’s data needs, in the case of a family, it’s through their Internet link — the call is free.

VoIP FOR WINDOWS
Microsoft Windows is the most widely used PC operating system, and VoIP technologies and products are now focusing on standards and corporate quality communication within Windows in order to add intracompany voice and fax communication capabilities to the desktop platform, using their existing routed IP network. In addition, the development by Microsoft of voice support for Windows in NetMeeting and the widespread dissemination of NetMeeting clients with Windows have widened the appeal and availability of VoIP applications.

In order to be successful, Windows based VoIP technologies must evolve to include support for voice calls originating from or directed to desktop PC telephony clients, particularly NetMeeting, in addition to supporting connection to standard telephony equipment such as phones, PBXs, key systems, and the PSTN. Companies have worked together since mid-1996 in the VoIP forum [later merged into the IMTC (International Multimedia and Teleconferencing Consortium), and the ITU (International Telecommunications Union)] to develop standards that define common call control and voice compression protocols. In particular, ITU Recommendation H.323 provides cross-product functionality, and many vendors in this market are either utilizing H.323 or are in the process of implementing it.

H.323
ITU recommendation H.323 describes equipment and services for multimedia communication over localarea networks (LANs) which do not provide a guaranteed Quality of Service (QoS). H.323 networks carry real-time voice, data, and video, or any combination of the three through use of the logical channel signaling procedures of Recommendation H.245, in which the content of each logical channel is described when the channel is opened. H.323 also contains procedures for describing receiver and transmitter capabilities, allowing transmis-sions to be limited to what receivers can decode, and allowing receivers to request desired modes of operation (e.g., telephony) from transmitters. A corporate VoIP network utilizing H.323 is shown in Figure 1. In addition to H.323 as a common basis for transport mechanisms and basic call management, both G.729 (compression at 8 kbps) and G.723 (compression to 5.3 kbps and 6.3 kbps) have been defined as baseline voice compression coders to be used for audio calls over an H.323 connection. Although the IMTC selected G.723, and the ITU recommended G.729 for audio-only calls, both coders have the potential for widespread deployment by VoIP vendors, including those developing PC telephony clients.

VoIP CALL TOPOLOGIES
Early VoIP products provided the fundamental mechanisms to support voice and fax calls over an IP data network. A typical VoIP phone/fax IP gateway, provided a dialing plan that defined a called number format consisting of a pair, a Phone Directory Database that handled gateway IP address resolution via the translation between traditional telephony dialed digits and IP network addresses, and call management with VoIP transport mechanism. In such systems, the VoIP product was principally a gateway that was located between the IP network and a conventional telephony connection, providing call setup and transport service translation between the two networks.

The major limitation of these early products was that they were based on proprietary protocols and did not allow interoperability outside of their specific domain. H.323 broadens the space and allows interoperability between the different products by use of standardsbased call setup, transport layer call management and directory services, and introduces new requirements in the Dialing Plan, Directory Services, and Transport Layer Call Management mechanisms that broaden the scope of the VoIP application, in particular in consideration for PC phone client support.

DIALING PLANS
Implementation of VoIP for Windows continues to support the essential gateway-to-gateway applications that allow calls from standard telephony endpoints (e.g., phone handsets and fax machines), including both a basic dialing plan and extended features, such as force connect and hunt group options. These possibilities are reflected in the following potential VoIP call scenarios:

Gateway to Gateway Calling
In this application, a call is originated at a telephone at one site, calling a destination phone, with the transport services provided by VoIP gateways directing the call over an IP network. This allows a configurable multi-step dialing process that defines how a call should be switched, depending on caller input on the originating phone’s DTMF keypad. VoIP routes the call to the appropriate remote VoIP gateway that will provide access to the remote PBX, key telephone system, or other telephony device.

Desktop PC to CPE Phone
VoIP also provides the ability for an H.323 desktop PC phone client, such as a Microsoft NetMeeting client, to place calls to phones connected to a PBX or key telephone system, or a standalone analog phone, or to call out to the Public Switched Telephone Network (PSTN). To support this application, a desktop PC phone client will be expected to allow any standard phone number to be entered by the caller to indicate the called-party destination.

Thus, key to the success of VoIP within Windows is a Directory Service (DS) infrastructure that can translate phone numbers to IP addresses, as well as provide dynamic IP address assignment for desktop PC phone clients and VoIP gateways (that must use dynamic IP address assignment on a corporate network). Most Windows VoIP systems address this issue through the use of the Lightweight Directory Access Protocol (LDAP). Issues relating to scaling, strong authentication, and extensibility are addressed in LDAPv3, including the dynamic updating of directory information needing to resolve the location and addressing of mobile users such as those utilizing NetMeeting PC clients. A solution based on LDAP will be easily deployed and maintained by corporate customers, building on directory service infrastructures being created today.

In addition, the creation of a User Locator Service (ULS) as part of VoIP Directory Services adds the ability to resolve the issue of multiple telephony devices for a person, forwarding calls to the currently “active” device, and allowing callers to use a Personal ID number to call a specific individual over the VoIP network. This allows VoIP administrators the option to assign phone numbers to individuals as their VoIP Personal IDs.

VoIP WINDOWS ARCHITECTURE
The basic architectural organization of a Windows-based VoIP product is shown in Figure 2.

VoIP compression and telephony services are provided by a plug-in card for the PC that contains DSPs to perform the voice compression/transmission and call control functions. Call Setup and Card/Channel Control are Windows applications that communicate with kernel-based functions managing the DSP-based plug-in card.

Functions performed by the card driver upon request by the Card/Channel Control module include configuration control, status requests, connection establishment, and card reset/addition/ deletion. The Call Setup module communicates with the Card Driver to place calls and handle directory services. A third Windows module, the Transport Driver, operates in the kernel to manage the voice transport through Windows with minimal delay.

Figure 3 shows the provision of Directory Services for VoIP within Windows, supporting the call setup among VoIP clients as well as handling connections from PC NetMeeting clients to standard telephony services and equipment. In addition, the Directory Services module will make the characteristics of all clients transparent to the caller, enabling a standard numbering system to be used regardless of the destination of the call.

Directory Services consists of three components: a Database Engine, Gateway Address Resolution, and Host Registration. The Database Engine stores user information for call setup, while the Gateway Address Resolution performs address translation of the database information for call setup. Host Registration allows the VoIP system to properly interact with Internet telephony meeting places. The Database Engine provides the foundation for the Directory Service, while Gateway Address Resolution and Host Registration utilize its information for setup of the operational environment and address resolution.

Through use of a standardsbased approach based on H.323, VoIP for Windows provides the ability to link PC-based NetMeeting clients with standard telephony equipment and applications and to greatly enhance the capabilities, interoperability, applicability, and savings of VoIP systems.

David Misunas is vice president of product development for MICOM Communications Corporation, a Nortel company. MICOM manufactures two main product lines which enable companies to dramatically reduce compa-nywide communication costs: network-ing solutions that integrate data, voice, fax, and LAN over public frame relay and private line networks; and phone/fax IP gateways that add a voice/fax overlay on top of any IP network. With a twenty-three year history in the communications market, MICOM is recognized among the worldwide leaders in providing data/voice network integration products. For more information, visit the company’s Web site at www.micom.com.