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First Quarter 1998

IP In The Real-Time World


There are two opposing views as to the probable course of Voice over IP (VoIP) technology. One side believes that the difficulties of achieving high voice quality plus high Quality of Service (QoS) will relegate Internet telephony to the limited niche of low-cost-for-low-quality voice communications. The other side asserts that the days of circuit switching are numbered because of the advantageous economics of packet switching.

Both views are essentially focused on the potential for directly displacing existing voice delivery systems with packet-based infrastructures. This is really much too narrow a context in which to predict the impact of VoIP technology on telecommunications. The real benefits to business lie in the ability to solve business problems that are not addressed efficiently or effectively in a world where voice and data are served by separate infrastructures. The range of potential applications is quite broad, as are the opportunities for economic advantage.

Internet Protocol (IP) is a packet switching protocol that allows terminals to communicate regardless of where they are physically located on the network, and regardless of the computer platforms on which they operate. It was the introduction of this geography and platform independent protocol that helped spur the phenomenal growth of the Internet.

Switching is the term used for a mechanism that allows a terminal -- be it a computer or a telephone -- to selectively address and talk to any one of a number of available target terminals. Although there are many variations of switches, there are only two fundamental types: circuit switches and packet switches. Circuit switching offers an approach optimized for voice communications while packet switching is optimized for data communications.

It has been commonly stated that once voice is digitized and presented to a data switching network, it looks the same as data to the network and therefore presents no special delivery problems. In fact, because voice does look the same as data, it presents a special challenge to the packet switching network. The special needs have to do with the communicating terminals - not with the plumbing that delivers the information.

Data communications can be best described as conversation between machines, and voice communications as conversation between persons.

Data Communications
Machines can be configured to:

  • Rearrange pieces of conversation that arrive out of sequence.
  • Accumulate letters and words received over a period of time and read them only when the word or sentence is complete.
  • Detect that a piece of transmitted information is corrupted or has not been received and request that it be retransmitted.

Voice Communications
Live conversations on the other hand are real-time communication, requiring:

  • That all pieces of the conversation be presented to the listener in the order created (spoken).
  • That all pieces of the conversation be presented to the listener with a minimum delay (nominally less than 250 milliseconds).
  • That the delay be essentially the same for all sequential elements of the transmitted speech (constant bit rate).

These elements comprise the measure of QoS for voice communications, and can be seriously impacted by the network. Satisfactory QoS is inherently guaranteed by the architecture of a circuit switch. It is not guaranteed in a packet switching environment, and certainly not on the open Internet, where indeterminate numbers of users vie for access to common network resources.

VoIP was first used for making voice calls from PC to PC via the Internet. That consumer application has been followed by commercial ventures looking to use the Internet as a "free" medium that bypasses the international carriers. This activity, requiring the placement of gateways at the originating and terminating points of Public Switched Telephone Network (PSTN) access is what has generally been viewed as the primary business case for VoIP. If this application were in fact the long-term primary driver for VoIP technology development, its future growth might well be in question since the current arbitrage opportunities are based on use of the "free" Internet and the current exemption of Internet Service Providers (ISP) from PSTN access charges.

International calling rates are not cost-based. The stroke of a pen can change these rates, and can also impose access charges. Either of these actions can instantly invalidate an established business plan based solely on a phone-to-phone model, particularly since it is not a given that a competing IP-based voice network (offering equivalent voice and service quality) can be built for lower cost than a circuit switched network.

What does drive the considerable VoIP activities are the benefits and business solutions that become available from combining voice and data communications into a single switching infrastructure. The functions of gateways in this model include using an existing data network to bypass long-distance carrier charges, and also include bridging between the IP and circuit environments so that "best of both world" solutions to business problems can be provided.

The economic benefits of building out, operating, maintaining, and administering a single network infrastructure would seem to be self-evident. The ability of VoIP to leverage the advantages of geography independence, platform independence, and multiple simultaneous multimedia (e.g., voice, data, video) connections to create virtual local environments may be less obvious except to those familiar with both the existing telephony challenges and the capabilities of VoIP technology.

With ten to twenty million people already accessing the Web, and that number continuing to grow, the Internet has to be viewed as a huge, growing, international marketplace and not as a competing telecommunications medium.

You can use VoIP to Web-enable an existing call center. The gateway allows a home page visitor to click on an icon to establish a voice connection to the existing ACD. The voice call is presented to the ACD without the caller having to disconnect from the home page or wait for a callback on a second line. The ACD is able to identify the incoming caller by name, address, telephone number, and e-mail address, and push selected URLs (e.g., FAQ pages, ads, etc.) while the call is in queue. When the call is answered, the system can also provide a real-time video of the agent.

In addition to providing an enhanced interactive experience for the calling customer, the call center operations benefit from each call that comes in from the Internet rather than the PSTN.

The list of benefits includes:

  • Caller information is retained even if the caller abandons.
  • 800 number and long distance call-back charges are eliminated.
  • "Free" access is available worldwide.
  • The agent can know what URL the caller is looking at and what pages have been visited since arriving at that Web site.
  • The agent can push URLs to the caller, as well as forms.

The ACD function can be provided within the IP environment. Unlike the scenario above, the gateway is provided here to deliver calls from the PSTN to the ACD rather than from the Internet to the ACD. In this VoIP implementation, the merging of voice and data into a single fabric means that the agent position is a single terminal rather than two terminals, one connected to the LAN and the other to the telephone switch. ACD management and administration functions are considerably simplified, with the ability to create splits, add agents, and reconfigure agent assignments in a matter of only a few minutes. In addition to significantly reducing the cost of operation over the life of the system, response times to changing requirements are reduced from days to minutes.

Another benefit from combining the agent functions into a single multimedia fabric is that it can take advantage of the geography and platform independence afforded by VoIP technology. Truly virtual call centers become possible without the need for high equipment investment or long installation cycles at the remote sites. Agent groups can be connected by dedicated digital links to the main site or  through a basic rate dial-up ISDN modem. Such a connection could support approximately fifteen agents without introducing any QoS issues associated with the Internet. A work-at-home agent could be on a 28.8 dial-up direct modem connection or, if really remote, through the Internet.

Geography independence also brings, for the first time, the feasibility of a viable disaster recovery program. Providing mirrored servers on the IP-connected network allows small, inexpensive remote sites to be configured for instant backup, and even permits agents and supervisors to be fully functional either from home or any other dial-up site.

The ability of a VoIP architecture to use a common infrastructure for common channel signaling and control as well as for telecommunications allows construction of a truly distributed virtual machine that is both scalable and extensible. The fact that it also supports multimedia communications to the desktop within the same infrastructure permits VoIP technology to offer a much richer workstation environment than does circuit switched ISDN. The superset of ISDN, which is characterized by the out-of-band, non-facilities associated signaling and control architecture of the Intelligent Network (IN) (referred to as Signaling System 7, or SS7), operates only to the edge of the PSTN.

The addition of SS7 or SS7 adjunct functionality to an IP-to-circuit switch capability brings with it the ability to allow this robust desktop environment to interoperate seamlessly with the SS7 network.

The first application planned for release by an operating telephone company is a service feature offering called Virtual Second Line. The purpose of this feature is to allow an incoming call from anywhere on the PSTN to reach a telephone subscriber whose line would otherwise be busy because it is connected to the ISP. As implemented, the VoIP application in the subscriber's terminal is seen by the PSTN the same as any standard telephone in the Intelligent Network.

This service is of value to customers because it allows them to stay on the Internet without worrying that they will be unavailable for incoming telephone calls without buying second line service. The carrier, of course, gets to sell a new service. At the same time, signaling traffic is reduced on the network by reduction of retries to the busy station, and the carrier's settlement income is augmented by the resultant increase in long-distance call completion rates.

The telecommunications environment can be viewed as comprising three different domains. The first is the desktop, where LAN and PBX (ISDN) competed in the past and currently coexist. The second is the backbone as represented by the PSTN with its Advanced Intelligent Network. The third is the edge, where the desktop and the backbone environments meet, typically by a PBX connecting to a Local Exchange Carrier (LEC) central office switch.

The merging of voice and data into unified packet-based backbone infrastructures is already well under way as represented by the evolution of SONET (Synchronous Optical Network) and Asynchronous Transfer Mode (ATM) delivery systems.

At the desktop, LAN technology has continued to progress in response to the growing demand for bandwidth. Circuit switching has remained, up to now, the choice for real-time voice communications due to issues of voice quality and the absence of edge technology capable of providing universal access. What we are calling Internet telephony, or VoIP, is directly addressing these issues, and opening up the door to real, multimedia, geographically-independent communications. The above examples are but a few of what will be a long list of drivers for the ongoing development and deployment of this new technology. It is, literally speaking, "leading-edge" technology that is helping to change the way we work, the way we communicate, and the way we build telecommunications systems.

Harvey Kaufman is executive vice president, and one of the founders of NetSpeak Corporation. His background includes more than forty years of experience in engineering and technical marketing management with General Electric Company, NEC, Tel Plus, and Siemens. NetSpeak develops, markets, licenses, and supports a suite of intelligent software modules which enable real-time, concurrent interactive voice, video, and data transmission over packetized data networks such as the Internet and local-area and wide-area networks. For more information, visit the company's Web site at www.netspeak.com.


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