Growing pains

Communications News, Jan, 1998 by Morris Edwards

It's maturing quickly, but Internet telephony still has a lot of growing up to do.

Until recently, Internet telephony was mainly the province of computer hobbyists who were willing to accept poor voice quality and other inconveniences in order to chat with people around the world virtually free of charge.

Now IT departments are taking a serious look at the Internet and corporate intranets as a means for carrying their organizations' voice and fax traffic at cut-rate prices.

That's because, compared with circuit-switched services, IP (Internet Protocol) networks such as the Internet can carry five to ten times the number of voice calls over the same bandwidth -- with no toll charges for long-distance or international calls. Given these benefits, Kagan Telecom Associates of Marietta, Ga., projects that 15% of all phone calls will be made over the Internet by the year 2000.

With fax traffic, which is less delay-sensitive than voice, the savings from Internet use are even more considerable. Since fax traffic may account for as much as 40% of a Fortune 1000 company's phone bill, the economies can be substantial.

Even so, the greatest potential of the Internet, intranets, and other IP networks may lie with applications that merge voice and data traffic, such as Web-enabled call centers.

In one such use, a customer browsing a firm's Web site would reach a sales or service agent simply by clicking on an icon instead of having to log off the Internet to call the agent.

Responding to this market shift, major vendors such as Lucent Technologies, Rockwell International, and Cisco Systems are starting to introduce gateways and other products to support voice and fax over IP networks. At the same time, the smaller companies who pioneered Internet telephony, such as VocalTec, NetSpeak, and Voxware, continue to push its capabilities.

OBSTACLES TO GROWTH

Despite the growing enthusiasm for Internet telephony, though, there are still several obstacles to overcome in deploying and using the technology. For one thing, response delay, voice quality, and call completion rates are well below business standards.

With Internet telephony, any connection may be a poor one,. depending on how it is routed through the Internet and the amount of congestion at the routing nodes. Response delays of 1 or 2 seconds are possible, compared with typical delays of 50 to 70 milliseconds with the regular phone network. However, since the infrastructure of corporate intranets is more controllable, Internet phones working over intranets should come close to matching the quality-of-service performance standards.

Once the caller is linked to another user, the telephony software samples and digitizes the caller's voice from the PC microphone, compresses the signal, and transmits the compressed packets to the other party using TCP/IP (Transmission Control Protocol/Internet Protocol).

When packets are late or lost, the software compensates by averaging over the gaps. This can hurt speech quality and, occasionally, when congestion is severe, may cause a loss of connection.

In addition, most PCs are not equipped for Internet telephony and will need to be upgraded. At a minimum. the PC should use a 25-MHz 486 design and come with microphone, speakers. sound card, 28.8 Kbps modem, and client software.

One company, Aplio, Inc. of San Bruno. Calif., has addressed this issue with a device that replaces the PC and software normally needed to make calls over the Internet. The Aplio/Phone connects to a regular telephone and, with the push of a button, transforms a normal phone call into all Internet call with any other Aplio/Phone user.

IMPROVED PERFORMANCE

Besides the inadequacies of the Internet's infrastructure, there's also the lack of interoperability among today's proprietary products With codecs, for instance, differences in the voice-compression algorithms compound the interoperability problems created by the various software designs.

Help is on the way, though. Last March, a working group of the International Multimedia Teleconferencing Consortium standardized the G.723.1 codec, partly because it can operate in low-bandwidth environments. The codec is also part of the ITU's (International Telecommunication Union's) H.323 specification for audio and video conferencing over unguaranteed networks. Telephony equipment can incorporate other, higher-performance codecs, but the G.723.1 design provides baseline interoperability.

Meanwhile, the Internet Engineering Task Force (IETF) has also developed specifications to improve the performance of telephony and other multimedia applications over the Internet.

RTP (Real-time Transport Protocol), for instance, includes packet sequencing information and time stamps for synchronizing data streams. A companion protocol, RTCP (Real-time Transport Control Protocol) provides feedback on current network conditions, allowing the telephony software to switch compression algorithms in response to degraded connections.

Another important protocol is RSVP (Resource reSerVation Protocol), which dynamically reserves network resources such as bandwidth or buffers to guarantee quality of service for telephony and other multimedia applications on the Internet.