By International Calling Cards

VoIP And The Web: Standards Make It Work

BY TIMOTHY RILEY

With millions of PCs connected to the Internet and that number growing each day, opportunities to communicate in ways once deemed futuristic have become commonplace. Staying in touch with family members while away on a business trip, talking to remote staff in a weekly meeting, exchanging messages with members of a common interest group, or speaking with a company's support staff from your PC are just a few of the ways we take advantage of computer-based communication.

Enhance Web-Based Apps With VoIP
As the World Wide Web and its business and entertainment applications have evolved, so has the technology for enabling both real-time and non real-time Web-based communication. Text chat and instant messaging (IM) have led the way for interactive communication on the Internet -- but there is still nothing like the power of voice as a means of communication.

Voice over Internet Protocol (VoIP) technology is currently being used to enhance a wide variety of Web-based applications. Although this technology has the capability to replace a standard telephone call, there is much more to voice-enabled Web applications than a better deal on long-distance.

Today's telephone network provides a high-quality service at a continually decreasing cost and is well-suited to private point-to-point conversations. The power of voice-enabled Web applications is in offering services that are integrated with existing Web functions such as community chat, business conferencing, online customer service, live Web events, online greeting cards, and other applications that improve the user experience.

A few examples of the endless applications for Web-based VoIP services are:

Having a live voice conversation with an e-business representative over a dial-up modem connection. Surf the Internet and talk over the same link, bypassing the need for two phone lines.
Joining a multipoint Web-based sales conference from your cellular phone. Being away from your PC shouldn't be a reason to miss an online conference.
Sending a streaming voice message with your e-mail from a LAN-based PC through a corporate firewall. Being on a LAN or dial-up connection should not make a difference to the user experience.
Having call center agents answer customer calls that originate from PCs or telephones and terminate through the PSTN (Public Switched Telephone Network) to the existing phone switch.
What all these solutions have in common is providing flexible and universal access to Web-based voice applications, and allowing users to benefit from Web-based voice services in a way that is most convenient to them. Although reduced toll charges may be a side benefit, it is the convenience and improved communication of reaching people in real-time that is the primary benefit of Web-based voice.

Integrate With Existing Solutions Through Industry Standards
As Web-based voice solutions grow in popularity, large-scale adoption will be accelerated by integrating with -- rather than competing against -- existing communications infrastructure. Leveraging industry standards can reduce deployment time and enable universal access to Internet voice applications from any interface, at any location. Other key benefits include making the user experience as easy and transparent as possible.

Achieving toll-quality voice over low-speed data links is one of the challenges of providing the transparent experience mentioned above. Most VoIP hardware gateways support ITU industry standard codec technologies including G.723, G.729, and G.711. G.711 provides a 64 Kbps audio data stream for applications in a high-bandwidth environment, but consumes too much bandwidth to be useful for a dial-up connection. G.729 and G.723 provide a 8 Kbps and 6.3 Kbps audio data stream respectively for low-bandwidth, toll-quality voice. At this bandwidth, it is possible to present multiple simultaneous voice conversations to an end user over a dial-up 28.8 Kbps Internet connection. By encoding voice conversations at the source using codec technology that is widely available in existing gateways, the user can avoid problems of transcoding between codec formats which is processor-intensive and can limit scalability and increase costs.

Establishing calls through the use of standard signaling protocols also improves the ability to integrate VoIP technology with existing infrastructure. The IETF's SIP (Session Initiation Protocol) and the ITU's H.323 protocols are being used for VoIP call setup and tear down.

SIP is a lightweight control protocol for creating, modifying, and terminating VoIP sessions with one or more participants. It provides signaling over TCP or UDP and has provisions for other services such as authentication and encryption. A SIP call is carried using RTP over UDP for best performance and minimum latency. A major advantage of a standard-based SIP phone client is that it can be built in a very small applet. Typical PC client-based voice applications will require an applet or application that can range in size from 100 kilobytes to several megabytes. A small PC phone client will provide the fastest download time and the best user experience.

Don't Forget The PSTN
In order for an Internet-based voice application to achieve maximum reach, it must also interface to the PSTN. This allows e-businesses and other Internet applications to allow users to originate a call on their PC that will be routed to an existing phone line or call center. Although many VoIP vendors are moving to support SIP, existing infrastructure for bridging VoIP calls from the Internet to the PSTN is composed of gateway hardware that primarily supports H.323. Initially designed for multimedia conferencing applications, H.323 is being deployed for standard VoIP control functions. As vendor support for SIP on the gateways expands, it will become possible to complete a PC-based SIP client call directly to the PSTN through a SIP-enabled gateway, without first converting to H.323. Until this happens, it is important to support both protocols.

Conclusion
Web-based voice applications are gaining rapid acceptance by Internet users. By supporting key industry standards such as SIP and H.323, these applications will integrate with existing solutions for conferencing, customer service, and e-commerce applications. Any business with a Web presence can improve the experience of their users by enabling live voice applications. Having a live agent only one click away can turn your browsers into buyers and your service inquiries into satisfied customers. These are just a few of the competitive advantages that are gained through Internet telephony -- beyond cost savings.

Timothy Riley is senior director of business development for AudioTalk Networks. AudioTalk, privately held and based in Mountain View, Calif., provides high quality Internet voice services to e-business. AudioTalk creates the enabling technology and turnkey services and infrastructure to add Internet voice communications to Web sites without installing additional hardware and software. AudioTalk's e-business customers add Internet voice services to enhance online Web communities, sell products, provide customer assistance and communicate in ways unique to the Internet. AudioTalk Networks was founded in August 1998 by executives and engineers from leading Internet, networking and multimedia application vendors, including Cisco, Creative Labs, FVC.com, Polycom, ShareVision, Bay Networks, Netcom and Oracle. The AudioTalk team brings years of combined experience and expertise in the areas of voice and video over data networks.