WWW: Beyond the Basics

11. Real-time Audio and Video

11.5 Applications

In this section we introduce two popular real-time audio and video applications: telephony and videoconferencing.

11.5.1 Telephony

Telephony enables people in different places in the world to talk to each other through computer networks. The primary advantage is that in using the Internet, people do not incur any long-distance telephone charges. However, you may suffer audio transfer delay caused by heavy traffic in computer networks. This delay is typically a half second.

There are many different telephony software products available on the market today, but most of them adapt the same working schemes. When one party starts talking, his or her speech signal is received by a microphone. The electric signal is then digitized through an ADC board on the talker's computer and stored in computer memory in a binary form. The data then are compressed and transmitted through the computer network. Upon arrival at the listening party, the data are decompressed and converted to an analog electric signal by a DAC board. The speaker converts this electric signal to sound.

Today, most telephony products support full-duplexed conversations (i.e., both parties talking and listening at the same time) and half-duplexed conversations (i.e., only one party speak at a time). However, many currently used sound cards are only support half-duplexed.

For more information, visit the Internet Telephone Page and Internet Telephone FAQ .

Telephony Products

In the following, we will study a telephony product - WebPhone to get a better understanding of telephony.

This is the introduction page of WebPhone.


WebPhone 2.0 is the hottest, professional Internet telephony product with integrated voice-mail on the market today. With it's sleek, user-friendly graphical interface, WebPhone allows you to speak to the world through the Internet or other TCP/IP based networks without incurring any long distance charges. WebPhone's full duplex, encrypted, point-to-point communication allows for unbelievable telephone-quality real-time audio that's hard to beat.

Using WebPhone, you can talk to anyone, anywhere without incurring long distance charges.


System requirements to run WebPhone 2.0

WebPhone 2.0 provides the following features:

For more information of WebPhone, visit WebPhone.

Voice on the Net shows many other telephony products.

Because different software products use different digital formats (i.e., sampling rate, quantization level), compression schemes, and transport protocols, the conversation quality is different.

A/V Streaming: Not Quite Ready for Prime Time evaluates eight programs that provide real-time streaming audio and video.

11.5.2 Videoconferencing

Videoconferencing is a telecommunication facility that enables a face to face meeting between groups of people at two or more different locations through both speech and sight. Every party involved can see, hear and speak just as they would at a conventional round the table meeting. Videoconferencing can be used as:

When a video conference starts, the participants in each party may gather in the office or meeting room that equipped with videoconferencing hardware (This may includes a multimedia-equipped computer, a TV camera, etc.). The video and audio information are recorded by TV cameras and microphones. After digitalization and compression, the information is send to other parties through computer networks.

Videoconferencing usually involves more than two parties, thus, multicasting is used for data transmission over the computer networks. There are two technical methods of obtaining a multicast functionality: (1) multiple point-to-point connection and (2) packet network multicast technology (e.g., MBone in IP). When the multiple point-to-point connection is used, The number of participating parties are usually restricted to six or eight. For a big conference (involving more than eight parties), multicast may be the only viable option.

Videoconferencing Product

There are numerous videoconferencing products available. They vary widely in features and cost. We study an example --- Cu-SeeMe to get a better understanding.

Cu-SeeMe is a videoconferencing program developed by White Pine Software and Cornell University. Following gives technology and major features overview of Cu-SeeMe.

Cu_SeeMe Technology Overview

Enhanced CU-SeeMe uses a unique protocol to manage, receive and rebroadcast video and audio data. The protocol was developed specifically for TCP/IP networks and the Internet. It is capable of running over ISDN networks with TCP/IP network support. Person-to-person, group conferencing, and large audience broadcasting over TCP/IP networks are all possible with CU-SeeMe technology - with little or no added cost for making connections.

CU-SeeMe achieves low bandwidth Internet connections through software only algorithms that reduce data transmission and save you money. It does not require expensive hardware compression/decompression (codec) boards. SLIP and PPP modem connections are supported; however it is recommended that you use a 28.8k modem connection or better.

CU-SeeMe is compatible with video codec and audio standards on both Windows and Macintosh systems, providing versatility and compatibility for the future. CU-SeeMe can be used with most video boards that support Video for Windows. Similarly, CU-SeeMe supports Apple's QuickTime to display video for Macintosh computers.

System Requirements to run Cu_SeeMe

General Requirements

PC Requirements
To RECEIVE:
To SEND: Equipped to receive as outlined above with the added requirements:

General Features of Cu_SeeMe

Visit Cu-SeeMe for more information.

For more information on videoconferencing , visit Videoconferencing FAQ. For more videoconferencing products information, visit desktop videoconferencing products survey.


Voice on the Net and desktop videoconferencing products survey list many other videoconferencing products.


[PREV] [NEXT] [UP] [HOME] [VT CS]

<shaohong@csgrad.cs.vt.edu>
Last modified: Sun Dec 8 1996