Audio teleconferencing is voice-only communication. It links people in remote locations via ordinary phone lines. Audio systems include telephone conference calls as well as more sophisticated systems that connect multiple locations - from just a few to many hundreds - via a central bridge that ties all the lines together.

Voice quality is of primary importance to user satisfaction with teleconferencing whether it is audio only, audio graphic or video. Uneven or weak voice levels, noise, acoustic feedback, voice clipping, and other impairments can act as barriers to system acceptance. Audio technology, therefore, plays an important role in influencing user attitudes with any form of teleconferencing.

This Chapter examines current technology for audio teleconferencing. It includes station equipment, transmission services, and bridging options. The discussion focuses on major products and services in today's market. First, however, the various approaches to audio teleconferencing will be covered.

There are many system options for audio teleconferencing but the most common forms are shown in the accompanying illustration, Audio Teleconferencing Options.

A dedicated network is one in which locations are permanently wired together via leased telephone lines. These systems tend to be large, averaging 25 sites or more, and the sound quality is generally better than dial-up conferences. They may connect ordinary rooms, or the rooms may be specially designed for teleconferencing. In this case, an organization leases transmission circuits for its private use and forms an established network of fixed locations. Leased channels are generally full-duplex lines that provide a more natural conversation among participants and better voice quality than the public telephone network. They may also be more cost-effective for high-use teleconferencing applications, and may be used for other communications, such as data transmission. Because dedicated networks connect fixed locations, they lose some of the flexibility of public telephone lines in linking any site into a teleconference. This problem, however, can be overcome by tying public lines into the dedicated network when needed.

Dial-up teleconferences use what is known as the public switched network, also called POTS (plain old telephone system). This is the network we use where we pick up the telephone to call a friend or family member. It is the network that connects millions of phones around the world.

As the simplest form of teleconferencing, dial-up, user initiated conference calls are the most frequently used type of teleconferencing. A telephone handset for example, can readily connect parties via three-way calling. Private Branch Exchange (PBX) systems, key telephones, and telepatchers also provide three-way calling with add-on features. Telephone or PBX conference calls, however have limitations. They usually can link only three or six parties, and there is little or no control over audio quality. Some PBX systems have the capability to connect 12 or more lines in multi-station conferencing, but the voice quality: decreases as lines are added.

Figure: Audio Teleconferencing Options

An operator-assisted conference call is another popular option for audio conferencing. The user simply dials "O" and asks for the conference operator who takes the names and phone numbers of the participants as well as the date and time of the call. The operator notifies each participant and, when the time for the meeting arrives, calls each individual one at a time to interconnect them. This system works well for a small number of locations, such as six or eight, but severs difficulties arise when the number increases. It takes more time to get everyone on-line, causing waiting periods for those contacted first; noise levels also increase as more sites are added; and, people need to be at a predetermined telephone number in order to be reached by the operator.

For these and other reasons, several private companies began offering a type of service called "meet-me" teleconferencing through a device called a bridge. The bridge links channels together so the parties at all locations can hear and talk to each other.

In meet-me conferences, each participant calls into the bridge at a conferencing center. If everyone calls promptly, a large number of locations can be interconnected and ready to conference in only a few minutes. Because participants initiate the call, it means they can be at any convenient location that has a phone. This gives people the flexibility to move around, a factor especially beneficial for business and sales people. Meet-me conferences also control noise levels for better audio quality.

With direct dial conferencing, a person can set up his or her own telephone conference through the use of a touch-tone telephone. The AT&T Alliance service is an example of direct dial conferencing.

Using audio teleconferencing for group-to-group communications involves different equipment considerations than communications among individuals. It usually necessitates that some type of amplified telephone equipment with loudspeaker and microphones be used to accommodate a group. The equipment may be portable and set up in an ordinary room, or the room may be designed specifically for teleconferencing with a built-in audio system and acoustical treatment. The next section will discuss these various configurations.

For station equipment, there are a number of options. These are:

• Telephone

• Desktop units (speaker phones) for inter-office conferencing

• Portable audio systems with tabletop microphone(s) for a group of people in an ordinary room environment

• Semi-portable audio systems in consoles or rollabouts with tabletop microphone(s) for a group of people in a quiet room environment with some acoustic conditioning

• Installed audio systems in acoustically treated teleconference rooms

Although an individual can use a standard telephone handset to participate in an audio teleconference, a variety of station equipment provides hands-free communication and sound amplification for groups of two or more people at the same location. This equipment includes simple amplified telephones, portable and semi-portable systems designed for group conferencing, and installed audio systems integrated into a teleconference room.

The earliest equipment on the market was the amplified telephone, or speaker phone - typically a desktop unit with a built-in microphone and speaker that interfaces with a telephone for hands-free operation.

Because of the limitations of speaker phones in terms of voice quality and flexibility, equipment designed particularly for group teleconferencing applications entered the market in the mid-1960s. A typical system has at least one loudspeaker, a group tabletop microphone or multiple close-talking microphones, and a control unit that interfaces with a telephone or wall jack.

Audio teleconferencing station equipment comes in a variety of configurations. As already pointed out, they include: (l) portable systems with carrying cases; (2) semi-portable consoles or cabinets; and, (3) installed systems. Within each of these, there is a control unit with multi-function controls, associated circuitry and the telecommunications interface. Other features include options in the following categories:

Telephone line connection

• 2-wire or 4-wire
• Plug into telephone
• Plug into wall jack (built-in dialer and ringer)
• Acoustically coupled to telephone handset
• Or, interface to other signal processing equipment and telecommunications channel.

Loudspeaker(s)

• Individual receive unit
• Group amplification
        - Built into control unit
        - Portable remote speakers (tabletop or floor)
        - Wall or ceiling mounted

Microphones

• Group tabletop
• Individual desktop
• Lavalier, clip-on, probe
• Suspended, mounted, lectern
• Wired, wireless
• Omnidirectional
• Directional
• Open
• Voice-switched
• Press-to-activate
• PZM, electret, cardioid, condenser

In selecting the best mix of equipment components, one must consider audio quality in relation to user needs, room environment, transmission network, compatibility and other factors in the overall design. System reliability is another important consideration in order to minimize equipment malfunctions and downtime.

The primary suppliers/manufacturers of audio teleconferencing station equipment systems are: Shure Teleconferencing Systems, Inc., AT&T, Teleconferencing Technologies, Inc. (formerly Darome Equipment), NEC, Coherent Communications, and AT Products. The systems offered by these suppliers differ in design features and represent the basic configurations available in today's market.

One of the primary advantages of audio teleconferencing is the availability of voice transmission channels. With millions of telephones in the world, voice circuits reach into virtually every corner of the developed nations and are rapidly expanding in lesser developed countries.

Voice service is a very large business and AT&T dominates the market, even the break-up of the Bell System and increased competition. However, the growing number of transmission companies, changes in services, and the uncertainty about access charges and tariff rates is causing some confusion as customers try to decide which service and provider to use.

With more transmission alternatives, customers must also look at transmission quality and make sure that specific teleconferencing equipment and transmission services are compatible.

The primary options for audio teleconferencing are dial direct calls via the public switched network and purchased bulk capacity through WATS (Wide Area Telephone Service) or private line services. Public switched service can interconnect almost any location in the world for a teleconference, but the half-duplex circuits of about three kHz present limitations in channel capacity and voice quality. (Half-duplex refers to a communications channel over which both transmission and reception are possible but only in one direction at one time. In practice then, this allows for one person to speak while others wait to respond or react.)

For better transmission quality and more natural interaction, some organizations have private, four-wire (full-duplex) networks dedicated to audio teleconferencing. In this case, full-duplex means a communication channel over which both transmission and reception are possible in two directions at the same time. A full-duplex system allows for a more natural conversation where users can respond and react as if they were together in the same room.

However, tariff costs and fixed-location limitations have caused some users to move away from private audio networks. An option that has grown in popularity is to use private line service for both voice and data. This approach takes advantage of leased channels to support multiple communications, including audio teleconferencing as well as other voice and data traffic.

With further expansion of wide-band digital services, there is greater choice of using shared or private circuits that handle voice, data and video.

Developments that have affected voice communications and audio teleconferencing include the following:

1. Commercial bridging services that interconnect multiple locations for audio conferences.
2. The expansion of long distance telephone services by companies other than AT&T, and the appearance of common and resale carriers to provide voice and data communications.
3. The growth in multi-tenant shared services that provide enhanced voice, data, and video services to tenants of office buildings, including audio teleconferencing, electronic and voice mail, facsimile and so forth.
4. The proliferation of fiber optic networks to expand digital telecommunications capacity for voice, data and video and to decrease costs.
5. The wide band digital services for high-volume voice, data and video communications.
6. New transmission equipment for efficient use of bandwidth and lower transmission costs.
7. An increase in satellite transponder capacity and improved access to satellites from urban areas via teleports.

When an audio teleconference involves more than two locations, some method must be used to interconnect, or bridge, the sites so that all parties can communicate with each other. Telephone handsets with three-way calling, telepatchers and PBX systems can perform a bridging function by tying lines together. However, these devices are usually limited to a small number of lines and are not designed to control audio quality in multipoint conferences.

A PBX system, for example, often has conferencing capability that includes add-on conference, attendant conference, meet-me conference or multi-station/ multi-trunk conference. While these features are many times limited to internal stations only, some PBX's do link a combination of internal and external lines (usually 6-12). However, because standard PBX systems do not have control features to adjust for line noise or voice levels, the noise in the telephone network accumulates as lines are added, interfering with audio quality.

PBX systems have progressed rapidly in the past few years, moving to become the central communications hub in the automated office. Today's digital PBXs support voice and data as well as dial-up and private-line networking to interconnect a variety of office and lower-speed data equipment (PBXs have a channel bandwidth of 64 kilobits per second or less).

There are bridges on the market today that are designed specifically for teleconferencing applications. Their primary function is to interconnect multiple locations and (1) provide smooth voice switching and the even distribution of audio signals to all locations; and (2) control for line noise, echo and transmission loss for good audio quality.

Some noise, echo and loss in signal energy are inherent in telecommunications facilities, but a teleconference bridge should control these factors so they do not go beyond acceptable tolerance levels.

("Noise" on telecommunications channels arises from sources like amplifiers, power line induction, crosstalk, signal frequency tones and background noise at end locations. "Echo" refers to the reflections of signal energy that cause it to return to the transmitter as talker echo or to the receiver as listener echo. "Transmission loss" is the decrease in signal energy as it is transmitted along a circuit due to resistance or impedance of the circuit or equipment.)

Major considerations in the operation of teleconference bridges include the following:

Voice switching

• Hard switch
• Voice switch

Access to bridge

• Dial-out
• Dial-in
        - Automatic
        - Operator assisted

Number of ports

Channel compatibility

• Half-duplex, two-wire
• Full-duplex, four-wire

Monitoring and control

• Automatic
• Attendant operator

Capacity to handle audio graphic transmission or interface to other telecommunications channels

Voice switching mechanisms in teleconference bridges help protect against accumulated line noise when linking multiple locations. The switch's operation and responsiveness will determine how well noise is controlled without creating syllable clipping, delay and other problems due to poor switching performance.

Some bridges use a "hard" switch. The switch is activated, or opens the transmission channel, in response to a voice message and remains open until the message is finished. During the message, no other talker can interrupt unless there is a pause long enough to close the switch. If two people begin to talk at the same time, the person with the loudest voice will activate the switch, blocking the other party. The switching between messages usually creates some unevenness (clipping and delay), especially during rapid exchanges.

Other bridges use a "soft" switch or circuitry that allows more than one talker to be heard at the same time. The switch provides a balance between primary and secondary messages. The primary message is given priority and transmitted at full audio energy. The second message is audible but muted so it does not override the main talker. It is easier for people to interrupt a talker and join the conversation. It also does not create the degree of clipping and delay found in hard switches.

The capacity to allow more than one person to talk at the same time is referred to as simulated, or"quasi," full-duplex operation. A bridge with this feature is often called a two-talker or multi-talker bridge.

Access to bridge. A combination of dial-in and dial-out capability is a standard bridge feature. For dial-in (meet-me) conferencing, most bridges provide automatic answering and linking of incoming calls made to the bridge. The degree of automation, however, varies from a totally automated system to those designed for some degree of attendant assistance.

Dial-out functions are normally handled by an attendant operator. Some bridges have an operator console; others are designed to interface to a PBX, key telephone system or centrex system for multipoint conferences.

Number of ports and channel compatibility. Bridges differ in port capacity, or the number of phone circuits they accommodate. Most bridges have from 8 to 24 ports per unit. The port capacity can usually be expanded by adding circuit boards, modules, or using the bridge in tandem with other bridges.

Most bridges are designed for half-duplex to two-wire channels, but most recently, bridge designs have been introduced to accommodate full-duplex channels.

Monitoring and control. For good audio quality, bridges must monitor ant control echo and transmission loss. Most bridges now employ automatic gait control to compensate for losses in signal energy that create low voice levels. The bridge automatically detects the peak level of incoming signals and provides the necessary adjustment so all signals are at an acceptable and equal loudness.

Echo suppression is also handled automatically by bridge electronics that reduce or eliminate circuit echo.

Security is also an important feature for users of audio bridges. Although the government and military are the primary users, some secured audio bridges are beginning to appear for commercial use today. Typically, these bridges are full-duplex, digital devices that can be linked to either analog circuits or T1 (1.544 mbps) circuits.

Many organizations opt for purchasing their own bridges, operating them as an "in-house" or internal service to departments. The decision to purchase and operate an in-house bridge many times is dependent on the amount of audio teleconferencing an organization does and the applications they have for it. Other organizations rely on commercial bridging services for their audio teleconferencing requirements where they "buy" time for specific conferences and applications. These services have gained in popularity and their cost is very competitive.

Commercial bridging services have grown out of the audio bridging marketplace. There are approximately 30 vendors of bridging services today. These services developed as more organizations wanted high-quality bridging to hold audio teleconferences without the overhead of having an in-house bridge.

Commercial bridging provides a wide array of services to users including call initiation options (meet-me, fully automated, operator dialed, customer dialed, dial-in WATS and so forth), operator monitoring, operator recall, tape recording or on-line playback of recorded messages/information, polling capabilities, subconferencing, and customized answering.

Commercial bridging services do not typically use secured bridges but rather secure conferences by using private passcodes or numbers that only conference participants are given. These passcodes or numbers are then cleared with an operator before a caller is admitted to the audio teleconference.

Audio teleconferencing, strictly defined, is voice-only communication that excludes the transmission of any form of visual information. Audio conferences may incorporate slides, videotapes, or print materials that are viewed locally at each site (e.g., mailed in advance). However, the use of telewriters, facsimile and other electronic media to send or receive visuals moves into the area of audio graphics teleconferencing.

The lack of a visual dimension is often perceived as the major disadvantage of audio systems, especially because people are accustomed to face-to-face meetings that include various types of visual information. The inability to see other people or to spontaneously show print or graphics leads some individuals to dislike audio teleconferencing, claiming it is cold and impersonal. In some cases, it causes people to overlook its possible advantages and to reject audio teleconferencing before ever trying it.

The fact that one cannot see people or transmit graphics does indeed have an impact on how a system is used. However, good program or meeting design can make audio effective. The lack of visual presence may even be an advantage in some situations.

Technically speaking, certain considerations should be kept in mind when developing audio teleconferencing applications or uses. As pointed out earlier, voice or audio quality is of primary importance to user satisfaction with teleconferencing, no matter what form it takes. An audio teleconference is only as good as the voice quality that is transmitted. Certainly, this is the first item to consider when developing an application around audio teleconferencing.

Other items to consider for audio teleconferences include the following:

• Number of locations or sites. What type of bridge do you require and what are its limitations in terms of number of sites that can be linked together?

• Number of users per location. What type of station equipment should be used - push-to-talk microphones or open ones? What type of interactive capabilities should exist - spontaneous as in full duplex or limited as in half-duplex?

• Usage. Will usage be frequent or occasional? Many applications with lengthy sessions may warrant an in-house bridge. Occasional usage may dictate using a commercial bridging service.

• Maintenance/service. Maintenance can be a crucial question when considering whether or not to purchase an on-premise bridge. What services cans vendor provide for maintenance as opposed to the range of services provided be a commercial bridging service. The real question comes down to which method best suits your requirements and the demands of your applications.