Cisco IP Telephony Applications

Cisco has developed software solutions to enhance their IP telephony solutions. IP telephony applications allow Cisco to augment their IP telephony hardware with features and services to provide an even more viable solution.

Cisco Web Attendant

Cisco WebAttendant is Windows Web-based TAPI software that allows the user to receive and dispatch calls from any IP telephone on the network. WebAttendant allows the IP phone to interface directly with the CallManager to direct calls and to monitor the status of lines, much like a traditional receptionist console. WebAttendant offers many of the same features of traditional PBX systems such as hunt groups and multiple attendant consoles.

WebAttendant is included in the basic CallManager package. It can scale to meet the size of almost any IP telephony infrastructure. A single WebAttendant console can monitor up to 26 calls at a time. A single CallManager cluster utilizing WebAttendant can support up 32 hunt groups with 16 members per hunt group. Also, a cluster can support up to 96 WebAttendant consoles, which means support for up to 512 (96 consoles x 26 calls) calls at one time.

Internet Communications Software

Internet Communications Software (ICS) is a suite of five tools for service and application providers to further gain the benefits of IP telephony.

• Network Applications Manager (NAM) A management console that enables utilization of Automatic Call Distribution (ACD), Intelligent Contact Mangement (ICM), CIS, and IP Contact Center (IPCC).

• Automatic Call Distribution Part of NAM, it reroute calls to different customers serviced via the same CO.

• Cisco IP Contact Center Allows call centers using IP telephony to receive regular POTS calls as well as IP telephony calls.

• Intelligent Contact Management Directs and relays customer contact information between resources such as Web, voice, and e-mail.

• Customer Interaction Suite Allows corporations and service providers to interact with their customers on the Internet or network in real-time. Four components: Cisco Media Manager, Cisco Media Blender, Cisco E-Mail Manager, and Cisco Collaboration Server:

IP Telephony Components

The components that must be added to your infrastructure in order to facilitate IP telephony blur the line between the traditional voice infrastructure and your data infrastructure. In this section, we will discuss some of these components and their features. The VoIP portion of the evolving Cisco Architecture for Voice, Video, and Integrated Data (AVVID) is Cisco IP Telephony, or CIPT shown in Figure 1.

Figure 1: Cisco IP Telephony

Cisco CallManager

Cisco CallManager provides a software-based call processing platform that runs on a Cisco Media Convergence Server (MCS). Cisco's CallManager offers a scalable, reliable, and manageable solution for an organization. While it may not be the ultimate choice for IP telephony, it has set a standard of performance for IP telephony call processing. The Cisco CallManager (CCM) takes the place of a PBX and performs several key functions:

• Registering IP telephony devices

• Call processing

• Administering dial plans and route plans

• Managing resources

The CCM provides a central point for call processing, connection services, signaling, and registration for IP telephone handsets, analog and digital gateways, and legacy telephony devices such as PBX systems. Communication with IP telephony devices is enabled by the use of several IP telephony protocols such as Skinny Station Protocol, H.323, Media Gateway Control Protocol (MGCP), and Simplified Message Desk Interface (SMDI). CallManager offers an open programming interface utilizing the Telephony Application Programming Interface (TAPI) and the Java Telephony Application Programming Interface (JTAPI).

Current releases of the CallManager platform allow a single CallManager server to support up to 30,000 IP telephones per cluster and up to 8 servers per cluster. CallManager is currently in release version 3.3 and offers increased reliability and scalability via clustering. Clustering allows multiple CallManager servers to be interconnected, in order to service more IP telephony devices and to provide redundancy. We will provide more detail on clusters later.

Clustering

Clustering, combines two or more CallManager servers into a logical unit known as a group. A group consists of the CallManager servers and their associated devices such as IP telephones, gateways, and logical devices such as the software-based Cisco IP Softphone. All the CallManager servers in the cluster share the same configuration database. Clustering provides enough servers so that if one fails, the other servers within the cluster assume its load without compromising service.

Cisco has outlined four primary roles a server can take on in the cluster:

• Primary CallManager server

• Backup CallManager server

• Database publisher server

• Trivial File Transfer Protocol (TFTP) server

The primary and backup CallManager servers are self-explanatory. The database publisher server maintains and distributes the master-configuration database. A second but equally important task is the storage of call detail records (CDRs), which is a record of the IP telephony call. The TFTP server provides the system image for devices such as IP telephones and gateways.

Call Detail Recording and Data Mining

Even with the lowered costs associated with using IP-based telephony systems, companies still want complete accounting records. This is true for the business user, as in the smaller company that wants to track where calls are being placed. This is a proactive strategy that emphasizes managed growth. Without understanding where calls are being placed, it is impossible to determine where bandwidth might need to be increased. Accounting records are also a major part of business for companies reselling phone services. A good accounting strategy involves CDR as well as bandwidth analysis obtained from the network. Without both, there is no way to determine if bottlenecks are being caused by voice calls or data traffic.

Call Detail Records

Many of the various pieces of the Cisco voice solution include the ability to capture call detail records. The CCM software can track a number of details and report them to a Radius or TACACS+ server. Those details include:

• Calling number

• Called number

• Call start time

• Call end time

• Bandwidth utilized

Cisco CM also supports output of its CDR to either a Microsoft Access database or an Open Database Connectivity (ODBC) database such as SQL. Once the data has been compiled to a database, a front-end interface can be designed to format the data in a useful way.

There are some limitations to consider before implementing a cluster. A cluster cannot cross a WAN link, so all cluster servers must exist on the same LAN. Servers must be interconnected at minimum by a 10 Mbps switched (not shared) connection. This is to ensure the proper QoS is maintained. A cluster is limited to 30,000 IP telephones. A maximum of 100 clusters can be interconnected, allowing support for up to 1,000,000 IP telephones within an organization. Figure 2 shows typical cluster architecture; notice that one of the backup servers has assumed the load of the failed primary server.

Figure 2: CallManager Clustering in Action

Cisco IP Phones

Cisco IP telephones provide the end-user with an interface to the IP telephony architecture. They contain digital signal processors to perform this function. There have been two generations of IP telephones produced by Cisco. Second-generation phones such as the 7940 and 7960 offer an LCD screen for a menu-based feature set, and utilize inline power via a specialized inline-power patch panel or modules for the Catalyst switch line.

There are currently several phones in Cisco's second-generation phone offering:

• Cisco IP Conference Station 7935 Polycom conference technology full duplex conversations with inline power. Supports Dynamic Host Configuration Protocol (DHCP), coordinated dial plans, and auto-configuration of telephony features. Single RJ45 10/100BaseTX interface.

• Cisco IP Phone 7905G (single user) Basic IP Phone. Supports CDP, 801.1q, TFTP, and DHCP. Single RJ45 10BaseT interface. Inline power.

• Cisco IP Phone 7910G and 7910G+SW (public/lobby area) Basic IP Phone. Supports DHCP and CDP. Single RJ45 10BaseT interface 7910G. 7910G+SW model provides 2-port 10/100BaseT switch. External power.

• Cisco IP Phone 7940G Full-featured IP Phone with Lightweight Directory Access Protocol (LDAP) organizational directory support and advanced messaging features. Supports web information such as weather or stock reports. Supports DHCP, 802.1Q, TFTP, and CDP. Provides 2-port 10/100BaseT switch Inline power.

• Cisco IP Phone 7960G "Executive" full-featured IP Phone with LDAP organizational directory support and advanced messaging features. Supports web information such as weather or stock reports. Supports DHCP, 802.1Q, TFTP, and CDP. Provides 2-port 10/100BaseT switch. Interoperates with CallManager systems, H.323, and Session Initiation Protocol (SIP) Inline power.

Cisco Gateways

Gateways connect your IP telephony infrastructure to the PSTN or to legacy PBX systems. These devices allow calls between the VoIP locations and off-net or PSTN locations. Calls made from your office IP phone to a traditional analog phone and vice versa pass through a gateway. Cisco's product line currently includes over 20 different gateway products that support various types of gateway protocols. Currently there are three different types of gateways supported by the Cisco IP telephony solution: SIP, H.323, and MGCP.

Gateways also provide redundancy. When the VoIP network is congested or when the WAN carrying VoIP traffic is down, the gateway can redirect your outgoing call to the PSTN. The caller's gateway converts the digital voice packets into a traditional time-division multiplexed voice stream and transmits the call through the PSTN. The destination gateway converts the incoming TDM voice stream into digital packets for processing by the destination IP phone.

Switches

Cisco Catalyst switches support IP phones and can provide transmission speeds of up to 1Gbps on a LAN. Transmissions from one IP phone are not broadcast to other IP phones on the same switch. Many Cisco switches can supply inline power to the IP phones via the Ethernet cable. This eliminates the need to provide separate power to the IP phones.

Cisco IP SoftPhone

Cisco IP SoftPhone allows users to utilize IP telephony on any networked computer with a microphone and speaker to obtain a fully functional IP telephone. The IP telephony software digitizes the voice signals and sends the voice packets across the IP network. A graphical user interface (GUI) on the Windows computer provides a dial-pad and other functions of a standard IP telephone. IP SoftPhone consumes 20 device units on a CallManager server, as opposed to the one used by a standard IP telephone handset. It must be installed with Microsoft NetMeeting.

Introduction to IP Telephony | Cisco VOIP

IP telephony describes products and solutions to transport voice traffic over IP. You can use IP to create a converged network to transport voice, video, and data communications. There are numerous benefits to this type of infrastructure, including simplified administration, cost savings on telecommunications fees, and unified messaging services.

Voice traffic and data traffic require two completely different solutions. Data traffic is relatively resilient and tolerant of slow WAN links, lost packets, and unsequenced packets. Voice traffic, on the other hand, is not. Voice requires packets to be received in the same order in which they were sent; if a packet is lost, it should remain lost, as retransmitting the packet would only confuse the person on the receiving end of the call.

There are several components that must be added to your infrastructure. These components include, but are not limited to, specialized router interfaces, specialized LAN switch modules and interfaces, IP telephone handsets, Cisco CallManager servers, and Cisco Unity Mail, as well as other unified messaging solutions.

Conferencing and Transcoding, and Other Services

Conferencing allows multiple participants to communicate in a single call. Cisco technologies support two types of conferencing: ad-hoc and meet-me. In ad-hoc conferencing, the originating caller controls the conference, and determines who will be on the call. The participants may even continue the call after the originating caller hangs up.

A meet-me conference allows participants to participate in a conference by calling an assigned number out of a pool of numbers. More participants can continue to join the conference call until the maximum number allowed is reached. DSP resources support both types of conferencing, and the Cisco CallManager uses DSP resources to provide conferencing services, as shown in Figure 1.

Figure 1: Conferencing

In this scenario, an IP phone caller joins another IP phone and an outside or PSTN initiated caller in a three-way conference call. This is an example of an ad-hoc conference. The Catalyst DSP resources are one way a Cisco CallManager is able to provide a conference bridge. A four-way G.711 conference call would utilize four DSP resources, one for each participant to stream into a single call.

Software conferencing is based on G.711, whereas hardware-based solutions support G.711, G.729a, and G.723. The newer Cisco IP phones 7900 Series supports G.711 and G.729a, but the older style supports G.711 and G.723.

Transcoding is the process of converting IP packets of voice streams between a low bit-rate (LBR) CODEC to and from a G.711 CODEC. An LBR is a CODEC such as G.729a or G.723. An example of a need for transcoding is when a user across the IP WAN wants to access a voice mail system which only supports G.711 and CallManager is configured to initiate remote IP calls using a G.729a CODEC, as shown in Figure 9.18. In this scenario, transcoding must be performed to convert the G.729a voice stream to G.711 in order to communicate with the voice mail system.

Figure 2: Transcoding

Figure 2 serves to illustrate that there are several components necessary to make your voice network a dream come true. Being familiar with these components is one of your first steps to achieving your voice goals. There are also several common telephone functions that any system should be able to provide, regardless of whether they are traditional PSTN or VoIP.

Call Transfer

Cisco IP phones support call transfer. By signaling back to the Call Manager (CM), a call can be transferred to the final destination.

Call Forward

Cisco IP telephony supports three types of call forwarding:

• Call Forward All Forwards all calls.

• Call Forward Busy Forwards calls only when the line is in use.

• Call Forward No Answer Forwards calls when the phone is not answered within a certain configurable number of seconds.

Call Park and Call Pickup

The Call Park feature allows a person to receive a call at another telephone for privacy. A Park soft key allows the receiver to place the caller on hold and dial a designated extension number. At another phone, the extension number can be dialed to pick up the call.

The Call Pickup feature is used to answer an incoming call that is ringing at an unattended telephone. Buttons or soft keys may be configured to activate this function.

Music on Hold

Music on Hold plays music when a caller is on hold, and the music stream may be a .WAV file or a fixed external device controlled by the CM.

Interactive Voice Response

IVRs are useful for routing calls to the appropriate person or department and are less expensive than having an individual do it. In most typical situations, there is an IVR on any public incoming line. Support on Cisco products for IVR is achieved with Tool Command Language (TCL) scripts and voice files, which are referenced in the configuration. When a call comes into the router and matches a set of criteria, the script is queried. The script runs and, depending on the digits it captures, plays an audio file. This audio file is stored in the router's flash and loaded into memory. The audio files use the standard .AU format. The scripts also have the capability to reroute calls.

Voice over HDLC | Non-IP Alternatives to Traditional Telephony

If your particular network situation involves a point-to-point T-1 circuit, you might want to consider voice over High-level Data Link Control (VoHDLC). HDLC is a Layer 2 protocol typically used for point-to-point circuits. VoHDLC is similar to VoFR in that it allows multiple calls to be placed over the T-1 using compression. In the instance of tie-line replacement using T-1s and Cisco equipment, VoHDLC can be an attractive alternative. One of the major drawbacks to VoHDLC is that it has very limited scalability. It was originally designed for point-to-point connections between Cisco routers (Figure 1).

Figure 1: VoHDLC Configuration

As you can see, you have a fair selection of technologies at your disposal for bypassing or supplementing your PSTN needs. No one technology is the "best"; instead, your requirements determine which you will need. VoIP best leverages the investment that you have likely made in your data network to handle your voice needs.

Voice over ATM | Non-IP Alternatives to Traditional Telephony

Voice over asynchronous transfer mode (VoATM) uses ATM to carry digitized voice packets in fixed, 53-byte cells (5-bytes of header plus 48-bytes of payload). ATM switches are extremely fast, a very high quality of service. ATM offers various classes of service (CoS) such as constant bit rate (CBR) specifically for transporting voice. CBR provides a better quality of service by minimizing time variations in the transmission of voice cells, a phenomenon known as jitter. As with VoFR, VoATM is faster than VoIP since it is a Layer 2 protocol.

Where to Use It

ATM will never match gigabit Ethernet when it comes to LAN interconnections, though it can provide true QoS for voice and video. These inherent QoS capabilities allow ATM to outperform IP solutions in many bandwidth-intensive, time-sensitive applications.

VoATM is best used if you already have ATM networks in place, or to interconnect remote offices. The network in Figure 1 shows ATM supporting end-to-end calls.

Figure 1: An ATM Network Providing End-to-End Calls

Calls can originate in Baltimore, pass through Denver, and end in Los Angeles without having to encode or decode the call multiple times. ATM QoS features ensure the timely arrival of calls without any degradation in quality. Data and voice are placed in queues that are treated differently throughout the ATM network.

Figure 2 illustrates how priority can be given to voice traffic over data traffic. In this example, a CBR is specified for voice traffic, which means that a fixed bit rate is assigned to minimize jitter. The data traffic is relegated to an available bit rate (ABR) queue. The data traffic on the ABR queue does not have a guaranteed bandwidth, but it can be allocated more bandwidth than the voice traffic when bandwidth is available. This type of queuing scheme can satisfy both voice and data traffic within the same network.

Figure 2: QoS Queuing with Voice and Data Traffic

When ATM's QoS measures are used, the tagging information is carried for the life of the cell. As long as the voice call is carried through the ATM network, it can retain its higher QoS tag. ATM is a completely different method of handling data flow.

One characteristic of ATM is that it segments data into fixed 53-byte cells. In a frame-based network such as Ethernet, the frame size can vary. This variance causes the switches to have to either wait for the entire frame (also called store and forward) or start sending the frame out the destination port as soon as it is received. Because ATM cells are always 53 bytes, the ATM switches know when the end of the cell has been received without an end-of-frame identifier or delimiter. This knowledge allows ATM switches to switch cells very rapidly.

ATM places as much data as possible in a cell, and if there is any space left over, it pads the cell, which potentially wastes space. The same voice call that was discussed in the VoFR section using G.729 for compression now takes up 14.13Kbps of bandwidth. Using G.729 results in voice being segmented into 30-byte payloads. This results in 23 bytes of overhead, almost as much as the payload itself (Figure 3). This overhead can consume a lot of bandwidth on slower connections, so it is important to understand the ramifications of using ATM on anything under T-1 speeds.

Figure 1: ATM Cell and Potential Overhead

Continuous Bit Rate

ATM provides a CBR service to handle real-time information such as voice. Configuring an ATM link for CBR minimizes variations in time delay or jitter between successive voice cells, providing a high QoS to the end user.

CBR service provides a specific bit rate for voice traffic. There is an unavoidable delay between the transmission and delivery of a voice packet. Although important to minimize that delay, it is also important to minimize the variations in the time delay between successive voice packets. Ideally, each voice packet must arrive on cue, not late and not early.

Costs associated with ATM are still higher than Frame Relay. Equipment costs have come down quite a bit, and in a campus environment, ATM can be a very attractive solution. However, ATM service provider circuits are typically more expensive than Frame Relay circuits. This could make it difficult to justify ATM as opposed to Frame Relay. Keep in mind the following advantages that VoATM has over VoFR:

• Complete and detailed QoS measures

• End-to-end call routing without multiple encoding and decoding

• ATM popularity is still growing and services will be available everywhere in the near future

• ATM in the backbone as well as the WAN means a homogenous topology throughout

Voice over Frame Relay | Non-IP Alternatives to Traditional Telephony

Voice over Frame Relay (VoFR) uses Frame Relay networks to carry digitized voice packets. Building a private network to carry voice and data might not be financially feasible for many companies. VoFR does not use IP as voice traffic is actually addressed and encapsulated in Frame Relay frames. VoFR is typically used for site-to-site voice communications. As Frame Relay is a Layer 2 protocol, it is actually faster than VoIP as there is one less layer in the mix.

Frame Relay has been around for several years, and its standards are mature, with stable interoperability. VoFR allows voice to be compressed and transferred across a Frame Relay permanent virtual circuit (PVC).

A gateway router can connect directly to the frame circuit through either a multiflex trunk (MFT) interface or through a digital signal unit (DSU) attached to its serial interface. Using the compression algorithm G729A, you can squeeze voice calls down to about 8Kbps (total with overhead would be about 10.8Kbps). With a 768Kbps circuit and G729A encoding, you could theoretically support 70 concurrent calls.

Most installations require that voice traffic share a PVC with data traffic. Frame relay works on the principle of having a committed information rate that is guaranteed and a port speed that you can burst to, which differs by provider. Any frames that stray into the burst area are tagged and could be dropped.

What happens if the data traffic starts pushing the bandwidth into the burst area? Voice traffic will mix with data traffic, and it could be tagged for discard. Imagine how a conversation might sound if every other second was dropped. Quality of service (QoS) issues are critically important when it comes to packetizing voice. Implementing QoS on VoFR is easier than with other VoIP implementations. If configuration of QoS is not done correctly, you could end up with a very expensive project as everyone starts calling long distance just to have a coherent conversation.

Frame Relay Benefits

Frame Relay offers several benefits:

• Costs are relatively low compared with other types of circuits.

• Once a port is purchased, adding PVCs between two ports is simple and usually inexpensive.

• Can be oversubscribed which allows more bandwidth to be mapped to a port than the port would theoretically allow, though this could have a severe impact on voice traffic.

• It is available just about everywhere.

• It can support a wide range of bandwidth, from 16Kbps up.

When Does Frame Relay Make More Sense?

The single most useful aspect of VoFR lies in taking advantage of existing or low-cost circuits to provide tie-line functionality between PBXs. The header for frame relay takes up only a little more bandwidth. A call that has been compressed using G.729A uses about 8Kbps, and with frame relay overhead, that increases to 10.8Kbps.

With a typical tie-line replacement, this level of compression and bandwidth savings provide much more room for data to travel over the same circuit (Figure 1).

Figure 1: Bandwidth Savings with VoFR

One of the most important things to avoid when packetizing voice is a multiple compressing and decompressing occurrence during a call. Every time the call goes through a CODEC, there is a delay as the call is encoded or decoded which degrades quality. It is important to note that the CODEC used at each end must match. Once the router has converted the bits into an analog waveform, it sends the call out of the appropriate phone port. (Figure 2)

Figure 2: Encoding and Decoding a Voice Call

When end-to-end calls are needed across multiple locations, it becomes an opportunity for other Layer 2 solutions such as ATM or point-to-point circuits. VoFR has the edge from an ease-of-implementation and management standpoint as well as for long-term cost savings.

PBX Terminology | Cisco Voice Over IP

The following is a list of terms that you should be comfortable using when working with PBXs.

• T-1 24 voice channels (DS-0s), with total bandwidth of 1.544 Mbps.

• ISDN PRI – T-1 Uses T-1 framing, but uses one DS-0 for upper layer signaling (23 voice channels).

• E-1 European standard. 2.048 Mbps (32 voice channels).

• ISDN PRI – E-1 Uses two channels for signaling and framing, (30 voice channels).

• E&M Analog signaling method. Used for trunk or tie lines between switches (network-to-network), and for connections to voice mail or legacy PBX systems.

• Foreign Exchange Station Link between the switch and an extension. Sometimes used to describe a connection that services an analog device attached to the PBX.

• Foreign Exchange Office Link between the PBX and the central office. It is a analog DS-0 tariffed at a flat-rate.

• Loop start Removing the receiver from the hook closes a circuit and creates a loop, allowing connections.

• Ground start Earth ground is needed to complete the loop and allow connectivity.

• Central office Local telephone company termination point for all numbers in a given area, and commonly connects to PBXs via T-1s.

• Coordinated Dial Plans (CDP) Defines numbers on your network and how callers will reach numbers outside your dial plan (for example, a coordinated dial plan may require a nine to be dialed to reach an external number).

• Call routing Physical act of routing a call through the network, and processing the call. This is static in PBX systems.

• Tip-and-Ring In single pair copper connections, identifies which end supplies the voltage on the wire.

• Direct Inward Dial (DID) Establishes a relationship between the extension and a public number. Assigns a block of numbers to a trunk line from the telephone provider to the PBX, and the PBX administrator can route those numbers to related extensions. Figure 1 illustrates the logical mapping of number 415-555-1706 to extension 51706. Please note that it is quite common to create five-digit extensions in North America that relate to the assigned DID numbers.

Figure 1: Direct Inward Dialing Illustrated

Our discussion so far has focused on traditional means of providing voice telephone service. The information provided thus far will help you to transition to accomplishing the same tasks using IP networks and techniques.

Cellular Networks | Voice Communications

The mobile telephone service that preceded cellular service was known as Improved Mobile Telephone Service (IMTS), which operated in several frequency ranges: 35 to 44 MHz, 152 to 158 MHz, and 454 to 512 MHz. But IMTS suffered from call setup delay, poor transmission, limited frequency reuse, and lack of service areas. IMTS was supplanted by Advanced Mobile Phone Service (AMPS) that operates in the 800- to 900-MHz range. AMPS overcame the limitations of IMTS and set the stage for the explosive growth of cellular service which continues today.

Proposed by AT&T in 1971, AMPS is still the standard for analog cellular networks. It was trialed in 1978, and in the early 1980s cellular systems based on the standard were being installed throughout North America. Although AMPS was not the first system for wireless telephony, the existence of a single standard enabled the United States to dominate analog cellular. Europe suffered from a multiplicity of competing standards such as Nordic Mobile Telephone (NMT) at 450 MHz, NMT and Total Access Communications System (TACS) at 900 MHz, and an assortment of other standards in individual countries.

Analog cellular systems have been a huge success. In just 15 years they have attracted around 50 million subscribers in 60 countries worldwide. Today over two-thirds of these subscribers are on the North American AMPS standard at 800 MHz. Although the AMPS standard was originally defined for networks in North America, it is now widely implemented throughout Europe, Latin America, Australia, and New Zealand, as well as many Asian countries, including China, Hong Kong, Malaysia, and Taiwan.

Over the years, AMPS has amply proven itself in terms of being easy to implement and expand to keep pace with increasing demand for mobile phones. It supports automatic roaming so that mobile phone users can continue to use their phones as they move into an area served by a different network. Analog cellular is delivered over networks which employ large cellular hubs and base stations. Despite its success, this method of transmission has its limitations. Analog signals can be intercepted easily and suffer signal degradation from numerous sources, such as terrain, weather, and traffic volume.

A digital version of AMPS—referred to as D-AMPS—solves many of these problems, while providing increased capacity and a greater range of services. Both AMPS and D-AMPS operate in the 800-MHz band and can coexist with each other. D-AMPS can be implemented with time division multiple access (TDMA) as the underlying technology. TDMA provides 10 to 15 times more channel capacity than AMPS networks and allows the introduction of new feature-rich services such as data communications, voice mail, call waiting, call diversion, voice encryption, and calling line identification. A digital control channel supports such advanced features as a sleep mode, which increases battery life on newer cellular phones by as much as ten times over the current battery capabilities of analog phones. D-AMPS can also be implemented with code division multiple access (CDMA) technology to increase channel capacity by as much as 20 times and provide a comparable range of services and features. Unlike TDMA, which can be overlayed onto existing AMPS networks, CDMA requires an entirely new network infrastructure.

D-AMPS also allows operators to build overlay networks using small micro- and picocells, boosting network capacity still further in high-traffic areas and providing residential and business in-building coverage. Advanced software in the networks' exchanges continuously monitors call quality and makes adjustments, such as handing calls over to different cells or radio channels, when necessary. The network management system provides an early warning to the network operator if the quality of service is deteriorating so that steps can be taken to head off serious problems. Graphical displays of network configuration and performance statistics help ensure maximum service quality for subscribers.

Cellular systems, through their interconnection with the public switched telephone network, allow users to originate or receive communications with more portability, and nearly the same degree of functionality as wired telephones. This is accomplished through a hybrid system that utilizes radio technology for the link between the mobile user and the mobile telephone switching office (MTSO), traditional telephone switching technology for the interconnection between the MTSO and those using the wireline public switched telephone network (PSTN) with whom the user communicates, and computer technology to continually monitor the location of mobile users.

In the early 1980s, cellular network service providers became licensed by the Federal Communications Commission (FCC) to operate based on limited competition in each service area. One provider is usually the local telephone company (also known as the "wireline" provider because of its traditional operation of the wired telephone network), and the other licensee is a competitor to the local telephone company, also known as the "nonwireline" carrier. Because of this limited competition, carriers could feel confident that their investment in developing a network would be rewarded with a significant enough portion of the subscriber base to support continued operations. Without this arrangement, it would have been unlikely that the current network would have evolved in such a rapid manner.

Each carrier, wireline and nonwireline, has been assigned separate radio frequencies under which their license permit them to operate. This allows the competitors to coexist within the same physical operating area without interfering with each other's systems. Cellular telephones are manufactured with the inherent capability to operate on either carrier's network, since they have the capability to transmit and receive on either group of frequencies or channels.

The primary wireless communications link established with the cellular telephone is to the nearest cell site. The cellular carrier's network consists of a number of cell sites, each typically covering a radius of approximately one to ten miles, which are in turn connected to an MTSO either via cable or microwave radio links (Fig. 1.1). The system is engineered so that the cell sites are located in close enough proximity to one another to provide seamless networking capability.

Figure 1.1: A typical cellular system.

The coverage areas for adjacent cells actually overlap in order to allow continuous coverage for a user in motion across the network as well as to allow for some load balancing of network traffic. Three hundred and twelve radio channels are available for use by each carrier for voice communications between telephones and the cell site, and the channels used by one cell can be reused by other nonadjacent cells since the transmitted power levels are relatively low.

The radio frequencies used for cellular communication between the mobile user and the cell site are in the range of 825 to 890 megahertz (MHz). Separate channels are utilized for transmitting and receiving voice communications, and the telephone equipment allows transmit and receive channels to be utilized simultaneously so that the parties communicating with each other experience a full-duplex conversation not unlike that of a conventional wireline telephone.

Additional radio communications between the telephone and the cell site takes place over control channels that exchange data between the telephone and the cellular network as to the active phones operating within a particular service area. These control channels also provide functions critical to the establishment of calls and the management of the voice communications channels. From the moment the telephone is turned on, even when idle, communication periodically takes place between the telephone and the nearest cell site. The phone and the cellular network repeatedly exchange information via control channel protocols as to the location and status of the phone and the relative strength of the radio signal between them. This allows the network to find the optimal cell site through which it should route incoming calls to the cellular telephone, to determine when the network should "hand off" an established connection from one cell site to another in order to maintain a strong radio connection, and to allow the phone and the network to synchronize their dynamic use of the many available communications frequencies.

A mobile unit operating outside its local service area is considered to be "roaming." The user's account is established with a local provider, but other providers will allow visitors to their network to use the service. Billing is through the home service provider. A service provider's coverage area might be statewide or might represent a particular area code. Billing to the user represents all on-air use or airtime, whether for outgoing or incoming calls, plus any long-distance charges. Most carriers offer an arrangement such that basic airtime charges, on a per-minute basis, are the only usage cost for an extended calling area. Calls to locations that might incur toll charges within the carrier's service area if made by conventional phone might not incur those charges for a cellular call, but they would be billed based on a flat airtime basis, usually in the area of $0.20 to $0.45 per minute. Calls made while roaming outside this area would come at a higher per-minute rate and/or with additional per-call surcharges. At this writing, only Nextel provides business customers with wireless communication nationwide in the United States without roaming charges.

Since a cellular telephone is so dependent upon a radio link to establish and maintain communications, most of the factors that affect their operation are related to aspects of radio technology. Some of these factors are outside the control of the end user and are specific to the engineering of the carrier's network. The location of cell sites, proximity of adjacent cells, transmitter power, receiver sensitivity, and antenna location can all have a significant impact on the quality of communications. In many locations, service quality between providers is virtually indistinguishable. It is quite likely that each service provider will have areas in which strengths and weaknesses exist, especially pertaining to signal coverage in any specific location.

Service providers are not always able to place their cell sites and antennas in the locations that their engineers might find to be ideal, but they do continually test and tune their network to attempt to provide the best level of service possible. An additional factor somewhat beyond the user's control is that of network traffic loading. Service can suffer even on the best of networks merely due to the congestion that results when too many users attempt to access the network at once. Newer cellular network technologies enable a greater number of channels to be derived from existing frequencies (i.e., frequency reuse) and permit the creation of smaller cell coverage areas or microcells to increase overall network capacity.

Voice-Data Integration | Voice Communications

As much as cellular telephones are a useful tool for mobile voice communications, they are also becoming indispensable for those requiring portable data communications capabilities. Cellular networks are used for the transmission of fax traffic, electronic mail, remote order entry and inquiries, file transfers, and most data communications applications for which the wired telephone network is used. Remote metering locations for pipelines, electrical substations, and other unattended locations that may be far from the nearest telephone lines rely on cellular equipment to provide a connection.

Cellular phones and the networks through which they communicate were not originally designed for data communications purposes, and, until recently, adapters were the only means by which a conventional data modem could interconnect with the cellular network. Even then, not all cellular phones were capable of connecting to a modem. To provide cellular connectivity, a cellular phone must have an outlet into which the user can insert a cable that connects the phone to a modem, the other end of which is plugged into a portable computer. The phone must also support special signaling features that allow it to communicate with the modem. In a properly designed cellular modem, the modem automatically reconfigures itself for cellular operation, enabling the user to send e-mail and faxes, or access on-line service providers like CompuServe or America Online, or "surf the Internet" with the portable computer.

Fortunately, a great number of telephones now incorporate data communications interface capabilities as part of the cell phone unit. Cellular phones do not present the typical dial tone and electrical characteristics to a modem as does a standard telephone line. In addition, while the process of a network hand-off from one cell site to another can be quite acceptable for carrying voice traffic, it can effectively terminate any data communications session in progress. Adapter units compensate for this when used in conjunction with telephones that are not inherently data capable, and data-ready phones do not require these adapters. The adapter units or special cellular-capable modems require that the remote end of the link between the phone and the MTSO also has a device that can communicate using the same cellular-capable protocol. These devices can be provided by the carrier in a pooled configuration within the MTSO available to all users, or the user can ensure that a proper unit is installed at the remote computer location to which the cellular phone is attempting to communicate.

Modems are available that allow the remote user to utilize them for both cellular and "landlines" (wired phone lines). The popular units for users of current generation laptop computers are the PCMCIA card modems that take far less space than a conventional modem or than early versions of cellular modems and also allow the computer to utilize standard phone lines when they are available. The cellular network is not as capable of carrying high-speed data communications as is the wired network, but speeds of near 9.6 Kbps and 14.4 Kbps are possible for data and fax traffic, respectively. Cellular digital packet data, or CDPD, transmission, which is currently being implemented in a number of areas, promises to provide more reliable data communications via existing cellular networks, and at slightly higher throughput rates (to 19.2 Kbps), although not at speeds equivalent to that of landlines. CDPD is appropriate for most applications that might also use conventional packet networks, such as for routine short duration use by individuals and bursty transaction processing.

Several manufacturers have included a data communications capability in their cellular phones. They not only support e-mail, pages, and fax alerts, but provide access to the Internet as well. AT&T Wireless Services, for example, offers an integrated mobile device that operates as a fully functional cellular phone and Internet appliance. The AT&T PocketNet Phone contains both a cellular circuit-switched modem and a CDPD modem to provide users with fast and convenient access to Internet information and two-way wireless messaging services.

At the heart of the AT&T PocketNet Phone is a specialized Web browser that is specifically tuned to send and retrieve only text-based information on the Internet, not burdensome multimedia and graphics which are bandwidth-intensive. With this approach, the browser optimizes the cellular phone's compact display size, memory footprint and wireless connectivity for information services. Web developers can program the PocketNet Phone for remote, wireless information access to corporate intranets and two-way messaging applications that effectively transform the device into a mobile e-mail terminal. Special "@.com" keys facilitate e-mail communication over the Internet.

Cellular data calls are subject to the same challenges as cellular voice calls; specifically, multipath distortion, signal fading, fluctuating power levels, poor frequency response, and external noise. A variety of factors, such as tall buildings, electronic equipment and street traffic, affect the quality of the connection. Although cellular modems contain advanced, cellular-specific error correction protocols (i.e., MNP-10, MNP-10EC, HST, ETC, EC2, and TX-CEL) to compensate for the external factors that impact cellular transmission, data calls should occur from a stationary position, away from power lines or electrical equipment, to ensure the highest transmission speed.

With more cellular phones supporting data communications, we will see a new breed of cellular phone that provides connectivity to PC desktop and databases via infrared or RS-232 connections.

Dual-Mode, Dual-Band Handsets | Voice Communications

Multimode and multiband refers to a type of wireless system that supports more than one technology for its mode of operation and more than one frequency band. An example of a multimode wireless system is one that supports both American Mobile Phone Standard (AMPS) and Code Division Multiple Access (CDMA) systems for analog and digital communication, respectively. An example of a multiband wireless system is one that supports both 800 MHz and 1900 MHz for cellular and Personal Communications Services (PCS), respectively. Of course, wireless telephone systems can be both multimode and multiband, depending on the standards and frequencies supported.

Multimode and multiband wireless systems allow operators to expand their networks to support new services where they are needed most, expanding to full coverage at a pace that makes economic sense. From the subscriber perspective, multimode and multiband wireless systems allow them to take advantage of new digital services that are initially deployed in large cities, while still being able to communicate in areas served by the older analog technologies.

With its multimode capabilities, the wireless system preferentially selects a digital channel wherever digital service is available. If the subscriber roams out of the cell served by digital technology—from one served by CDMA to one served by AMPS, for example—a handoff occurs transparently. As long as subscribers stay within CDMA cells, they will continue to enjoy the advantages the technology provides, such as better voice quality and soft handoff, which virtually eliminates dropped calls. When subscribers reach a cell that supports only AMPS, voice quality diminishes and the chances for dropped calls increases.

The chipsets used in the handsets permit the wireless phones to switch between modes and frequency bands. Handsets using these chipsets have been available since 1995. When sending data, some of the newer chipsets offer even more flexibility, permitting the subscriber to use the Public Switched Telephone Network (PSTN) as well.

Dual-mode AMPS/N-AMPS handsets

N-AMPS, or narrowband AMPS, is a system-overlay technology that allows enhanced digital-like features, such as Digital Messaging Service, to phones operating in a traditional analog-based AMPS network.

Among the vendors offering dual-mode AMPS/N-AMPS handsets is Nokia, the world's second largest manufacturer of cellular phones. The company's 232N is a N-AMPS version of its Nokia 232. The new phone features a large 16-character display with permanent signal and battery strength indicators, four one-touch dialing keys for instant access to emergency services, voice mail, frequently called numbers, and a user-friendly menu interface. In addition, the 232N is data ready via an optional cable which connects the phone to any compatible PCMCIA modem card, allowing the user to send and receive faxes, data, and e-mail via a cellular network.

To support Digital Messaging Service, an optional enhanced service offered by most N-AMPS cellular networks, the 232N is capable of receiving and storing up to 20 short messages in the same manner as a pager. These messages can take the form of short text messages such as CALL HOME or CALL OFFICE, the calling party's phone number, or a notification that voice mail is waiting. The 232N even simplifies the process of responding to messages by enabling the user to call back a number left in a message or to retrieve voice mail messages with a single keystroke.

In its standard configuration with a 550-mAh NiMH battery, the Nokia 232N weighs 7.6 ounces, provides 1 hour 10 minutes of talk time and 15 hours of standby time. Optional NiCd and NiMH slim and ultra extended battery packs are available which provide up to 2.5 hours of talk time and 32 hours of standby time.

Dual-mode AMPS/TDMA handsets

The inherent compatibility between AMPS and TDMA, coupled with the deployment of dual-mode/dual-band handsets, offers full mobility to subscribers, with seamless handoff between PCS and cellular networks. TDMA systems with IS-136, use the Digital Control Channel (DCCH) for support of new applications and teleservices. This enables operators to offer a new generation of advanced wireless capabilities including:

• Revenue-generating features such as Short Message Service (SMS) and Private Networks
• Fraud protection features such as Voice Privacy, Authentication, and Signaling Message Encryption
• Enhanced subscriber features such as Alphanumeric System Identification (SID), Calling Number Identification Presentation (CNIP), Calling Number Identification Restriction (CNIR), and Message Waiting Indicator (MWI)
• Network features such as Enhanced Registration
• Private Networks, which enable service providers to create virtual private networks that charge special billing rates and/or offer group feature set packages
• Over-the-Air Activation, which allows new subscribers to activate cellular or PCS service with just a phone call to the service provider's customer service center

Several manufacturers now offer dual-mode AMPS/TDMA cellular phones, including Nokia, Lucent Technologies, and Nortel.

Nokia introduced the industry's first AMPS/TDMA handset in March 1996. The Nokia 2160 supports all of the most advanced IS-136 TDMA digital features that are available through the Digital Control Channel (DCCH), including authentication, call forwarding, calling line identification, call waiting, selective call acceptance, short message service, and voice privacy. In addition, the 2160 is analog data ready via an optional cable that enables the user to connect the phone to a compatible PCMCIA modem card for sending and receiving faxes, data, and e-mail anywhere within a cellular network.

Lucent Technologies also offers TDMA-based dual-band handsets that support both the cellular (800-MHz) and PCS (1900-MHz) bands with roaming and feature transparency. The dual-band/dual-mode capability of terminals allows a user to move between the 800-MHz and 1900-MHz bands without call interruption. This means operators can use either frequency band to expand geographically into new areas, develop new customer segments, upgrade an existing service offering, to boost capacity.

Nortel's Companion Microcellular System provides seamless communication between private company locations using Meridian 1 PBXs and the public cellular network. Standard IS-136 dual-mode AMPS/TDMA handsets are used with the Microcellular system, which has the capacity to support up to 1500 cellular phones within an area of 10 million square feet. Cellular capacity depends on the coverage provided by the local cellular operator. The system is also capable of handling data transfer applications such as fax and e-mail.

The Microcellular system reuses the standard 800-MHz public cellular spectrum to provide wireless communications inside a building. Because the cellular spectrum is licensed to regional cellular operators by the Federal Communications Commission (FCC), the cellular channels used by the Microcellular system must be obtained from these operators.

When integrated with a Meridian 1 network of systems, the Companion Microcellular System provides the added benefit of a Multi-Site Networking option. This means users can make and receive calls at different company locations throughout the country that use the Companion Microcellular System. If operating in a campus-type environment, this option allows users to make and receive calls from different buildings.

The base stations contain the radio transceiver and may be placed at various locations within a building, up to 3000 feet away from the Meridian 1 system. All radio channels may be simulcast onto all antennas within the same partition to cover high-density areas economically.

Dual-mode AMPS/CDMA handsets

QualComm has been offering dual-mode AMPS/CDMA handsets since 1995. Its QCP-800 portable cellular phone operates at 800 MHz. Using CDMA technology, the QCP-800 portable phone offers superior voice quality, coverage, and privacy while transmitting at RF power levels of only 1/25th to 1/100th as much as an analog cellular phone. This lower power consumption, and the use of lithium ion battery, ensures longer talk and standby time. Users no longer have to carry extra batteries, or lose calls because their phones are turned off to save battery life.

In March 1997, QualComm added several CDMA digital handsets to its QCP series of portable phones. Among the new additions include the QCP-2700, the first CDMA 1900-MHz PCS/800-MHz analog dual-band phone and the QCP-820 CDMA 800-MHz digital/analog dual-mode phone.

The QCP-2700 is QualComm's first dual-band, dual-mode phone that offers expanded coverage for today's PCS only subscribers. The new phone provides carriers with an opportunity to capture customers seeking the inherent benefits of CDMA digital PCS performance and the ability to roam outside their PCS coverage areas.

Dual-mode E-AMPS/CDMA handsets

In 1996, the RF Devices Division of ALPS Electric Co., Ltd. introduced the URP Series transceiver unit for E-AMPS/CDMA dual-mode cellular handsets. Designed for both CDMA (digital) and E-AMPS (analog) systems, the URP Series is a dual-mode cellular transceiver unit that conforms to the IS-98 standard recommended by the Telecommunication Industrial Association (TIA) in the United States.

While CDMA systems are already in commercial use in the United States, Korea, Hong Kong, and other countries, CDMA also faces many competitive challenges in these and other countries. Many operators plan to stay with E-AMPS systems even as they introduce CDMA. In such cases, the same cellular phone will need to be compatible with both systems. Depending on local market conditions, this will provide operators with the means to make an economical transition from E-AMPS to CDMA or enable operators to exploit both technologies to enlarge market share.

Dual-band GSM handsets

In April 1997, Motorola introduced its International 8800 Cellular Telephone, the first dual-band phone capable of operating on both GSM 900 and GSM 1800 networks in Europe. The 8800 allows GSM 1800 subscribers to roam on either their home or other GSM networks (where roaming agreements are in place), using a single cellular telephone.

The 8800 features fax and data support at up to 9.6 Kbps using one of Motorola's CELLect data cards (available separately). Digital Data Fast (DDF) data compression technology offers even faster communication, with effective data throughput speeds of up to 56 Kbps with the CELLect card.

The Motorola phone includes the Personality interface with user configurable Quick Access, which allows users to access preferred functions with as few as two keystrokes. The large graphics display shows four lines of text and graphic icons.

Dual-band/dual-mode handsets

Ericsson's dual-band/dual-mode systems support communication over both 800-MHz AMPS/D-AMPS and 1900-MHz D-AMPS networks. As such, they offer the following competitive advantages to carriers:

• Identical PCS applications and services are provided to subscribers operating in both bands.
• PCS operators can use the same switch for 800-MHz and 1900-MHz services.
• Seamless interworking between 800-MHz and 1900-MHz networks through dual-band/dual-mode mobile stations.
• Using dual-mode/dual-band phones, subscribers on a D-AMPS 1900 channel can handoff both to/from a D-AMPS channel on 800 MHz as well as to/from an analog AMPS channel.

For PCS operators, dual-band/dual-mode service offers several immediate advantages. Existing 800-MHz infrastructure can be used for 1900-MHz services, providing rapid and cost-effective service availability. Existing radio base and switching infrastructure, as well as trunk networks for 800-MHz cellular networks, can also be used for the 1900-MHz traffic.

Roaming and hand-off between 800-MHz D-AMPS, 800-MHz AMPS, and 1900-MHz D-AMPS networks are supported. This provides numerous advantages to PCS operators at 1900 MHz:

• Full coverage can be offered from day one through cooperation with 800-MHz operators in the same geographical area.
• Extended coverage is available through cooperation with 800-MHz operators, or other 1900-MHz operators, in different geographical areas.
• Existing 800-MHz D-AMPS operators can use the 1900-MHz spectrum to increase capacity and develop new user segments in their 800-MHz networks.

  For example, 800-MHz cells can cater for wide-area coverage and act as umbrellas for 1900-MHz micro- and picocells. The small cells can cover the indoor office environment, shopping malls, airports, and difficult spot coverage. The umbrella cells would cater to the fast-moving users and also users moving between two isolated microcells.

Ericsson's D-AMPS 800/1900 dual-band/dual-mode system architecture consists of four major parts:

• The Switching System controls call processing and subscriber-related functions.
• The Base Station performs radio-related functions.
• The Operation and Support System supports the operation and maintenance activities required in the network.
• The Mobile Station is the end-user device which supports the use of voice and data communications as well as short message services.

Ericsson's intelligent roaming capability automatically chooses the best system for the subscriber to use. The company offers dual-band/dual-mode phones that are offered exclusively by Southwestern Bell and AT&T Wireless Services.

As competing technologies for wireless networks emerged in the early 1990s, it became apparent that subscribers would have to make a choice: the newer digital technologies offered more advanced features, but coverage would be spotty for some years to come; the older analog technologies offered wide coverage, but did not support the advanced features. A compromise was offered in the form of wireless multimode/multiband systems that let subscribers have the best of both worlds.

At the same time, wireless multimode/multiband systems allowed operators to economically grow their networks to support new services where the demand is highest. With multimode/multiband handsets, subscribers can access new digital services as they become available, while retaining the capability to communicate over existing analog networks. The wireless system gives users access to digital channels wherever digital service is available, while providing a transparent handoff when users roam between cells alternately served by various digital and analog technologies. As long as subscribers stay within cells served by advanced digital technologies, they will continue to enjoy the advantages provided by these technologies. When they reach a cell that is supported by analog technology, they will have access only to the features supported by that technology. The intelligent roaming capability of multimode/multiband systems automatically chooses the best system for the subscriber to use at any given time.

There is talk in the industry of developing an integrated phone that can work over all major types of wireless networks. Such a "world phone" would be a frequency agile device that accommodates both GSM in standard frequency bands (900 MHz and 1800 MHz) as well as PCS-1900 in North America, among others. The device could serve more than 25 million GSM subscribers worldwide—a number that may grow as high as 100 million by the year 2000.

Although it is unlikely that there will be only one technical standard in the future, today's dual-mode phones are viewed as the first step in the trend toward increasing integration. Dual-mode wireless may quickly advance to triple mode and more. With rapid advancements in chip technology, multimode phones and multifrequency phones offer the same design costs as today's mainstream wireless phones for the consumer market.

Billing Services | Voice Communications

A number of innovations in billing have been introduced in recent years to help wireless service providers acquire new customers efficiently, meet customers' billing needs, reduce the chances of fraud, manage inventory, and provide unified invoicing to roaming subscribers.

Convergence billing

Cellular calls often must traverse different systems, especially if the mobile user is roaming between systems that are based on different international standards. With many different technologies involved in supporting mobile communications—including wireline and wireless telephony and cable TV—there has come the need for billing software that is capable of tracking and billing cellular calls globally from a single location so that subscribers can receive a unified invoice. This is accomplished with convergence billing systems that use powerful Unix-based engines that reach across differing networks to grab call detail information to which appropriate rates and surcharges can be applied. These convergence billing systems can be linked to third-party credit qualification and inventory control systems via application programming interfaces (APIs) to provide wireless network operators with a fully integrated management solution that can include:

• Enhanced collections capability enables service providers to monitor their highest risk customers and assign them to specific collectors. The system provides a complete audit trail of all collector activities to support reporting and analysis of collection activities.
• Telephone number inventory permits service providers to give a new wireless customer a working phone number in real time. The system can track blocks of phone numbers, assign resellers blocks of phone numbers, and assign a block of numbers to a national account.
• Integration with credit qualification system enables service providers to access third–party credit decision systems for the purpose of screening potential subscribers or existing customers. Credit decisions are made in real time and the information can limit opportunities for subscription fraud.
• Interface with customer acquisition/mobile equipment inventory management systems allows a service provider to reduce the time it takes to acquire customers in retail, direct, or telemarketing channels and better manage existing inventory of mobile equipment, including handsets and SIM cards. These features enable service providers to rely on one full-featured system to decrease the cost of acquiring new customers and increase inventory turns.

Prepaid cellular

Prepaid cellular service protects cellular revenues through advance payment for service, while allowing service providers to penetrate new markets. Anywhere from 30 to 60 percent of cellular applicants do not meet credit requirements. Prepaid cellular monitors all incoming and outgoing cellular airtime and debits the usage against the subscriber's balance that has been paid in advance. This service can be offered to customers with little or poor credit history, as well as to short-term subscribers such as vacationers and convention attendees.

Prepaid cellular service allows providers to grow their businesses by expanding their pools of prospective customers, as well as protects carriers against customers who might extend their network usage beyond their ability to pay. Through these applications, those customers who have credit problems and may not otherwise qualify for service can now gain access to cellular airtime.

Other prepaid services allow carriers to offer short-term promotional and convenience products, such as prepaid student services. Additional services include prepaid advertising fulfillment, "circle of friends" calling, and budgeted field service calling.

Subscribers hear the status of their account when they log on and can make multiple calls during a prepaid connection. This service can be accessed from any phone and can have prompts designated in various languages for worldwide markets.

The ability to implement prepaid cellular service has spawned a whole new industry based on the renting of cell phones at trade shows, conventions, sporting events, and hotels. This allows people who otherwise would not become paying subscribers to be activated and buy airtime. Prepaid subscribers represent net new minutes for the carriers.

Prepaid cellular is not only for small retailers and agents looking for a market niche. BellSouth Cellular Corp. is among the major companies offering a prepaid cellular program in select service areas. Establishing a new account is very easy. A customer purchases a phone, or can use an existing phone that is not currently linked to a cellular contract, and pays a $30 activation fee. As part of promotional offering, a free $10 calling card is given so customers immediately can start using the service.

BellSouth's prepaid cellular targets several markets including moderate users of about one hour a month, as well as spouses of current users, college students, and people of all age groups who want the security, reliability, and convenience that cellular offers. Because this market segment is not frequently targeted by cellular carriers due to low usage rates, BellSouth now has a way of bringing wireless services to this new customer in a cost-efficient and convenient way.

Customers have the option of purchasing a $30 card good for 30 days, a $60 card good for 60 days, or a $100 calling card good for 90 days. The rate is $0.99 per minute, reflecting a blended rate of access and airtime which enables customers to only pay for actual time used, giving them control of their spending. Customers will be able to make long distance calls and roam out of their home market at reasonable rates.

Prior to placing a call, the account balance is announced based on the type of call (local, roaming, long distance), and is debited in real time. After each call ends, the customer is notified of the dollar balance that remains; when the account reaches a 5 minute balance, the customer is notified that his or her balance is low and is reminded to refill the account. The customer also hears a tone at 3-, 2-, and 1-minute intervals indicating that the account is reaching zero.

When the account is depleted, the customer simply refills the account by purchasing a new card. The account can be refilled in two ways: purchasing a new card at a local BellSouth Mobility store or at a participating agent or retailer, or by credit card which becomes an automatic way to refill the account and requires a form to be completed prior to activation.

Free calls offered include 911 Emergency Services and *611 for Prepaid cellular customer service. By calling *611, a customer reaches a service professional to handle inquiries or render assistance in establishing or replenishing an account.

Call accounting

The key to a successful prepaid service is a highly fraud-resistant call accounting mechanism to track prepaid time and to debit available time in real time as calls occur. There are two main alternatives to performing call accounting: the use of proprietary debit cell phones and a switch-based approach.

Debit cell phones can be programmed with a certain amount of dollars or airtime units, and they debit any prepaid amounts in real time as the subscriber speaks. However, there are significant negative ramifications associated with this billing method:

• Debit phones require retailers and dealers to set up and support a separate inventory of special-purpose cellular phones; in many cases, special equipment to program the phones would also be required, along with associated costs, training requirements, and security problems.
• Debit phones are expensive, since production volumes are low and they require special features. This makes it hard to serve the "credit challenged" segment without repeating the problem of requiring high deposits for the phones, and it makes it expensive to get into the prepaid services business.
• There may be significant security issues, since debiting occurs within the cell phone, which is in subscribers' hands and therefore vulnerable to breach. Since the dealers or agents own and operate the debit phones, they are ultimately responsible for any airtime charges incurred.

Although these problems are overcome with switch-based prepaid services, debit phones may be the only solution if the switch-based solution is not available in the service area.

As for switch-based prepaid services, there are two principal types of switches which can be used for prepaid service accounting and switching. Some prepaid service providers have chosen a large switch architecture, where switches typically cost hundreds of thousands of dollars and are based on proprietary hardware. In other cases, a high-speed PC-based switch is used, though the software is typically unique to the prepaid service provider. "Class of service" restrictions on the cell phone number ensure that all calls pass through the switch's debit subsystem. The subscriber's account is then debited in real-time during the call.

Besides the call accounting system, prepaid service providers must also provide a means to replenish subscribers' accounts as they use cellular services and run through money in their account. For debit phone systems, various methods can be used to refresh a debit phone, including a credit card swipe, special keyboard codes to unlock additional time in the phone's chip, paying a dealer to use special equipment to update the debit phone's account, etc. For switch-based providers, monthly access and other usage fees can typically be paid by buying special phone cards. Subscribers then call into the prepaid service provider's switch to refresh their account with the authorization number from the prepaid card.

Personal account codes

Cellular and PCS subscribers not only can get detailed billing, which itemizes all calls, but they can have an associated feature called personal account codes, which enables them to assign client or account codes to the calls made while in the home area. This makes it easier to charge these calls back to clients, projects, or departments as appropriate.

With Cellular One service, for example, a personal account code is added to a call as follows:

1. The subscriber enters the phone number he or she wishes to call.
2. The subscriber presses * plus a 1-, 2-, or 3-digit account code.
3. The subscriber presses SEND to initiate the call.

When personal account coding is used, calls will be itemized by the account code the subscriber assigns. All uncoded calls and all received calls are listed on the invoice after the coded calls.