Technologies : Automatic Call Distribution (ACD), Interactive Voice Response (IVR)

Some of the key technologies used in private telephone systems include digital packet voice telephony, automatic call distribution (ACD), and interactive voice response (IVR).

Digital Packet Voice Telephony
Digital packet voice telephony is a communication system that uses digital data to represent and transfer analog signals. These analog signals can be audio signals (acoustic sounds) or complex modem signals that represent other forms of information.

Modern private telephone systems use digital telephony to connect the handset to the local switching system. The analog signal is converted to digital form in the telephone set. By using digital information to represent analog signals, the digital communication system can integrate digital voice information along with advanced signal processing control messages.

Figure 1 shows the digital communications process that uses packet data to connect telephone sets (packetized voice). This diagram shows that the sending telephone set samples and converts the audio signal into digital form. The telephone set may compress the digital information to increase the system efficiency. As data is created, it is divided into packets and the destination address is added to each packet along with a sequence number. Each packet is then transmitted through a packet switching network where they are reassembled at their destination. The received data is then decompressed in turned back into its original analog form.

Figure 1: Packetized Voice over Data Networks

Automatic Call Distribution (ACD)
ACD is a system that automatically distributes incoming telephone to specific telephone sets or stations calls based on the characteristics of the call. These characteristics can include an incoming phone number or options selected by a caller using an interactive voice response (IVR) system. ACD is the process of management and control of incoming calls so that the calls are distributed evenly to attendant positions. Calls are served in the approximate order of their arrival and are routed to service positions as positions become available for handling calls.

ACD systems allow for incoming calls to be distributed specific customer service representative (CSR) or among a group of people. The call distribution is based on previously stored programs, or algorithms that determine the routing of the incoming call. Application of ACD is found primarily in customer service, catalogue sales, and other customer relation areas. ACD systems can forward calls to CSR representatives that are located in other areas or even who are operating on other systems. This allows companies to locate CSRs anywhere in the country or possibly in other countries.

Figure 2 shows a sample ACD system that uses IVR system to determine call routing. When an incoming is initially received, the ACD system coordinates with the IVR system to determine the customer’s selection. The ACD system then looks into the databases to retrieve the customers’ account or other relevant information and transfer the call through the PBX to a qualified CSR. This diagram also shows that the ACD system may also transfer customer or related product information to the

Figure 2: Automatic Call Distribution (ACD)

Interactive Voice Response (IVR)
IVR is a process of automatically interacting with a caller through providing audio prompts to request information and store responses from the caller. The responses can be in the form of touch-tone(tm) key presses or voice responses. Voice responses are converted to digital information by voice recognition signal processing. IVR systems are commonly used for automatic call distribution or service activation or changes. IVR systems use pre-stored voice prompts and a structured menu system that is layered under each option. Layering allows callers to navigate to specific information areas.

Figure 3 shows a sample IVR system that is used to route an incoming call. When this call is received by the PBX, an initial voice prompt informs the user of the system along with initial menu options. The user selects and option. This results in the playing of another prompt indicating new menu options. The user enters the data for the option and the IVR system retrieves data and creates a new verbal response.

Figure 3: Interactive Voice Response (IVR)

To avoid some of the customer dissatisfaction and to handle miscellaneous customer needs, there are often options available in each layer that allows the caller to switch other main menus or to switch to a live “operator.” IVR systems may also regularly provide feedback to the caller of the timing of queuing delays.

Market Growth : Private Branch Exchange Market

Market Growth
The market for key systems has primarily been replaced by PBX systems and the market for PBX systems is decreasing as they are being replaced by computer telephony systems.

Private Branch Exchange Market

The PBX market has been experiencing a decrease in annual sales since the mid 1990s. In 2000, sales of PBX systems declined by 10%. With computer telephony becoming a cost-effective solution for most companies, traditional PBX systems are slowly being phased out. However, there continues to be a growing market for small PBX that are used in small office/home office (SOHO).

Figure 1 shows the trend that has sent PBX manufacturers scurrying to shore up other technological areas. Emphasis has been recently shifted to networks and distributed intelligence via those networks. PBX’s have not become the networking “mother ship” predicted in the 1980’s and early 90’s. The Internet, VPN’s, and ATM functionality are replacing larger PBX systems.

Figure 1: PBX Market Growth

Computer Telephony Market
The computer telephony market is the key growth area in the private networking industry. In the year 2000, 17% of all U.S. businesses with existing PBX systems began a transition to computer telephony systems.

By 2005, CTI systems are expected to penetrate into 80% of all United States businesses. Computer telephony systems are becoming popular because CTI systems only cost $300-$500 per seat compared to PBX systems that cost $800 or more per seat.

Sales of computer telephony equipment in 2000 was $138 million dollars, up from less than $10 million in 1998. CTI equipment sales are expected to exceed to $3.2 billion dollars by 2005. Figure 2 shows a growth of CTI market worldwide.

Figure 2: Computer Telephony Market Growth

Private Telephone Networks : Switching Systems, Numbering Plan

Switching Systems
Private telephone switching systems are network devices that are small versions of telephone switching systems. Early key telephone switches used mechanical levers (crossbars) to interconnect lines. These were called key service units (KSUs). PBX systems use a time slot interchange (TSI) memory matrix to dynamically connect different communications paths through software control. Computer telephony and LAN telephony systems use packet switching systems to interconnect one of more telephone station with each other.

For large private telephone systems, some of the switching functions may be distributed to remote points. An example of distributed switching is the Nortel RPE that allows the Meridian PBX to remote a portion of its station interface to a remote site via a pair T1’s or E1’s.

Numbering Plan
Each extension in a private telephone system has a unique extension number. The station numbering plan for private telephone systems is controlled by the owner of the private telephone system. Many private systems have a limited range for extension numbers (e.g., 1000 -1999. This extension range is restricted due to hardware configurations.

When private telephone systems are interconnected to the public telephone network, the CCITT world numbering plan (E.164) and national numbering plans are used. PBX call processing systems are able to filter numbers to enable least cost routing (LCR). LCR is a telephone system feature that routes the connection of a call over the least expensive route available at the time the call is originated.

To allow automatic routing of incoming calls, direct inward dialing (DID), or higher-level trunk lines (e.g., T1 or E1) with advanced signaling may be used. DID connections are 2-wire trunk-side (network side) EO connections that provide additional information to the PBX to allow the automatic routing of calls within the PBX system. Although network signaling on incoming 2-wire circuits is primarily limited to one-way, incoming service, DID connections employ different supervision and address pulsing signals than dial lines. Typically, DID connections use a form of loop supervision called reverse battery, which is common for one-way trunk-side connections. Until recently, most DID trunks were equipped with either Dial Pulse (DP) or dual tone multifrequency (DTMF) address pulsing. While many carriers would have preferred to use multifrequency (MF) address pulsing, a number of LEC’s prohibited the use of MF on DID trunks.

Private Telephone Networks : Telephone Stations, Local Wiring

Telephone Stations
Telephone stations are telephone instruments that are connected to a telephone network for the purpose of telephony. When these telephone stations are used as part of the private network, they are often identified by the type of system they are connected with. For example, telephone stations that are connected to key systems are called key telephones.

Telephone stations can vary from simple POTS telephones (sometimes called 2500 telephones) to complex Internet telephones (IP Telephones). Telephone stations usually receive their power from the telephone line (loop current) but may receive it from an external source (such as the PBX switching system).

Figure 1 shows a typical telephone station that is used in a PBX system. This diagram shows the difference between standard analog telephone stations and more advanced PBX stations. This diagram shows that analog telephones receive their power directly from the telephone line and digital PBX telephones require a control section that gets its power from the PBX system. Analog telephones also use in-band signaling to sense commands (e.g., ring signals) and to send commands (e.g., send dialed digits). Digital telephones use out-of-band signaling on separate communication lines to transfer their control information (e.g., calling number identification).

Figure 1: Analog and Digital Telephone Stations

Local Wiring
The local wiring for private telephone systems includes the lines between the telephone stations and switching assemblies or interconnection equipment. Local wiring systems may include interconnection points or wiring rooms. Incoming trunks enter the building in a main distribution frame (MDF). Trunks between multiple floors or other buildings are called intermediate distribution frames (IDFs). These IDF areas are often referred to as “wiring closets” because in the past telecommunications wiring was seldom considered when designing a building and the space available for terminating cables was usually in utility closets.

Private telephone system wiring is usually connected through unshielded twisted pair (UTP) wire. There are exceptions where shielded cable, coaxial cable, or fiber is used. Shielded cable is used in areas the may cause or be sensitive to electromagnetic interference (EMI) such as hospital radiology areas or near high voltage equipment in manufacturing plants. Coaxial cable and fiber is used for high-speed data inter-connection trunks or for LAN backbone systems that are part of the private telephone system.

Figure 2 shows the typical wiring systems that are used for private telephone systems. Key systems required many pairs (12 or 25 pairs typical) of lines for each telephone. These telephones were wired to a punch block (splice point) near the telephones. The punch block was connected to the key service unit (KSU). Analog PBX systems usually required 4 wire connections from extension telephones to the PBX switching unit. Two wires were used for audio and two (or more) wires for status or special feature lines. Digital PBX systems connect to each extension using 4 wires. Two wires are used for audio (analog or digital) and two are used for external power. This diagram shows that some digital PBX systems may use multiplexed digital lines to allow multiple phones to share a single line (e.g., a line between building). When this occurs, it is called a remote peripheral equipment (RPE). The RPE separates (demultiplexes) the digital line from the PBX to multiple digital stations.

Figure 2: PBX Local Wiring

Private Telephone Networks : Overview

Private Telephone Networks
Private telephone networks are communication systems that are owned, leased or operated by the companies that use these systems. These telephone systems include key systems, private branch exchange (PBX) systems, computer telephony, and local area network (LAN) telephones. Private telephone systems that are owned or operated by a company or private individual are called customer premises equipment (CPE).

Private telephone systems primarily allow the interconnection of multiple telephones within the private network with each other and provide for the sharing of telephone lines from a public telephone network. Private telephone systems can vary from simple multi-line telephones (key systems) to integrated voice and data service LAN telephone networks.

The first private telephone systems were key telephone systems. These systems used multi-line telephones to provide access to outside lines and used intercom features to allow inter-station connections.

As switching technology improved, small switching systems were installed to provide private branch exchange (PBX) systems. PBX systems provide switching between incoming trunk lines (multiple channel lines) and provide for advanced inter-system calling features.

To offer similar services as PBX systems, central exchange (Centrex) services were developed for end office (EO) switches. Centrex software allows local telephone companies to provide similar features as private telephone systems. These features include 3 or 4 digit abbreviated dialing, automated attendant (call transfer), least cost routing (LCR), and other local switching functions.

CTI systems integrate computer networks and telephony systems. CTI allows PBX technology to provide for voice mail, interactive voice response (IVR), and automatic call distribution (ACD) functions.

The combining of LAN systems with telephone systems is called LAN telephony. LAN telephony allows the sharing of equipment data network cost with telephone system cost.

Figure 1 shows the different types of private telephone systems. This diagram shows the first telephone systems were multiple line key telephone systems. This changed to private branch exchange (PBX) systems. Computer telephony (CT) systems are communication networks that merge computer intelligence with telecommunications devices and technologies. Local access network (LAN) telephony (sometimes called TeLANophy) use LAN systems to transport voice communications.

Figure 1: Private Telephone Systems


Private telephone systems are composed primarily of telephones (called “stations”), local wiring, and switching systems. Telephone stations are the interface between the user and the telephone network. Wiring connects telephone stations to switching systems or distribution points. Local wiring in private systems varies from shared lines (key systems) to individual lines (digital stations). Switching systems interconnect stations to each other or to outside telephone lines or interoffice trunks.

Companies purchase or lease private telephone system and have one or more of their personnel trained to handle day-to-day administrative functions of the system. Practically all PBX’s and key systems today are computer-based and thus allow for soft changes to be made through an administration terminal or PC. Unless the business has a need for technical telecommunications personnel on staff for other reasons, the business will normally contract with their vendor for routine adds, moves, and changes of telephone equipment.

PBX systems are often equipped with key assemblies and systems including voice mail, call accounting, a local maintenance terminal, and a dial-in modem. The voice mail system is controlled by the PBX only receiving calls when the PBX software determines a message can be left or retrieved. The call accounting system receives system message details on all call activities that occur within the PBX. The local terminal provides onsite access to the PBX for maintenance activities. The dial-in capability also provides access to the PBX for maintenance activities.

Future Enhancements: Packetized Voice, High-Speed Multimedia Services, Fiber Distribution Networks, Soft Switches

The future enhancements to public switched telephone networks include the conversion from circuit switched systems to packet networks, expanded fiber networks, multimedia services, and soft switching systems.

Packetized Voice
Packetized voice is the process of converting audio signals into digital packet format, transferring these packets through a packet network, reassembling these packets into their original data form, and then recreating the audio signals.

By the end of 2001, over 5% of international calls from the United States were over the Internet and more than 9.5% of all inter-exchange telecommunications calls were on managed packet switching networks [7]. Packetized voice transmission allows for key features such as dynamic bandwidth allocation and advanced services. To convert to packetized voice, the EO exchange is either replaced or supplemented by a packet switch.

Various protocols such as resource reservation protocol (RSVP) and real time protocol (RTP) have been developed to ensure the quality of service of voice packets that are transmitted through a packet network.

High-Speed Multimedia Services
High-speed multimedia services is the term used to describe the delivery of different types of information such as voice, data or video. Communication systems may separately or simultaneously transfer multimedia information. High-speed multimedia usually refers to image based media such as pictures, animation, or video clips. High-speed multimedia usually requires peak data transfer rates of 1 Mbps or more.

The providing (provisioning) of multimedia services requires communication lines that can have multiple channels and each of these channels may have different quality of service (QoS) levels. As a result, many emerging multimedia services are likely to use ATM.

Fiber Distribution Networks
Fiber distribution networks use optical fiber to distribute communication channels from the PSTN to end customers. There are three key distribution networks: fiber to the neighborhood (FTTN), fiber to the curb (FTTC), and fiber to the home (FTTH).

Figure 1 shows that public telephone networks have growth options. Initially, they are likely to install (FTTN) and use existing copper lines to reach the home. As demand grows for high-speed data communication services, additional fiber may be installed from the node to the curb (FTTC) to replace copper lines. Eventually, to achieve extremely high data rates to the home or business, FTTH or fiber to the basement (FTTB) may be installed.

Figure 1: Fiber Optic Networks

Soft Switches

Soft switches are interconnection switching systems that can dynamically change its connection data rates and protocols types by software control to provide for voice, data, and video services. Soft switches were developed to replace existing EO switches that have limited interconnection capabilities. Soft switches are packet based and can simulate multiple protocols such as Internet protocol and ATM. This allows for multiple types and simultaneous services to each customer with varying levels of QoS.

Services : Switched Data Service, Leased Lines and Digital Subscriber Line (DSL)

Switched Data Service

Switched data transmission services is the providing of data transmission service between points that are setup by or for the customer. A specific form of switched data transmission service offered in public telephone networks is called switched multi-megabit data service (SMDS). SMDS provides high-speed data connections in metropolitan areas that are often used for LAN-to-LAN connections where there are several interconnection points (ports). Interfaces to the SMDS networks normally range from T-1 (1.544 Mbps) to T-3 (44.736 Mbps) although 56 Kbps and fractional T-1 are offered in some locations.

Figure 1 shows an example of the cost structure of SMDS data transmission services. This diagram shows that the user pays an installation fee, a port fee for each access port to the data transmission network, and a monthly usage fee based on the data transmission rate used by the customer.

Figure 1: Cost of SMDS Data Transmission Service

Leased Lines
Leased lines are telecommunications circuits (either two-wire or four-wire) rented/leased from a telephone company to connect two or more locations on a permanent basis. Leased lines are normally associated with data services or voice PBX tie line services. Leased lines are ordered as either analog or digital circuits. Analog circuits provide a single full duplex (two-way) path between locations. They terminate in either telephone switches/instruments or in modems. Digital leased lines, on the other hand, terminate in customer service units (CSU’s) rather than modems. The cost of leased lines depends on the region of service, specific carrier pricing plan, and on distance (line length). As a result, leased lines often connect the end user to another carrier that interconnects another leased line to allow connection to its destination. As a result, leased line prices are often quoted from the customer’s location to an EO or POP of a carrier.

Figure 2 shows the typical costs involved in pricing of point-to-point leased lines in the United States. This table shows that average leased line costs for 56 kbps lines is approximately $240 per month. For a T1 line, the average cost is approximately $900 per month and the monthly cost for a DS3 (45 Mbps) connection is approximately $4800.

Figure 2: Typical Cost of Leased Line Service in United States

Digital Subscriber Line (DSL)
Digital subscriber line (DSL) service is a data service that offers varying data transmission rates to customer. DSL service usually connects users directly to an Internet service provider (ISP). DSL service is generally lower in cost than leased line cost. The difference between DSL service and leased line service is that DSL service does not usually guarantee a data transmission rate.

Figure 3 shows an example of cost of DSL service for an ADSL line. This table shows that a customer pays an initial DSL connection fee, purchases or leases a data interface (e.g., router), and pays a monthly subscription of approximately $50 per month.

Figure 3: Cost of Digital Subscriber Line Service

Services : Voice and Centrex

The key services provided in public switched telephone networks include voice (audio bandpass), Centrex, switched data communications service, leased line, and digital subscriber line.

Voice service is the providing of audio communication circuits that can pass analog frequencies below 3.3 kHz. Voice service is commonly called plain old telephone service (POTS). Voice service remains the core of telephone service as in 2000, the amount of voice traffic transferred per month was more than 53,000 terabytes per month [5].

The newer CO switches have enhanced voice services to allow residential customers to have practically all the features normally associated with PBX’s that serve businesses such as: call waiting, distinctive ringing, voice mail (with signaling or stutter dial tone), feature telephones, and incoming WATS. Some of the newer features are packaged (bundled) together so their actual cost is not readily known.

Figure 1 shows the cost of local telephone service in the United States and that the costs are based on a recurring charge with unlimited usage. The customer may also pay additional recurring fees for advanced services. The cost elements are reasonably standard but the costs vary among LEC’s/CLEC’s. At times the variations between LEC’s in geographies are substantial.

Figure 1: Cost of Local Telephone Service in the United States. Source: Federal Communications Commission (FCC)

Outside the United States, the cost structure for local telephone service is often based on actual usage with a per minute rate ranging from 2 to 6 cents per minute.

Centrex is a service offered by a local telephone service provider (primarily to businesses) that allows the customer to have features that are typically associated with a PBX. These features include 3 or 4 digit dialing, intercom features, distinctive line ringing for inside and outside lines, voice mail, call waiting indication, and others.

Centrex services have had many names over the years, but, whatever the name, the purpose of this offering was always the same: an alternative to customer premises PBX’s. Centrex services flourished and still have a place for many large, dispersed entities such as large universities and major medical centers.

One of the major selling points for centrex is the lack of capital expenditure up front. That coupled with the reliability associated with centrex due to its location in the telephone company CO have kept centrex as the primary telephone system in many of the businesses referenced above. PBX’s, however, have cut into what was once a quite lucrative market for the telephone companies and are now the rule rather than the exception for business telephone service. This has come about because of inventive ways of funding the initial capital outlay and the significantly lower operating cost of a PBX versus a comparable centrex offering.

System : Passive Optical Network (PON)

Passive Optical Network (PON)
A passive optical network (PON) combines, routes, and separates optical signals through the use of passive optical filters that separate and combine channels of different optical wavelengths (different colors). The PON distributes and routes signals without the need to convert them to electrical signals for routing through switches.

PON networks are constructed of optical line termination (OLT), optical splitters and optical network units (ONUs). OLTs interface the telephone network to allow multiple channels to be combined to different optical wavelengths for distribution through the PON. Optical splitters are passive devices that redirect optical signals to different locations. ONU’s terminate or sample optical signals so they can be converted to electrical signals in a format suitable for distribution to a customer’s equipment. When used for residential use, a single ONU can server 128 to 500 dwellings. In 2001, most PON’s use ATM cell architecture for their transport between the provider EO or point of presence (POP) and the ONU (in some case even to the user workstation). When ATM protocol is combined with PON system, it is called ATM passive optical network (APON).

Figure 1 shows an APON that locates ONUs near residential and business locations. This network uses ATM protocol to coordinate the PON. ONU interfaces are connected via fiber to an OLT located at the provider’s EO or POP. Each ONU multiplexes user channels (between 12 and 40) into an optical frequency spectrum allocated to that ONU. Up 32 ONU’s can share access to a single PON using the features of dense wave division multiplexing (DWDM). Some newer PON’s use high density wave division multiplexing (HDWDM). Use of HDWDM increases the number of ONU’s per PON from 32 to 64. This diagrams shows that a PON that uses HDWDM can support approximately 2500 residential customers.

Figure 1: Passive Optical Network (PON)

System : Digital Loop Carrier (DLC)

Digital Loop Carrier (DLC)
Digital loop carrier (DLC) is a high efficiency digital transmission system that uses existing distribution cabling systems to transfer digital information between the telephone system (central office) and a telephone or other communication device. There are two types of DLC: universal digital loop carrier (UDLC) and integrated digital loop carrier (IDLC).

The UDLC is a system that consists of RDTs and central office terminals (COTs). Optical systems such as synchronous optical network (SONET) can transfer signals transparently through the COT to the RDT. The RDT provides an interface between the digital transmission line (e.g., DS1) and the customer’s access line. The RDT can dynamically assign time slots from the communication line to customer access lines.

Integrated digital loop carrier (IDLC) is a digital line interface that has been re-engineered to integrate within a switch (usually as card) and shares the internal bus structure of the switch. This function (or card) is called an integrated digital terminal (IDT). Using the IDT, the switch can directly communicate with a remote digital terminal (RDT) that is closer to the end customer using an efficient multi-channel communication line. The RDT provides an interface between the high-speed digital transmission line (e.g., DS1) and the customer’s access line. The RDT can dynamically assign time slots from the communication line to customer access lines. Because customer access lines are not used at the same time, an RDT that interfaces to a DS1 line (24 channels) usually provides service to 96 customer access lines.

The key advantages to DLC carrier systems are the cost effective transmission and the ability to rapidly add, delete, or change customer services without having to dispatch an installation technician. The DLC system offers improved efficiency through the use of existing distribution cabling systems. DLC systems also offer the ability to extend the range of access lines from the central office to the end customer as the RDT effectively operates as a repeater.

An RDT is divided into three major parts: digital transmission facility interface, common system interface, and line interface. The digital transmission interface terminates the high-speed line and coordinates the signaling. The common system interface performs the multiplexing/de-multiplexing, signaling, insertion, and extraction. The line interface contains digital to analog conversions (if the access line is analog) or digital formatting (if the line is digital).

DLC initially allowed 40 analog telephone connections to be extended to the remote neighborhoods using a device called an SLC-40. Later an SLC-96 (known as a “slick 96”) was put into service that allowed 96 voice grade analog circuits to be extended from the CO on just ten (10) pairs thus reclaiming 86 pairs per installation. Still in use the SLC-96 has allowed the LEC’s to conserve much of their installed outside copper infrastructure.

Unfortunately, DLC systems are not transparent to other systems such as DSL systems. Although it is possible to install digital subscriber line network equipment (co-locate) along with RDT equipment, the RDT equipment housings and power supplies were not originally designed to hold additional equipment.

Figure 1 shows the deployment of an integrated digital loop carrier (IDLC) application in a local telephone distribution network. This diagram shows that a switching system has been upgraded to include an IDT and an RDT has been located close to a residential neighborhood. The IDT dynamically connects access lines (actually digital time slots) in the switching system to time slots on the communications line between the IDT and RDT. The RDT can connect to up to 96 residential telephone lines. When a call is to be originated, the RDT connects (locally switches) the residential line to one of the available channels on the DS1 interconnection line. The IDT communicates with the RDT using the GR-303 standard.

Figure 1: Integrated Digital Loop Carrier (IDLC)

System : Digital Subscriber Line (DSL)

Digital Subscriber Line (DSL)
Digital subscriber line is the transmission of digital information, usually on a copper wire pair. Although the transmitted information is in digital form, the transmission medium is usually an analog carrier signal (or the combination of many analog carrier signals) that is modulated by the digital information signal.

A DSL network is composed of several key parts; this includes a local access line provider, DSL access provider, backbone network aggregator, ISP provider, and other media providers. DSL services can be provided by a single service provider or may result from the combination of processes from different service providers. The communication network can be divided into several parts; local access lines (copper), voice communications network (PSTN), high-speed digital subscriber line (DSL), aggregator (interconnection), Internet service provider (ISP) and content provider (media source). These network parts and the service providers who operate them, must interact to provide most DSL services.

The physical parts of a DSL network include a subscriber access device, network access lines and digital subscriber line access module (DSLAM). There are many configuration options for a DSL network. They vary from a simple end-user’s modem bridge that connects a single end-user’s computer to the DSL network to complex multi-channel, asynchronous transfer mode (ATM) systems that connect routers and set-top boxes.

Figure 1 shows the functional parts of DSL network. This diagram shows that end user equipment adapts, or converts analog and digital signals to a high-speed DSL transmission signal via a DSL modem (an ATU-R for an ADSL system). The copper wire carries this complex DSL signal to a DSL modem at that connects to the central office (an ATU-C for an ADSL system) where it is converted back to its analog and digital components. The analog POTS portion of the signal (if any) is routed to the central office switching system. The high-speed digital portion is routed to a digital subscriber line access module (DSLAM). The DSLAM combines (concentrates) the signals from several ATU-Cs and converts and routes the signals to the appropriate service provider network.

Figure 1: DSL Network Diagram

System : Integrated Digital Services Network (ISDN)

Integrated Digital Services Network (ISDN)
A structured all digital telephone network system that was developed to replace (upgrade) existing analog telephone networks. The ISDN network supports for advanced telecommunications services and defined universal standard interfaces that are used in wireless and wired communications systems.

ISDN provides several communication channels to customers via local loop lines through a standardized digital transmission line. ISDN is provided in two interface formats: a basic rate (primarily for consumers) and high-speed rate (primarily for businesses). The basic rate interface (BRI) is 144 kbps and is divided into three digital channels called 2B + D. The primary rate interface (PRI) is 1.54 Mbps and is divided into 23B + D. The digital channels for the BRI are carried over a single, unshielded, twisted pair, copper wire and the PRI is normally carried on (2) twisted pairs of copper wire.

The “B” channels operate at 64kb per second digital synchronous rate and the “D” channel is a control channel. The D channel is used to coordinate (signal) the communication with the telephone network. When used on the BRI line, the D channel is 16kbps and when provided on the PRI channel, the D channel is 64 kbps. Because the amount of telephone system control signaling is relatively small, the D channel can also be used for low speed packet data messaging. The 64 kbps “B” channels can be used for voice and data. On the BRI system, the two B channels can be combined for 128 kbps data connection.

ISDN telephone lines exclusively use digital transmission. This requires a customer to replace their analog telephones with ISDN digital telephone equipment if they upgrade to ISDN service. ISDN service is typically provided using modular plugs. These plugs include a RJ45 interface (8 pin) for data equipment (called a BRI-S/T) and the other physical connection type is a two-wire, RJ11 type standard (called the BRI-U).

The maximum distance for a BRI-S/T line is approximately 3,000 feet and the maximum distance for the BRI-U is 18,000 feet. Beyond these distances, the service provider may install repeaters to provide service. However, repeaters are expensive to install and setup.

The ISDN BRI allows the user to change the use of the B channels whenever desired. For example, an ISDN user may be sending data using the two B channels at 128 Kbps. If a voice call comes in or is initiated, the data transmission is not interrupted; but is automatically reduced to one B channel at 64 Kbps. When the voice call ends, the data transmission returns to 128 Kbps on the two B channels.

Figure 1 provides the different interfaces that are available in the integrated services digital network (ISDN). The two interfaces shown are BRI and PRI. These are all digital interfaces from the PSTN to the end customers network termination. 1 (NT1) equipment. devices that are ISDN compatible can directly connect to the NT1 connection. Devices that require other standards (such as POTS or data modems) require a terminal adapter (TA).

Figure 1: Integrated Digital Services Network (ISDN)

Technologies : Advanced Intelligent Networks (AIN)

Advanced intelligent networks (AIN’s) are telecommunications networks that are capable of providing advanced services through the use of distributed databases that provide additional information to call processing and routing requests.

In the mid 1980’s, Bellcore (now Telcordia) developed a set of software development tools to allow companies to develop advanced services for the telephone network[4]. The advanced intelligent network (AIN) is a combination of the SS7 signaling network, interactive database nodes, and development tools that allow for the processing of signaling messages to provided for advanced telecommunications services.

The AIN system uses a service creation environment (SCE) to created advanced applications. The SCE is a development tool kit that allows the creation of services for an AIN that is used as part of the SS7 network. A service management system (SMS) is the interface between applications and the SS7 telephone network. The SMS is a computer system that administers service between service developers and signal control point databases in the SS7 network. The SMS system supports the development of intelligent database services. The system contains routing instructions and other call processing information.

To enable SCPs to become more interactive, intelligent peripherals (IPs) may be connected to them. IPs are a type of hardware device that can be programmed to perform a intelligent network processing for the SCP database. IPs perform processing services such as interactive voice response (IVR), selected digit capture, feature selection, and account management for prepaid services.

To help reduce the processing requirements of SCP databases in the SS7 network, adjunct processors (APs) may be used. APs provide some of the database processing services to local switching systems (SSPs).

Figure 1 shows the basic structure of the AIN. Companies that want to enable information services use the SMS to interface to SCP databases within the SS7 network. This diagram shows how a prepaid calling card company manages a portion of a SCP node using the SCE tool kit. The SCP is connected to an IP that contains an IVR unit that prompts callers to enter the personal identification number (PIN). The IP then reviews the account and determines available credit remains and informs the SCP of the destination number for call routing.

Figure 1: Advanced Intelligent Network (AIN)

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