Telecom Basics by Lawrence Harte.

Table of Contents
Telecom Basics, Second Edition Preface Chapter 1 Chapter 2 Chapter 3 Chapter 4 Chapter 5 Chapter 6 Chapter 7 Chapter 8 Chapter 9 Chapter 10 Chapter 11 Chapter 12 - Introduction to Telecommunications - Signal Fundamentals - Signal Processing - Transmission Systems - Switching - Signaling - Protocols - Networks - Systems - Voice over Data Networks - Services - Call Processing

Chapter 1: Introduction to Telecommunications
Telecommunication systems and services involve the transfer of some type of information from one point to another. Telecommunication may consist of the transfer of voice, data, video media and telecommunication systems may combine and control this media in many ways. Telecommunications technologies and systems are designed to reliably transfer information between the originating source of the information and the intended destination(s). Certain types of communication systems are designed to effectively transfer specific types of information. Although these systems may be capable of other types of telecommunications services (e.g. voice and data), other types of communication systems may be better choices. For example, mobile telephone system is very good at transferring voice information, but not very capable of handling video. Telecommunications systems are characterized as being made up of end user equipment, access lines, interconnection equipment, and a coordinating (controlling) structure. Examples of end user equipment are telephone handsets, computer terminals, or pagers. These devices communicate with a telecommunication network through access lines and/or access points. Access systems can be interconnected with each other to form large networks. Control systems are responsible for authorizing access to the system, managing network resources, and measuring usage for billing and accounting purposes. Some communication networks that have been traditionally used for non-telecommunication services have evolved to provide voice services. For example, cable television systems now offer high-speed Internet services and it is relatively simple to upgrade these two-way data networks to offer voice services.

Key Telecommunications Services
Telecommunications services can be divided into three key categories; voice, data, and video. Each of these categories has specific characteristics such as maximum transmission delay time, minimum and maximum transmission rates, and acceptable transmission error types and rates. Voice services involves the receiving of audio signals, processing audio signals into various formats (analog and digital), storing and transporting these signals, and converting the signals back into a form that is similar to its original form. The characteristics of voice networks are very small transmission delay (below 100 msec typical), maximum of 64 kbps for each digital voice channel, and reasonable tolerance to errors. Examples of voice services are Telephony, Voice Messaging, Call Processing, and Computer and Telephony Integration, CTI. Data services provide transport of digital information from one point to one or more points. The characteristics of data networks are moderate transmission delays (above 1 sec may be acceptable), minimum of 28 kbps for each dial-up digital customer and 1 Mbps for each broadband customer, and very low tolerance to errors. Examples of data services include switched connections (circuit switched channels / dial-up), dedicated lines (leased lines/circuits), packet switching (e.g. Internet), and multicast and broadcast (one to many) data transfer. Video services transport high information content signals (video) from one point to one or more points. The characteristics of video networks are very long transmission delay (above 15 seconds for digital broadcast acceptable), minimum of 1 Mbps for each digital video channel (3.2 Mbps for DVD), and reasonable tolerance to errors. Examples of video services include television, closed circuit TV (CCTV), video on demand (VOD), videoconferencing, and interactive multimedia.

Basic Communications Systems
A basic communications system consists of end user equipment, network access connections, network interconnection devices (e.g. switches) and a control system that coordinates the network. A carrier or service provider is company that is engaged in transferring electrical signals or messages for hire through one or more telecommunications systems. Customers (users) request and may receive telecommunications services from the telecommunications network. Because customers request and receive services, a customer is sometimes called a service subscriber or end user. A telecommunications service provider offers communications for a fee directly to the public, or to such classes of users as to be effectively available directly to the public, regardless of the facilities used. A network operator is a provider of telecommunication services. A network operator manages the network equipment parts of a communications system to allow authorized customers to transfer and/or process information via the network. The network operator may provide services directly to end customers or may only manage network equipment and another company (service provider) may manage the provision of services to customers. Figure 1.1 shows a basic telecommunications system. This diagram shows various types of enduser equipment that allow customers to access one or more communication systems via an access network. The access network links the customer (usually by copper wire, coax or fiber) to a communication network. Communication networks connect end-users to other end-users or information services. Different types of networks may be interconnected to each other. Each network must have some form of intelligence that controls the network.

Figure 1.1: Basic Telecommunication System End-user equipment converts various types of information from a user (such as audio or computer data) into a signal that can be transferred via a communications system. Since the late 1800’s, different types of systems had very specific types of end user devices. For example, public telephone systems have a telephone, data communication systems have a channel service unit (CSU), and wireless systems have a mobile telephone. As technology has evolved, end user equipment devices began to combine functionality. This can be found in voice telephone systems that can transfer digital data by using a modulator/demodulator (modem). Access connections are the link between the end user equipment and the wide area network, WAN, owned by the service provider. Access connections can be provided via pairs of copper wires, radio links, or fiber connections. Twisted pairs of copper wires can carry low frequency audio signals such as voice and high-speed digital signals (e.g., 11 Mbps DSL). Radio access can carry low speed information signals (such as wide area cellular) or can be high-speed data transmission (such as microwave directional signals). Each strand of fiberopitic cable (and there may be several hundred fibers per cable) can carry more than 1 Terabit per second of data (1,000,000 million bits per second). Interconnection systems connect of all the various types of equipment. Interconnection systems may include signal taps, splitters, bridges, gateways, switches, and routers to move the information from one part of the network to another along its path between originating and destination points. The interconnection can be completely dumb such as the form of signal taps and splitters that only direct part of the signal energy to multiple points. Some interconnection devices such as bridges and gateways adapt the format of the information to another form (e.g., different packet length or type of packet) between dissimilar networks. Active devices such as switches and routers can direct signals from one source to various other paths depending on call setup information (e.g., telephone number) or an address contained in a data packet (such as a signal router that transfers Internet packets of data). System control and coordination functions ensure that the various resources of the network are coordinated in their actions by detecting equipment and network status. Commands are issued to direct the various network elements in order to configure the network parts and to maintain a high level of network service. Network operators can centrally coordinate system control or multiple network operators can independently and dynamically control it. An example of a centrally managed control system is the signaling system number 7 (SS7) packet control system that coordinates the public telephone network. The SS7 network contains packet switching points and databases that are controlled by the public telephone network operators. Distributed network control is demonstrated by how the Internet is dynamically managed. The Internet is composed of thousands of independent networks that use intelligent routing devices within each network to forward packets throughout the Internet.

End User Equipment
End user equipment, (often called “terminals”) are the interface between the customer and the network. Terminals may translate electrical or optical signals to forms understandable by people or may be translation devices for other electronic equipment (such as computers).

Most telephone equipment converts electrical analog (audio signals) or digital (digitized voice) into acoustic energy that the customer can hear. The basic function of analog telephone service is called plain old telephone service (POTS). The standard telephone (also known as a 2500 series phone,) continuously monitors the voltage on the telephone line to determine if an incoming ring signal (high voltage tone) is present. When the ring signal is received, the telephone alerts the user through an audio tone (on the ringer). Figure 1.2 shows a block diagram of a standard POTS telephone (also known as a 2500 series phone). This telephone continuously monitors the voltage on the telephone line to determine if an incoming ring signal (high voltage tone) is present. When the ring signal is received, the telephone alerts the user through an audio tone (on the ringer). After the customer has picked up the phone, the hook switch is connected. This reduces the line connection resistance (through the hybrid) and this results in a drop in line voltage (typically from 48 VDC to a few volts). This change in voltage is sensed by the telephone switching system and the call is connected. When the customer hangs up the phone, the hook switch is opened increasing the resistance to the line connection. This results in an increase in the line voltage. The increased line voltage is then sensed by the telephone switching system and the call is disconnected.

Figure 1.2: POTS Telephone Block Diagram

Network Access Lines
Copper telephone lines are a primary method of connecting customers to the telephone systems. Copper wire pairs typically use twisted pairs of copper. The twisting of wires reduces the effects of electrical noise from distorting the desired audio signals. In essence, when the noise is received on one twist, the same noise is received on the other twist. The voltage goes positive on one line while it also goes positive on the other. Basically, the two noise signals are at the same level and they cancel each other (balance). Coax lines use one wire (a shield) to surround the other wire to help contain electrical energy from leaking out. Telephone lines usually start from the central office’s switching center in the form of bundles of many wire pairs (trunks). These trunks connect the central switching office to distribution cables (cables with a reduced number of wire pairs) that eventually are connected to individual homes or businesses. Trunks may contain thousands of pairs of wires while local distribution cables only contain 25 to 100.

Cables are produced in rolls with a limited length (often 500 feet long). The installation of telephone cables requires several splices points as the large trunk cables connect to the distribution cables that connect to the drop cables to the home. Figure 1.3 shows a typical layout of a telephone wiring between the central office and the home. This diagram shows that 600 pairs of copper line start from the central office (CO) switch. The switch wires are connected to the local loop lines in a main distribution frame (MDF). This trunk cable is connected to three 200 pair distribution cables that supply circuits to nearby neighborhoods. As the cables enter into a neighborhood, they are connected at splice points to smaller distribution cables until it is connected to a final distribution cable that only holds 25 pairs reaches a telephone pole located near a house. At the telephone pole, usually 2 pairs of wires are tapped to the drop line that enters into the house (to allow up to 2 separate phone lines). These 2 pairs of wire are attached to a network interface device (NID) that protects (isolates) the wiring in the home from the telephone network wiring. Once in the home, twisted pairs of wires are looped from the NID to telephone jacks within the house. This illustration also shows that there is significant potential for different types and sizes of wire and many splice points. This inconsistency can dramatically affect the ability to transfer high-speed digital signals.

Figure 1.3: Access Telephone Line Wiring

Interconnection Networks
Local access networks often connect to other networks (such as the PSTN, PTT or Internet) via switching systems or gateway connection. Various types of gateway connections can connect the local telephone switching system (often called the “End Office” or “Central Office”) to other public (e.g. Internet) or private (e.g. corporate) networks. A gateway transforms data that is received from one network into a format that can be used by a different network. It usually has more intelligence (processing function) than a bridge as it can adjust the protocols and timing between two dissimilar computer systems or data networks. A gateway can also be a router when its key function is to switch data between network points. Interconnections to other public telephone networks are classified by the type of connection. Basically, the lower the connection type number, the more simple (and more limited) the connection is. The connection types include the basic customer type POTS (type 1) and interswitch types 2. Type 1 POTS connection provides for basic signaling and low speed (audio) connection. The higher types of connection include various capabilities such as types of information services available (operator assist, emergency number support). In the United States, the typical interconnection types include those designated as type 2A, 2B and other variants of type 2, each serving a specific purpose. Type 2 interconnections link the LEC into a tandem (standard local switch interconnect) office. When using the type 2 connections, the CO appears as a standard end office switching facility. Networks commonly increase their data transmission capacity or quality levels as you move towards the top of the network (away from the endpoints). For cable networks, this is called

feeders and for telephone networks, these are called high-speed backbone interconnects. Highspeed interconnection lines between switches and tie lines are called trunks. Figure 1.4 shows some of the different types of interconnection networks. These vary from distribution networks (no switching functions such as cable television), centrally controlled networks (such as the public telephone system), and packet switching networks (such as the Internet).

Figure 1.4: Interconnection Networks

Network Control
Network control is the transmission of signaling messages that perform call-control functions such as supervision, call setup routing, provisioning (authorizing) of services, and call processing control. Networks are either common to all users or privately leased by a customer for some specific application. The term “network” also refers to a group of two or more broadcast stations or cable systems interconnected physically and organizationally so as to broadcast the same program schedule simultaneously without any switching functions. In the early telephone systems, network control routing of a telephone connection was manually monitored and processed by human operators. Human operators would supervise the call by listening for request tones (ringing sounds) and manually coordinate the connection by talking to end customers (who originate calls) and other operators (for cross-connections). When the call setup process had been agreed (all the switching points established), the connection was made through physical connections (patch panels).

To provide for more efficient network control, telephone control signals (tones) were created to allow the transfer or call control information on the same audio lines as the voice signals for call setup. These control tones would either be mixed with the audio or temporarily replace the audio signals. This type of audio signal control is called in-band signaling. As the design of telephone networks advanced, it was necessary to add more intelligence to the call setup (e.g., automatic forwarding of telephone calls), it became necessary to shift the control signaling to circuits outside the audio path. This allowed more rapid call setup and better overall control over the communications connection. When the control signals are separated from the actual communication channel, these are called out-of-band signaling. Provisioning of a network is a process within a company that allows for establishment of new accounts, activation and termination of features within these accounts, and coordinating and dispatching the resources necessary to fill those service orders. Provisioning involves customer care and billing systems. Figure 1.5 shows how different types of networks can be controlled. This diagram shows that a network can have no control (distribution only), can use intelligent databases to control dumb switches, or it can use intelligent switches to route information through a dumb network.

Figure 1.5: Network Control

Telephone and Device Numbering
Each device within a network must have its own unique address. Some of the different types of addresses that are available include telephone numbers and data network addresses.

International Numbering Plan (ITU)
The International Telecommunications Union (ITU), a division of the United Nations, has defined a world numbering plan recommendation, “E.164.” The E.164 numbering plan defines the use of a country code (CC), national destination code (NDC), and subscriber number (SN) for telephone numbering. The CC consists of one, two or three digits. The first digit identifies the world zone. The number of digits used for telephone numbers throughout the world varies. However, no portion of a telephone number can exceed 15 digits. There are several “E” series of ITU

numbering recommendations that assist in providing unique identifying numbers for telephone devices around the world.

North American Numbering Plan (NANP)
An 11 digit-dialing plan is used within North America. It contains 5 parts: international code, optional intersystem code (1 +), geographic numbering plan area (NPA), central office code (NXX), and station number (XXXX). The NPA code defines a geographic area for the serving telephone system (such as a city). The NXX defines a particular switch that is located within the telephone system. Finally, the station code identifies a particular line (station) that the switch provides service to. Figure 1.6 shows the telephone numbering systems.

Figure 1.6: Telephone Numbering Systems

Internet and Data Network Numbering
Most data network addresses are hierarchical where the beginning of the address identifies the entire network and each progressive address number (or group of numbers) identifies more specific parts within the network. Data networks are usually composed of several interconnected links. These links can be of different technologies with each of their end points identified by a unique numbering system. Figure 1.7 shows how different types of data network addressing systems can co-exist. This diagram shows a data connection that is composed of several parts. An end-user is connected to an application server through a company Ethernet network. The computers network interface card (NIC) has an address unique to the Ethernet hub. The Internet address is included as part of the data message after the Ethernet address. The company’s network is connected to an ISP by a high-speed frame relay connection. The frame relay access device (FRAD) has a unique identifier to the ISP. The ISP connects the data connection via asynchronous transfer mode (ATM) to the ASP.

Figure 1.7: Internet and Network Numbering Systems

Number Portability
Number portability involves the ability for a telephone number to be transferred between different service providers. This allows customers to change service providers without having to change telephone numbers. Number portability involves three key elements: local number portability, service portability and geographic portability. The first part of the telephone number (NPA-NXX) usually identifies a specific geographic area and specific switch where the customer subscribes to telephone service. If a telephone number is assigned to another system (different NXX) in the same geographic area (same NPA), the interconnecting carriers (IXCs) connecting to that system must know which local system to route the calls based on the selected local service providers. In this case, the IXC must look up the local telephone number in a database (called a database dip) prior to delivering the call to the end customer. Figure 1.8 shows an example of how local number portability (LNP) can be used to redirect telephone numbers as customers move their telephone number to different geographic areas or transfer the phone number to different system operators (e.g. transfer from wired to wireless operators). In this diagram, a caller in Los Angeles is calling to someone in Chicago. The call is routed through the LEC in Los Angeles and routed through a long distance provider who needs to connect the call into a local telephone company in Chicago. Because there are several local providers in Chicago, the IXC must look into a database to see if the number has been ported to a different service provider. This is the next to last switch before the call reaches the end office switch (called “N-1”). This switch uses the dialed digits to find which local carrier is providing service in the Chicago area. When the IXC finds which exchange is serving the number, the call is routed to the correct local switching office and the call is completed.

Figure 1.8: Local Number Portability (LNP) Operation Service portability allows a customer to take their telephone number to a different type of service provider. Service portability involves determination of the type of service provider (e.g., wireless or wired) who is responsible for completing the call using the area code and NXX. The interconnection and call processing for different types of service providers varies. Geographic portability involves the transfer of telephone numbers outside the normal geographic boundaries of the service provider’s area. Geographic portability allows a customer to keep their same area code when they move to new cities or other distant geographic regions.

Chapter 2: Signal Fundamentals
Telecommunication technology involves the transfer of information signals through wires, fiber, or through the air by the means of electrical or optical signals. Communication signals are usually characterized by their intensity (voltage and current) and frequency (cycles per second). To allow information to be transferred using communication signals, an information source (audio, data or video) is either represented by the signal itself (called the baseband signal) or the information slightly changes the wave shape of the communication signal (called the broadband signal). The information is imposed on the carrying signal (called the carrier) by varying the signal level or time changes (frequency shift).

Signal Types
There are two basic types of signals: analog and digital. Many communication systems receive analog signals (e.g., audio signals), convert them to a digital format, transport the digital signals through a network, and reconvert the digital signals back to their analog form when they reach their destination.

An analog signal can vary continuously between a maximum and minimum value and it can assume an infinite number of values between the two extremes. Analog information can be represented by a continuous and smoothly varying amplitude or frequency over a certain range such as voice or music. Analog communication lines allow the representation of information to closely resemble the original information. Figure 2.1 shows a sample analog signal created by sound pressure waves. In this example, as the sound pressure from a person’s voice is detected by a microphone, it is converted to its equivalent electrical signal. Also, the analog audio signal continuously varies in amplitude (height, loudness, or energy) as time progresses.

Figure 2.1: Analog Audio Signal Processing

The frequency of an electrical or optical wave is the number of complete cycles or wavelengths that the wave has in a given unit of time (second). The standard measurement for this is number of cycles per second, also known as Hertz (the scientist, not the car company), abbreviated Hz. Figure 2.2 displays how frequency is measured. In this example, there three cycles of a wave that are transmitted over a 1 second time period. This equals a frequency of 3 Hertz.

Figure 2.2: Frequency

Digital signals have a limited number of discrete states, usually two, that transfer signal levels at predetermined time intervals. The two levels of most digital signals are on (logic 1) and off (logic 0). The information contained in a single time period is called a bit. The number of bits that are transferred in one second is called the data transfer rate or bits per second (bps). Because many bits are typically transferred in 1 second, the data rate is typically preceded by a multiplier k (thousand) or M (million). For example, if the data transfer rate is 3 million bits per second, 3 Mbps would indicate this. Bits are typically combined into groups of 8 bits to form a byte. When the reference is made to bytes instead of bits, the b is capitalized. For example, 10 thousand bytes is represented by kB. Figure 2.3 shows a sample digital signal. In this example, the bits 01011010 are transferred in 1 second. This results in a bit rate of 8 bps.

Figure 2.3: Digital Signal The earliest form of digital radio communication was Morse Code. To send Morse Code, the radio transmitter was simply turned on and off to form dots and dashes. The receiver would sense (detect) the radio carrier to reproduce the dots and dashes. A code book of dots and dashes was used to decode the message into symbols or letters. The on and off pulses or bits that comprise a modern digital signal is sent in a similar way. The trend in communication systems, just as in other types of electronics products such as compact discs, is to change from analog systems to digital systems. Digital systems have a number of important advantages including the fact that digital signals are more immune to noise. Unlike analog systems, even when noise has been introduced, any resulting errors in the digital bit stream can be detected and corrected. Also, digital signals can be easily manipulated or processed in useful ways using modern computer techniques.

Baud Rate
Baud rate is the number of the signaling elements (symbols) per second on a transmission medium. For some communication line transmission systems (called line codes), the baud rate is the same as bit rate. However, in many applications, the baud rate is below the bit rate. For example, in 2B1Q coding, each quaternary (4 level) signaling element conveys 2 bits of information, so the baud rate is one-half the bit rate. The spectral characteristics of a line signal depend on the baud rate, not the bit rate. For high-speed digital communications systems, one state change can be made to represent more than one data bit. Figure 2.4 shows that the baud rate is not always the same as the bit rate as each baud (symbol) can have several states that represent multiple binary bits.

Figure 2.4: Baud Rate

Radio Frequency (RF)
The radio frequency spectrum is divided into frequency bands that are authorized for use in specific geographic regions. Globally, the International Telecommunications Union (ITU) specifies the typical use for radio frequency bands. Within each country, government agencies create and enforce the rules for which specific types of systems and services are used in specific frequency bands and which companies will be able (will be licensed) to own and operate these systems.

RF Channels and Bandwidth
An RF channel is a communication link that use radio signals to transfer information between two or more points. To transfer this information, a radio wave (typically called a radio carrier) is modulated (modified) within an authorized frequency band to carry the information. The modulation of the radio wave forces the radio frequency to shift above and below the reference (center) frequency. Typically, the more the modification of frequency, the more information can be carried on the radio wave. This results in RF channels typically defined by their frequency and bandwidth allocation. Bandwidth allocation is the frequency width of a radio channel in Hertz (high and low limits) that can be modulated to transfer information. The amount type of information being sent determines the amount of bandwidth used and the method of modulation used to impose the information on the radio signal. A government regulation agency (such as the FCC in the United States) defines a total frequency range (upper and lower frequency limits) that a radio service provider can use to transmit information. In some systems (such as AM or FM radio station broadcasting), this is a single radio channel. For other systems (such as cellular, PCS, or PCN), this is a range of frequencies that can be sub divided into smaller radio channels as determined by the radio carrier. When the allocated frequency range is further subdivided into smaller allowable bands, these subdivided areas are referred to as channels.

Signal modulation is the process of modifying the characteristics of a carrier wave signal using an information signal (such as voice or data). The characteristics that can be changed include amplitude modulation (AM), frequency modulation (FM), or phase modulation (PM). A pure electrical, radio, or optical carrier signal carries no information aside from either being in on or off state. When the carrier signal is modified from a normalized state, it is called a modulated signal. This modulated signal is the carrier of the information that is used to modify the carrier signal. When the carrier signal is received, its signal is compared to an unmodulated signal to reverse the process (called demodulation). This allows the extraction of the original information signal. A carrier wave signal can be carried by wire, fiber, or electromagnetic waves transmitted through the air (radio). When a carrier signal is modulated, the frequency changes above and below its reference frequency. The difference between the upper and lower maximum frequency changes is called the bandwidth. The relationship between the amount of frequency bandwidth of an information signal (the baseband) and the channel bandwidth of the modulated carrier determines if the system is a narrowband or wideband system. Narrowband systems have a relatively small communications channel bandwidth, typically below 100 kbps. When the bandwidth of the broadband carrier is much higher than the bandwidth of the information source, it is called a wideband system. The device that modifies the carrier signal with the information source (baseband signal) is called a modulator. An assembly or device that combines the function of modulating and demodulating signals is called a Modulator/DEModulator (MODEM).

Amplitude Modulation (AM)
Amplitude modulation involves the transferring of information onto a carrier signal by varying the amplitude (intensity) of the carrier signal. Figure 2.5 shows an example of an AM modulated radio signal (on bottom) where the high of the radio carrier signal is change by using the signal amplitude or voltage of the audio signal (on top).

Figure 2.5: Amplitude Modulation (AM) Operation

AM was one of the first and simplest forms of modulation. Unfortunately, in addition to changes in intensity caused by an AM modulator, electrical noise that may be on the communication circuit may appear as part of the modulating signal. This can cause distortion in the received signal.

Frequency Modulation (FM)
Frequency modulation involves the transferring of information onto a radio wave by varying the instantaneous frequency of the radio carrier signal. In 1936, the inventor Armstrong demonstrated that a frequency modulated (FM) transmission system was much less susceptible to noise signals than AM modulation systems. Figure 2.6 is a process known as frequency modulation (FM). In this diagram, as the modulation signal (audio wave) increases in voltage, the frequency of the radio carrier signal increases. As the voltage decreases, the frequency of the carrier signal also decreases.

Figure 2.6: Frequency Modulation When frequency modulation is used to transmit digital information, it is called frequency shift keying (FSK). To represent a digital signal, the FSK modulator transmits on one frequency to signify an on signal (usually a logic level 1) and a different frequency to signify an off signal (usually a logic level 0).

Phase Modulation (PM)
Phase modulation is a modulation process where the phase (relative time shift) of the carrier signal is modified by the amplitude of the information (e.g., audio or data) signal. Changes from an input source is reflected by correspondingly varying the phase (or relative timing) of the carrier wave signal. Figure 2.7 shows a sample of phase modulation (PM). In this diagram, a digital signal (on top) creates a phase modulated carrier signal (on bottom). As the digital signal voltage is increased, the frequency of the radio signal changes briefly so the phase (relative timing) of the transmitted signal advances compared to the unmodulated radio carrier signal. This results in a phase-shifted signal (solid line) compared to an unmodulated reference radio signal (dashed lines). When the voltage of the digital signal is decreased, the frequency changes again so the phase of the transmitted signal retards compared to the unmodulated radio carrier signal.

Figure 2.7: Phase Modulation When used to transfer digital information, a phase shift that occurs over a specific amount time represents one or more bits. This is called phase shift keying (PSK) modulation. When the amount of phase shift is 180 degrees (out of a 360 degree cycle), it is called binary phase shift keying (BPSK). Quadrature phase shift keying (QPSK) is when the number of possible amount of phase shifts is four (90 degrees each).

Frequency Shift Keying (FSK)
Frequency shift keying (FSK) is a specific form of frequency modulation in which the modulating signal shifts the output frequency between predetermined values to represent a digital signal. One frequency shift is used to represent a digital one (sometimes called a mark) and the other frequency shift represents a digital zero (sometimes called a space). Figure 2.8 shows a sample of frequency shift keying (FSK). In this diagram, each pulse from the digital signal (on top) creates a change in carrier signal frequency (on bottom). As the digital signal voltage is increased, the frequency of the radio signal changes above the center (unmodulated) carrier frequency. When the voltage of the digital signal is decreased, the frequency changes again so the frequency of the transmitted signal is below the center (unmodulated) carrier frequency.

Figure 2.8: Frequency Shift Keying (FSK) Modulation

Quadrature Amplitude Modulation (QAM)
QAM is a combination of amplitude modulation (changing the amplitude or voltage of a sine wave to convey information) together with phase modulation. There are several ways to build a QAM modulator. In one process, two modulating signals are derived by special pre-processing from the information bit stream. Two replicas of the carrier frequency sine wave are generated; one is a direct replica and the other is delayed by a quarter of a cycle (90 degrees). Each of the two different derived modulating signals are then used to amplitude modulate one of the two replica carrier sinewaves, respectively. The resultant two modulated signals can be added together. The result is a sine wave having a constant unchanging frequency, but having an amplitude and phase that both vary to convey the information. At the detector or decoder, the original information bit stream can be reconstructed. QAM conveys a higher information bit rate (bits per second) than a BPSK or QPSK signal of the same bandwidth, but is also more affected by interference and noise as well. Figure 2.9 shows that amplitude and phase modulation (QAM) can be combined to form an efficient modulation system. In this example, one digital signal changes the phase and another digital signal changes the amplitude. In some commercial systems, a single digital signal is used to change both the phase and the amplitude of the RF signal. This allows a much higher data transfer rate as compared to a single modulation type.

Figure 2.9: Quadrature Amplitude Modulation (QAM)

Combined types of Modulation
Today’s sophisticated modulation systems can combine all three modulation types: amplitude, frequency, or timing (phase) at the same time to transfer analog or digital information. One of the more popular forms of combined efficient modulation technologies is multiple level Quadrature Amplitude Modulation (QAM). Because QAM can send information at many amplitude levels with combined with 4 phase positions for each symbol, this allows some QAM systems transmit up to 64 or even 128 bits per symbol. With many levels being so close to each other, these complex modulation technologies require a very stable signal with minimum interference. Multi-level QAM systems are often used on pointto-point microwave systems because of the high quality of the transmitted signals.

Demodulators are devices or systems that can recover original information signal from a carrier signal. The output from the unit may be in baseband (original information) composite form. Figure 2.10 shows how a demodulator converts a modulated carrier signal into an information signal. This diagram shows that the demodulator compares the modulated carrier (carrier with the

changes) to an unmodulated carrier (pure carrier signal) to produce the information signal (representing only the changes of the carrier signal).

Figure 2.10: Demodulator Operation

In general, a channel is a stream of information transmitted as part of a distinct communication, or conversation, or for a particular purpose or end use. One channel may be distinguished from other channels by the time of occurrence of the transmission, by the format or organization of its content, by the frequency of a carrier signal used to transmit it, by some secondary property such as the type of error detecting code used for it, or by other properties. Due to the advance of technology, a single channel may at some time historically be modified so that it carries multiple channels. For example, when the only distinguishing feature of two radio signals was their carrier frequency and each one carried only one conversation, each one was described as a channel. Different carrier frequencies were designated by distinct channel numbers. Each of these channels is called a physical channel. At a later date, time division multiplexing was used with radio signals to distinguish 3 or 8 distinct conversations on one modulated radio carrier frequency. Confusion arose because the 3 or 8 distinct conversations, each using a designated time slot in time division multiplexing, were described as separate channels. At the same time, the entire signal (comprising 3 or 8 channels) was also described by some as a channel. Each of these channels is called a logical channel. Figure 2.11 shows the difference between physical and logical channels. In this example, a physical channel transports information between two points using electrical signals. The physical channel is divided into frames that contain various fields (groups of information within the frame). This diagram shows that the frames on the physical channel are divided into 4 logical channels; 3 logical channels for data and one logical channel for control. The exact relationship between the frame structure and the logical channels is called mapping. This example shows two different mapping examples. In the first example, the bits in the information portion of each frame are equally divided. In the second mapping example, more bits are proportioned to channel 2. This results in a lower data transfer rate for channels 1 and 3 while channel 2 has a higher data transfer rate.

Figure 2.11: Physical and Logical Channels

Physical Channel
A physical channel is a communication path or system that uses a transmitter, a physical medium, and a receiver to transfer information. Physical channels may be characterized by their transmission medium type (e.g. copper or fiber), bandwidth (lower and upper frequency limits), or data transmission rate. Figure 2.12 shows different types of physical channels. This diagram shows that the physical channel transports information from a transmitter to a receiver. A physical channel may be a copper line, fiber optic line, or a radio channel.

Figure 2.12: Physical Channels

Logical Channel

Logical channels are portions of a physical communications channel that are used to for a particular (logical) communications purpose. Logical channels are created by dividing physical channels into time, frequency or digital coded channels to create separate logical channels. Figure 2.13 shows how a single digital communication channel is divided into several different logical channels. In this example, there are four fields per frame. Field 1 contains system information, field 2 contains the channel status information (busy/idle), field 3 transfers channel assignment commands and field 4 is defined for other sub channels.

Figure 2.13: Logical Channels

Radio Channel
A radio channel is a communications channel that uses radio waves to transfer information. Radio channels are used to transfer information because radio signals can travel much further than the information (baseband) signal and to allow many signals to be transmitted over the same geographic area (different radio frequencies.) Radio channels are often characterized by their bandwidth. Bandwidth is the difference between the highest and lowest carrier frequency that the transmitted signal will occupy. Bandwidth is measured in units of Hertz or cycles per second. Bandwidth defines the portion of the frequency range (frequency spectrum) where the waveform contains all (or substantially all) of the power contained within a signal. The bandwidth can be found in a designated portion of the frequency spectrum. Bandwidth also can refer to the amount of data transmission rate associated with a digital signal. Figure 2.14 shows how a radio carrier wave is modulated by a user’s information signal to produce a radio channel signal. In this example, the user data is multiplexed (time shared) with a control information signal. The combined data signal (user data and control data) is supplied to the radio frequency (RF) modulator along with the radio carrier signal. This diagram shows that the modulation process (changing) of the radio carrier signal causes the frequency of the carrier to change and the energy of the carrier wave to be distributed within a pre-defined frequency band (channel bandwidth). This example shows that some of the modulated signal energy does fall out of the prescribed channel bandwidth. This is typical for most systems. Radio system specifications (and/or government transmission requirements) usually exist to ensure that the levels transmitted outside the prescribed frequency band do not interfere with other systems.

Figure 2.14: Radio Channel Operation

Channel Multiplexing
Channel multiplexing is a process that divides a single transmission path into several parts that can transfer multiple communication (voice and/or data) channels. Multiplexing may be frequency division (dividing into frequency bands), time division (dividing into time slots), code division (dividing into coded data that randomly overlap), or statistical multiplexing (dynamically assigning portions of channels when activity exists). When several communications channels are connected over a common channel, a device called a multiplexer is used. The multiplexer combines multiple incoming (input) signals onto one common communications channel through the process of time, frequency, or code sharing. At the other end of the communication line, a demultiplexer device is used to separate the channels (output) at the receiving end. When a digital channel is divided into multiple digital sub channels, the separate channels are called logical channels. Each logical channel is assigned a portion of the bits from the digital communications channel. A device that converts the information signal into a format that is suitable for transmission is called a transmitter. The device that receives and decodes the transmitted signal is called a receiver. When a transmitter and receiver are combined into one device, it is called a transceiver.

Frequency Division Multiplexing (FDM)
Frequency division multiplexing is a process of allowing multiple channels to share a frequency band by dividing it up into smaller frequency bandwidth channels. Each of these smaller channels provides for a separate communications channel. Figure 2.15 shows how a frequency band can be divided into several communication channels. When a device is communicating on a FDM system using a frequency carrier signal, it’s carrier channel is completely occupied by the transmission of the device. For some FDM systems, after it has stopped transmitting, other transceivers may be assigned to that carrier channel frequency. When this process of assigning channels is organized, it is called frequency division multiple access (FDMA). Transceivers in an FDM system typically have the ability to tune to several different carrier channel frequencies.

Figure 2.15: Frequency Division Multiplexing Carrier signals can co-exist with each other on an FDM system without interference if they are operating at different frequencies. Because the modulating signal slightly changes the carrier signal, this produces small changes in frequency. This results in a single radio signal that occupies a frequency range, depending on the type and amount of information that is changing the electromagnetic wave. The maximum amount of frequency change is typically called the channel bandwidth. Hence, a carrier signal should not typically operate in areas that other radio carrier signals may occupy.

As a carrier signal is modulated (amplitude, frequency, or phase), several other small energy signals at different frequencies are created. Some of the signals produced by the modulation process fall outside the designated frequency bandwidth. Although the amount of energy that falls outside the designated bandwidth is usually small, they may cause interference with other devices that are communicating on other nearby channels. To help protect from unwanted interference, when multiple carrier signals are operating in an FDM system, a guard band is usually used to protect adjacent carriers from interference. Guard bands are a portion of a resource (frequency or time) that is dedicated to the protection of a communication channel from interference due to radio signal energy or time overlap of signals. While guard bands protect a desired communication channel from interference, the guard band also uses part of the valuable resource (frequency bandwidth or time period) for this protection.

Time Division Multiplexing (TDM)
Time division multiplexing (TDM) is a process of sharing a single carrier channel by dividing the channel into time slots that are shared between simultaneous users of the carrier channel. When a transceiver communicates on a TDM system, it is assigned a specific time position on the carrier channel. By allowing several users to use different time positions (time slots) on a single carrier channel, TDM systems increase their ability to serve multiple users with a limited number of channels by dividing a frequency band into time slots. Time slots are grouped into repetitive frames. Each communication channel is assigned to one (or several) time slot(s) within a frame. To allow TDM systems to provide continuous voice communication to a transceiver that can only transmit for brief periods, TDM systems use digital signal processing to characterize and compress digital signals into short time-slices. Figure 2.16 shows how a single carrier channel is time-sliced into three communication channels. Transceiver number 1 is communicating on time slot number 1 and mobile radio number 2 is communicating on time slot number 3. Each frame on this communication system has three time slots.

Figure 2.16: Time Division Multiplexing (TDM) Digital multiplexing is a specific form of time division multiplexing that is used to combine multiple digital signals onto a common higher speed digital channel. Figure 2.17 shows how digital multiplexing combines two or more low speed channels into one higher speed communication channel. In this diagram, there are eight 8 kbps communication channels that are supplied to a multiplexer. The multiplexer stores and sends 8 bits of each slow speed communication channel during each 125 usec time slot on the 64 kbps channel.

Figure 2.17: Digital Channel Multiplexing

Code Division Multiplexing (CDM)
Code division multiplexing uses a method of spreading an information signal using different codes on a wide bandwidth communication channel (typically digital signals). For CDM channels, the frequency bandwidth of the carrier channel is much larger than the bandwidth of the original information signal. Because the channel bandwidth is very large, information from other channels operating in the same frequency band is relatively small. This allows multiple communications channels to operate in the same frequency bandwidth at the same time. There are various forms of CDM. The most popular forms of spread spectrum include frequency hopping and direct spread spectrum. Frequency hopping is a multiplexing technology where transceivers may share a frequency band by transmitting for brief periods of time on an individual carrier channels and then hopping to other carrier channels to continue transmission. Each transceiver is assigned to a particular hopping pattern and collisions that occur are random. These errors only cause a loss of small amounts of data that may be fixed through error detection and correction methods. Direct spread spectrum is relatively new commercialized (verses militarized) modulation technique that is used primarily in cellular and satellite systems. Direct sequence spread spectrum systems mix a relatively long digital code with a small amount of communication data (information signal) to produce a combined signal that is spread over a relatively wide frequency band. To receive the signal, the long code is used to extract the original signal. Because the energy is spread over a wide bandwidth, multiple spread spectrum channels with different codes can co-exist with minimal interference. Figure 2.18 shows how a single direct sequence spread spectrum communication channel can have several channels. In this example, there are 3 different code patterns that are used for communication channels. When a receiver uses the reference code, a direct sequence spread spectrum system uses a mask pattern to allow only that information which falls within the code mask to be transmitted or received.

Figure 2.18: Code Division Multiplexing

Digital Speech Interpolation (DSI)
In addition to multiplexing through channel division, statistical multiplexing can also be used by distributing transmission of a communications channel over idle portions of multiplexed channels. An example of statistical multiplexing is digital speech interpolation (DSI). DSI is a technique that dynamically allocates time slots for voice or data transmission to a user only when the have voice or data activity. This increases the system capacity as transmission for other users can occur when others are silent. Digital speech interpolation (DSI) is a digital form of a process known as time assigned speech interpolation (TASI). The DSI technique dynamically allocates channels (usually time slots) for voice or data transmission to a user only when they have voice or data activity. This increases the system capacity as transmission for other users can occur when others are silent. A system that has DSI capability assigns information transmission based in speech activity. The DSI system senses activities of speech signals and availability of communication channels in a system and dynamically transmits information signals on available communications channels. Because speech conversation is composed of pauses and alternating directions of communications (usually one person speaks at a time), the use of TASI increases the efficiency of a communications system by approximately 2:1. For example, a 96 channel communications circuit that uses TASI can provide service to approximately 192 calls. Figure 2.19 shows the process of multiplexing using DSI. This diagram shows a communication circuit that has 96 independent communication channels (one communication link that has 96 time slots). The DSI system monitors the activity of each voice conversation (a voice channel) using a voice activity detector (VAD). The VAD is an electronic circuit that senses the activity (or absence) of voice signals. This is used to inhibit a transmission signal during periods of voice inactivity.

Figure 2.19: Digital Speech Interpolation (DSI) When the VAD detects that speech is active, the DSI system assigns the information to specific time slots on the communications channel. The DSI transmitter system identifies the voice channel at the beginning of the transmission so the DSI receiver can assign it to an output voice channel. When the voice activity detector senses a pause in communication, the DSI transmitter sends an ending message on the channel allowing the channel to be placed back into a pool of available communication channels. The next time the speech activity detector senses that the voice channel is again active, the DSI transmitter will select a channel from the pool of available communication channels and the process begins again. Each time, the DSI receiver will assign the information to the correct output voice channel.

Line Coding
Line coding is the process of modulating and formatting data for transmission on a communications line. The selection of a line coding type determines how signals will be

synchronized (external or from the data itself,) and how the transmitted signal will react to the transmission line during transmission. The transferring of digital information on a communication line may require a precise timing synchronization signal to determine the start period of groups of data (e.g. logical channels) and the transition period for each bit of data (bit interval.) The use of incorrect timing signals could result in errors. Some line coding systems force changes in data transmission to allow the receiver to extract a timing signal directly from the received signal. Transmission lines contain physical characteristics (such as capacitance) that can alter the transmitted signal. For a long transmission line that has capacitance, the transmission of continuous positive or negative pulses would result in the creation (buildup) of a DC voltage. To reduce the effect of DC bias, some line coding systems alternate between positive and negative levels so the positive pulse represents a logical 1 (on) and a negative level represents a logical 0 (off). There are many types of line coding systems. Some of the more common line coding systems include return to zero (RZ), non-return to zero (NRZ), alternate mark inversion (AMI), and Manchester coding.

Return To Zero (RZ)
Return to zero line coding uses a transmitted data format in which the logic level for a data 1 is a 1 during the time the data clock is high, but returns to 0 during the time the data clock is low. For a data 0, the logic level is 0 for both high and low states of the data clock. Figure 2.20 shows that return to zero (RZ) encoding uses the logical level voltage during half of the bit period of each logical bit. In this example, the data is transmitted at 1 kbps so each logical bit period is 1 msec. During this entire period, the transmitted logical level remains at the same voltage associated with the logical level for 1/2 a bit period (0.5 msec).

Figure 2.20: Return to Zero (RZ) Operation

No Return To Zero (NRZ)
No (non) return to zero (NRZ) is a digital line coding system that transmits a low for a 0 bit and high for a 1 bit and does not return to 0 between successive 1 bits. Figure 2.21 shows that no-return to zero (NRZ) encoding uses the logical level voltage during the entire period of each logical bit. In this example, the data is transmitted at 1 kbps so each logical bit period is 1 msec. During this entire period, the transmitted logical level remains at the same voltage associated with the input logical level.

Figure 2.21: No Return to Zero (NRZ) Operation

Alternate Mark Inversion (AMI)
Alternate mark inversion (AMI) is a line coding method used in many digital wire transmission systems (such as T-1). AMI in T-1 systems uses return-to-zero (RZ) in an alternating bipolar pulse string. The AMI signal uses a logical binary zero corresponding to zero volts and logical binary one corresponding to alternating serial positive or negative 3 volt pulses. This process allows self-synchronization when a limited number of zeros is transmitted. There are therefore limitations (called zeros density limitations) on the number of consecutive zero volt pulses in such systems (a limit of 15 consecutive zero volt pulses is the maximum allowed for DS-1 systems, for example). Figure 2.22 shows how alternate mark inversion (AMI) uses an alternating bipolar scheme to represent digital information. The AMI system represents information by alternating between a positive (e.g +3 Volts) and negative (e.g. -3 Volts) levels. This AMI process allows for selfsynchronization by forcing during each bit sequence transmission.

Figure 2.22: Alternate Mark Inversion (AMI) Operation

Manchester Encoding
Manchester encoding is a digital coding technique that divides each bit period into half periods. A negative to positive transition represents a binary I, and a positive to negative transition represents a binary 0. The use of Manchester encoding allows for clock recovery as transitions occur on every bit transmission. Figure 2.23 shows how Manchester encoding transfers digital information in the form of positive or negative transitions. This example shows that a logical 1 is indicated by a positive transition and a logical 0 is indicated by a negative transition. This example also shows that Manchester coding forces continual transitions during each bit period and these transitions can be used to synchronize the clock timing signal.

Figure 2.23: Manchester Encoding Operation

Duplex Systems
Duplex systems transfer information between 2 or more users. Duplex systems may transfer information in one direction at a time on the same channel (simplex), in two directions at different times (half duplex), or simultaneously on two different channels (full duplex). There are various approaches including TDD that allow the appearance of full duplex operation although the actual transmission system uses simplex or half duplex operation.

Simplex communication allows the transmission of information between users, but only one direction at a time on the same channel or frequency. The common use of Simplex systems is traditional television or audio broadcast radio systems that transmit a signal from a single transmitter to many receivers. Figure 2.24 shows the communication process between two people using alternating (simplex) transmission and reception. This diagram shows that person 1 turns on (keys-on) the transmitter and begins to transmit audio. When person 1 transmits, their receiver is off or disconnected from the antenna. Person 2 hears the communication from person number one. When person 2 determines that person 1 has finished talking, person 2 turns on (keys-on) their transmitter and begins to talk on the same radio channel frequency. When person 2 transmits, their receiver is off or disconnected from the antenna. This diagram shows that simplex communications does allow two-way conversation. However, only one person can talk at any specific time.

Figure 2.24: Simplex Operation <ag_simplex_operation>

Half Duplex
Half duplex communication provides the ability to transfer voice or data information in either direction between communications devices but not at the same time. The information may be transmitted on the same frequency or divided into different channels. When divided into different

channels, one channel of frequency is used for transmitting and the other channel or frequency is used for receiving. The use of different frequencies is common in half duplex radio transmission because the transmitter and receiver are commonly connected to the same antenna. If the same transmitter and receiver frequency were used, the high transmitter power would probably destroy the receiver circuitry.

Full Duplex (FDX)
Full Duplex communication is the process of transferring of voice or data signals in both directions at the same time. Full duplex operation normally assigns the transmitter and receiver to different communication channels. When the communications system uses two different frequencies for simultaneous communication, it is called frequency division duplex (FDD). One frequency is used to communicate in one direction and the other frequency is required to communicate in the opposite direction. The definition of full duplex becomes confusing when it is applied to the end result of simultaneous voice and data communication. This is because it is possible to provide information at the input and output of a communication system while not actually sending the information simultaneously in a communication system. When a communication system provides for simultaneous two-way communication by time sharing, it is called time division duplex (TDD).

Frequency Division Duplex (FDD)
Frequency division duplex is a communications channel that allows the transmission of information in both directions (not necessarily at the same time) via separate bands (frequency division). Figure 2.25 shows a frequency division duplex (FDD) system. In this system, the transceivers contain a transmitter and receiver that are operating on two different frequencies. One frequency is used to send signals in one direction and the other frequency is used to send signals in the opposite direction. FDD allows for the simultaneous audio communication between users.

Figure 2.25: FDD Duplex System Operation

Time Division Duplex (TDD)
Time division duplex (TDD) communication uses a single channel or frequency to provide simultaneous two-way communications between devices by time-sharing. Figure 2.26 shows the basic operation of a TDD system Time division duplex (TDD) communications uses a signal channel or frequency to provide simultaneous two-way communications between devices by time-sharing. When using TDD, one device transmits (device 1), the other device listens (device 2) for a short period time. After the transmission is complete, the devices reverse their role so device 1 becomes a receiver and device 2 becomes a

transmitter. The process continually repeats itself so data appears to flow in both directions simultaneously.

Figure 2.26: TDD Duplex Systems

Access Multiplexing
Access multiplexing is a process used by a communications system to coordinate and allow more than one user to access the communication channels within the system. There are four basic access multiplexing technologies used in wireless systems: frequency division multiple access (FDMA), time division multiple access (TDMA), code division multiple access, (CDMA), and space division multiple access (SDMA). Other forms of access multiplexing (such as voice activity multiplexing) use the fundamentals of these access-multiplexing technologies to operate. FDMA systems use a process of allowing mobile radios to share radio frequency allocation by dividing up that allocation into separate radio channels. There, each radio device can communicate on a single radio channel during communication. TDMA systems allow several users to share a single radio channel by dividing the channel into time slots. When a mobile radio communicates with a TDMA system, it is assigned a specific time position on the radio channel. By allow several users to use different time positions (time slots) on a single radio channel, TDMA systems increase their ability to serve multiple users with a limited number of radio channels. Code division multiple access (CDMA) is a form of spread spectrum communication. CDMA is a method of spreading information signals (typically digital signals) so the frequency bandwidth of the radio channel is much larger than the original information bandwidth. Some systems coordinate system access on the same radio channels that are used for communication and other systems use a separate (dedicated) control channel. When using a control channel to coordinate access to the system, it is called an access control channel. The access control channel coordinates the random requests for service that is received from users (mobile radios) in the system. The control channel may also transfer identification information that allows the system to determine if the user is authorized to receive access to the system. Figure 2.27 shows the common types of channel-multiplexing technologies used in wireless systems. This diagram shows that FDMA systems have multiple communication channels and each user on the system occupies an entire channel. TDMA systems dynamically assign users to one or more time slots on each radio channel. CDMA systems assign users a unique spreading code to minimize the interference receive and cause with other users. SDMA systems focus radio energy to the geographic area where specific users are operating.

Figure 2.27: Channel Multiplexing

Network Access Control
Network access control is a process of coordinating access of data communication devices to a shared communications media (transmission medium). Network access control is a combination of media access control (MAC) and service authorization. There are two key ways data communication devices can access communication systems: noncontention based and contention based. Non-contention based regularly poll or schedule data transmission prior to allowing access attempts to the system. An example of a non-contention based data communication system is token ring. In the token ring system, only the data communication device that has the token is allowed to transmit. This ensures that other data devices will not interfere with the data transmission. Contention based access control systems allow data communication devices to randomly access the system through the sensing and coordination of busy status and detected collisions. Carrier sense multiple access (CSMA) with collision detection (CSMA/CD) or collision avoidance (CSMA/CA) listen to the data activity first to determine if the systems is not busy (carrier sense) before they begin a transmit request. After the device transmits its required, it waits to hear if the system has acknowledged its required (usually an echo of its original signal). If the CSMA/CD device does not hear an acknowledgement, it will wait a random amount of time before transmitting another data transmission service request. The CSMA/CA system differs from the CSMA/CD system by the assignment of different access wait periods to different priority groups of devices. This allows high priority devices (such as a system management data terminal) to attempt access before a lower priority device (e.g., web browsing terminal). Figure 2.28 shows the key ways networks can control data transmission access: non-contention based and contention based. This diagram shows that non-contention based regularly poll or schedule data transmission access attempts before computers can begin to transmit data. This diagram shows that a token is passed between each computer in the network and computers can only transmit when they have the token. Because there is no potential for collisions, computers do not need to confirm the data was successfully transmitted through the network. This diagram also shows contention based access control systems allow data communication devices to randomly access the system through the sensing and coordination of busy status and detected collisions. These devices first listen to see if the system is not busy and then randomly transmit their data. Computers in the contention-based systems must confirm that data was successfully transmitted through the network, because there is the potential for collisions.

Figure 2.28: Data Network Access Control

Chapter 3: Signal Processing
Signal processing is the method used to modify signals from one form to another. Some of the more common signal processing functions in telecommunications involves the modifying of analog, digital, and radio frequency signals.

Analog Signal Processing
Analog signals (continuously varying signals) may be processed by filters, shaping circuits, combiners, and amplifiers to change their shape and modify their content.

Signal Filtering
Signal filter may remove (band-reject) or allow (band-pass) portions of analog (possibly audio signals) that contain a range of high and low frequencies that are not necessary to transmit. In some cases, filters may be used to combine signals (at different frequencies). These additional signals may be multiple channels or they may be signals that are used for control purposes. If control signals are added to an analog signal that is transmitted, they are usually removed from the audio signal in the receiver by filtering. Figure 3.1 shows typical audio signal processing for a communications transmitter. In this example, the audio signal is processed through a filter to remove very high and very low frequency parts (audio band-pass filter). These unwanted frequency parts are possibly noise and other out of audio frequency signals that could distort the desired signal. The high frequencies can be seen as rapid changes in the audio signal. After an audio signal is processed by the audio band-pass filter, the sharp edges of the audio signal (high frequency components) are removed.

Figure 3.1: Audio Signal Filtering Operation

Low Pass Filter (LPF)
A low pass filter passes all frequencies below a specified frequency with little or no loss, but that significantly attenuates higher frequencies.

Figure 3.2 shows how a low pass filter is used to block mid and high frequency component parts of an input signal and allow low frequency components (signals) to pass through. In this example, both the mid and high frequency noise signals are highly attenuated by the bandpass filter while the desired low frequency is allowed to pass through the filter with minimal attenuation.

Figure 3.2: Low Pass Filter Operation

High Pass Filter (HPF)
A high pass filter is a device or circuit that passes signals of higher than a specified frequency but attenuates signals of all lower frequencies. Figure 3.3 shows how a high pass filter is used to block low and mid frequency component parts of an input signal and allow high frequency components (signals) to pass through. In this example, both the low and mid frequency noise signals are highly attenuated by the band pass filter while the desired high frequency is allowed to pass through the filter with minimal attenuation.

Figure 3.3: High Pass Filter Operation

Band Pass Filter (BPF)
A band pass filter is designed to reject or block all frequencies not within a given bandwidth. Such filters may be used to reject noise or other signal bands close in frequency to that of the desired signal. Filters typically require a trade-off among how much signal they pass (the amount of loss), how strongly they reject the undesirable signal, and how sharp the dividing line between passed and unpassed signals is. Figure 3.4 shows how a band pass filter is used to block high and low frequency component parts of an input signal. In this example, both the high frequency noise and low frequency noise are attenuated by the band pass filter. This example shows the desired frequency is allowed to pass through the filter with minimal attenuation.

Figure 3.4: Bandpass Filter Operation

Notch Filter (Band Reject)
A notch filter is a circuit or assembly that is designed to attenuate a specific frequency band; also known as a “band-stop” or “band reject” filter. Notch filters are sometimes used to restrict access to video signals that are transmitted through a cable television distribution system. Figure 3.5 shows a notch (band reject) filter that is used to block a specific frequency band (television channel) from a multi-channel input signal. In this example, a television system is broadcasting many television channels. This diagram shows how a notch filter can block a specific channel (such as a pay for subscription channel) from being received by a customer.

Figure 3.5: Notch Filter Operation

Signal Amplification
Signal amplification is a process of sensing an input (usually low level) signal and converting the signal into a larger version of itself. An amplifier device provides this conversion process. Amplifiers increase both the desired signal and unwanted noise signals. Noise signals are any random disturbance or unwanted signal in a communication system that tends to obscure the clarity of a signal in relation to its intended use. Figure 3.6 shows the basic process of amplification. An amplifier uses an input signal to control the current or voltage of a device that has an external power supply. The ability of the input signal to control the current or voltage allows the replication of the original signal. The ratio of input signal to the output signal level is called the gain (amount of amplification).

Figure 3.6: Signal Amplification Operation

Signal Shaping
Audio signals may be processed by shaping circuits to add or remove emphasis of frequency (tone) or intensity (volume). When the signal processing involves differences of amplification of specific frequency components of an input signal, it is called pre-emphasis and de-emphasis. Signal processing that involves relative changes in the amount of amplification dependent on the level of input signal, it is called companding and expanding. Some analog transmission systems use pre-emphasis circuits to amplify the high frequency components of the audio input signal that allow the modulation system to be more effective. Certain modulation systems do not respond well to low amplitudes of high frequency input signals. By boosting the high frequency component of the input audio signal, the modulator better translates the input signal into a modulated carrier signal. When pre-emphasis is used for transmission, a matched de-emphasis system is used in the receiver to convert the boosted high frequency component back into its original low signal level. The intensity of an audio signal can vary dramatically because some people talk loudly and others talk softly. A system that reduces the amount of amplification (gain) of an audio signal for larger input signals (e.g., louder talker) is called companding. The use of companding allows the level of audio signal that enters the modulator to have a smaller overall range (higher minimum and lower maximum). High signals and low signals input to a modulator may have a different conversion level (ratio of modulation compared to input signal level). This can create distortion so companding allows the modulator to convert the information signal (audio signal) with less distortion. Of course, the process of companding must be reversed at the receiving end, called expanding, to recreate the original audio signal. Figure 3.7 shows the basic signal companding and expanding process. This diagram shows that the amount of amplifier gain is reduced as the level of input signal is increased. This keeps the input level to the modulator to a relatively small dynamic range. At the receiving end of the system, an expanding system is used to provide additional amplification to the upper end of the output signal. This recreates the shape of the original input audio signal.

Figure 3.7: Analog Signal Companding and Expanding Operation

Mixers are circuits or assemblies that are used to combine two or more signals to produce a third signal that is a function of the input waveforms. Figure 3.8 shows how a mixer combines two signals to produce a sum or difference frequency. This diagram shows this mixer contains a diode (non-linear device) that allows the two-incoming signals to interact with each other to produce the difference (subtractive) frequency and sum (additive) frequencies. The output of this mixer circuit contains a tuned circuit (resonant circuit) that only allows the difference frequency to transfer out of the mixer.

Figure 3.8: Mixer Operation

Digital Signal Processing
Digital signals processing refers to a category of electronic devices that represent and process information that are in discrete signal level (digital) formats. Digital signal processing refers to the manipulation of digital signals to change their content and to add error detection and correction capability. Digital signals typically vary in two levels; on (logic 1) and off (logic 0). A bit is the smallest part of a digital signal, typically called a data bit. A bit typically can assume two levels: either a zero (0) or a one (1). A byte is an agreed-upon group of bits, typically eight. A byte typically represents one alphabetic or special character, two decimal digits, or eight binary bits of information. When analog signals are converted to digital format, the digital signals represent the original analog waveform. Just like analog signals that may be processed by filters, shaping circuits, combiners and amplifiers, digital signals can be processed to produce similar functions. However,

because the signal is in digital form, these functions are performed by software programs that manipulate the data. Unlike analog signals, digital signals can be recreated to their original form. This process is called signal regeneration. To increase the efficiency of a transmission signal (allow more users per channel), digital signals can be analyzed for redundancy and the digital signal data can be compressed. Digital signals can also be processed in a way that helps overcome the effects signal distortion that can result in the incorrect determination of a digital signal (whether a zero or one had been sent). This is called error detection and error correction processing. When digital signals represent the original analog signal, advanced echo canceling software programs can be used to reduce the effects of echoes that are caused by feedback in the audio and transmission system. Some systems use dedicated digital signal processors (DSPs) to manipulate the incoming digital information via a program (stored instructions) that produce a new digital output. This allows software programs to perform many functions (such as signal filtering) that previously required complex dedicated electronic circuits.

Digitization of an Analog Signal
Analog signals must be converted to digital form for use in a digital wireless system. To convert analog signals to digital form, the analog signal is digitized by using an analog-to-digital (pronounced A to D) converter. The A/D converter periodically senses (samples) the level of the analog signal and creates a binary number or series of digital pulses that represent the level of the signal. The common conversion process is Pulse Code Modulation (PCM). For most PCM systems, the typical analog sampling rate occurs at 8000 times a second. Each sample produces 8 bits digital that results in a digital data rate (bit stream) of 64 thousand bits per second (kbps). Figure 3.9 shows how an analog signal is converted to a digital signal. This diagram shows that an acoustic (sound) signal is converted to an audio electrical signal (continuously varying signal) by a microphone. This signal is sent through an audio band-pass filter that only allows frequency ranges within the desired audio band (removes unwanted noise and other non-audio frequency components). The audio signal is then sampled every 125 microseconds (8,000 times per second) and converted into 8 digital bits. The digital bits represent the amplitude of the input analog signal.

Figure 3.9: Signal Digitization Digital bytes of information are converted to specific voltage levels based on the value (weighting) of the binary bit position. In the binary system, the value of the next sequential bit is 2 times larger. For PCM systems that are used for telephone audio signals, the weighting of bits within a byte of information (8 bits) is different than the binary system. The companding process increases the dynamic range of a digital signal that represents an analog signal; smaller bits are given larger values that than their binary equivalent. This skewing of weighing value give better dynamic range. This companding process increases the dynamic range of a binary signal by

assigning different weighted values to each bit of information than is defined by the binary system. Two common encoding laws are Mu-Law and A-Law encoding. Mu-Law encoding is primarily used in the Americas and A-Law encoding is used in the rest of the world. When different types of encoding systems are used, a converter is used to translate the different coding levels.

Digital Signal Regeneration
To overcome the effects of noise on transmitted signals, digital transmission systems use digital signal regeneration to restore the quality of the signal as it moves through a network. Digital signal regeneration is the process of reception and restoration of a digital pulse or lightwave signal to its original form after its amplitude, waveform, or timing have been degraded by normal factors during transmission. The resultant signal is virtually free of noise or distortion. Figure 3.10 shows the process of digital signal regeneration. This example shows how an original digital signal (a) has a noise signal (b) added to it that produces a combined digital signal with noise (c). The regeneration process detects maximum and minimum expected values (threshold points) and recreates the original digital signal (d).

Figure 3.10: Digital Signal Regeneration Operation

Data Compression
To increase the amount of information that a transmission system can transfer, digital systems may use data compression. Data compression is a processing technique for encoding information so that fewer data bits of information are required to represent a given amount of data. Compression allows the transmission of more data over a given amount of time and circuit capacity. It also reduces the amount of memory required for data storage. Digital compression analyzes a digital signal for either redundant information (repeated 1’s or 0’s) or may analyze the information content of the digital signal into component parts (such as speech patterns or video frames). All of this processing allows the data transmission rate to be reduced by sending only the characteristics of the signal rather than the complete digital signal. Some data compression systems can only reduce data rates by a factor of 2:1 (e.g., ADPCM audio compression) while advanced digital audio compression can only reduce data rates by a factor of approximately 200:1 (e.g., MPEG video compression). When used in combination of data compression and decompression, the device is called a COder/DECoder (CoDec).

When a digital signal is compressed for voice communications, it is called a voice coder (Vocoder) or speech coder. The Vo-coder is a digital signal processing device that analyzes speech signals so that it can produce a lower data rate compressed digital signal. The difference between standard data compression and voice data compression is the analysis of the information source (speech) and elimination of compression process for non-voice signals. Speech coding usually involves the use of data tables (called code books) that represent information parts that can be associated with human sound. Because non-human sounds can be eliminated from the code book, this allows the number of bits that can be used to create a compressed digital voice signal to be reduced. Figure 3.11 shows the basic process that is used for digital voice compression process. In this diagram, a digital audio signal (64 kbps PCM signal) is continuously applied to a digital signal analysis device. The analysis portion of the speech coder extracts the amplitude, pitch, and other key parameters of the signal and then looks up related values in the code book for the portion of sound it has analyzed. Only key parameters and code book values are transmitted. This results in data compression ratios of 4:1 to over 16:1.

Figure 3.11: Digital Voice Compression Operation

Error Detection and Error Correction
To help reduce the effects of errors on data transmission, error detection, and error protection systems are used in most communication systems. Error detection systems use a process of adding some data bits to the transmitted data signal that are used to help determine if bits were received in error due to distorted transmission. Error correction is made possible by sending bits that have a relationship to the data that is contained in the desired data block or message. These related bits permit a receiver of information to use these extra information bits to detect and/or correct for errors that may have occurred during data transmission. A common measurement of the performance of a communication system is the amount of bits received in error, called the bit error rate (BER). The BER is the ratio of bits received in error compared to the total number of bits received. Error detection processing involves the creation of additional bits that are sent with the original data. The additional check bits are created by using a formula calculation on the digital signal prior to sending the data. After the digital signal is received, the formula can be used again to create check bits from the received digital signal. If the check bits match, the original digital signal was received correctly. If the check bits do not match, some (or all) of the digital signal was received in error. This process is called error detection. Some digital systems use sophisticated mathematical formulas to create the check bits so that the check bits can be used to make corrections (or predictions of the correct bits) to the received digital signal. This process is called error correction.

Figure 3.12 shows the basic error detection and correction process. This diagram shows that a sequence of digital bits is supplied to a computing device that produces a check bit sequence. The check bit sequence is sent in addition to the original digital bits. When the check bits are received, the same formula is used to check to see if any of the bits received were in error.

Figure 3.12: Error Detection and Correction

Convolutional Coding
Convolutional coding is an error correction process, which uses the input data to create a continuous flow of error protected bits. As these bits are supplied to the convolutional coder, an increased number of bits is produced. The length of this code is k. Convolutional coding is often used in transmission systems that often experience burst errors such as wireless systems. Figure 3.13 shows a convolutional coder that continuously receives the data bits in sequence to create a new digital signal that combines both the original information and new error protection bits. This example shows a 1/2 rate convolutional coder that generates two bits for every one that enters.

Figure 3.13: Convolutional Coding

Echo Cancellation
Echo cancellation is a process of extracting an original transmitted signal from the received signal that contains one or more delayed signals (copies of the original signal). Echoes may be created in a baseband or broadband signal. When echoes occur on an audio baseband signal, it is usually through acoustic feedback where some of the audio signal transferring from a speaker into a microphone. When echoes occur on a broadband signal, it is usually the result of the same signal (such as a radio signal) that travels on different paths to reach its destination. In either

case, echoed signals cause distortion and may be removed by performing advanced signal analysis and filtering. Figure 3.14 shows how an echo canceling system can remove unwanted echo signals in telecommunication circuits. In this diagram, the sources of echo signals include electrical mismatches in hybrid combining circuits and signal leakage in end user devices. The echo cancellation system uses a process that senses a complex audio signal and predicts the distortions that are produced by echo signals. This allows it to create an echo canceling signal (inverse signal of the echo signal) that is combined with the input signal so the distortion effects of the echo canceling can be removed.

Figure 3.14: Echo Cancelling System Echoed signals can also occur in signals other than audio signals. When echoes occur on radio channels (the broadband signal), it is usually the result of the same signal that travels on different paths to reach its destination. This is called multipath propagation. Echo canceling can be used to reduce the effects of radio multipath propagation.

Digital Signal Processor (DSP)
Digital signal processors are integrated circuits (chips) or assemblies that are designed specifically for high-speed manipulation of digital information. DSP chips operate using software programs to allow them to perform complex signal processing operations such as filtering, modulation, data compression, and information processing. Figure 3.15 shows a typical digital signal processor that is used in a digital communication system. This diagram shows that a DSP contains a signal input and output lines, a microprocessor assembly, interrupt lines from assemblies that may require processing, and software program instructions. This diagram shows that this DSP has 3 software programs, digital signal compression, channel coding, and modulation coding. The digital signal compression software analyzes the digital audio signal and compresses the information to a lower data transmission rate. The channel coding adds control signals and error protection bits. The modulation coding formats (shapes) the output signal so it can be directly applied to an RF modulator assembly. This diagram also shows that an optional interface is included to allow updating of the software programs that are stored in the DSP.

Figure 3.15: Digital Signal Processor (DSP) Operation

RF Signal Processing
RF signals are radio frequencies within the electromagnetic spectrum normally associated with radio wave propagation. RF sometimes is defined as transmission at any frequency at which electromagnetic energy radiation is possible, usually above 150 kHz. Radio frequency (RF) signal processing refers to a category of electronic devices that represent and process information that are in analog high frequency level formats. RF signal processing for telecommunication usually involves the manipulation of signals by changing their signal level (gain control) or through the manipulation of modulation characteristics.

Automatic Gain Control (AGC)
Automatic gain control (AGC) is an assembly or circuit that is part of a communications receiver. AGC automatically adjusts the received signal level so that it’s approximately the same regardless of the received radio signal level. AGC is often used to supply a constant level signal to a demodulator assembly. Figure 3.16 shows how a varying level signal that is supplied to a communication receiver can be adjusted to a near constant by an automatic gain control (AGC) system. This diagram shows that a varying radio signal is supplied to a signal level detector (diode) and a variable gain amplifier. The output of the detector is used to inversely vary the gain of the amplifier. As a result, the amplifier produces a near constant signal level that can be provided to the demodulator assembly.

Figure 3.16: Automatic Gain Control (AGC) Operation

Digital Modulation
A common form of modulation used for RF signals used in telecommunication systems is digital modulation. Digital modulation is the process of modifying the amplitude, frequency, or phase of a carrier signal using the discrete states (On and Off) of a digital signal. When modulating a carrier signal using a digital information signal, a rapid change to the carrier wave can occur. These rapid changes result in the creation of other signals that are usually undesirable. As a result, digital modulation usually includes a process of adjusting the maximum rate of change of the input signal (rounding the digital signal edges) and filtering out some of the unwanted signals that are created during the transition. Figure 3.17 shows different forms of digital modulation. This diagram shows ASK modulation which turns the carrier signal on and off with the digital signal. FSK modulation shifts the frequency of the carrier signal according to the on and off levels of the digital information signal. The phase shift modulator changes the phase of the carrier signal in accordance with the digital information signal. This diagram also shows that advanced forms of modulation, such as QAM, can combine amplitude and phase of digital signals.

Figure 3.17: Digital Modulation

Chapter 4: Transmission Systems
Transmission systems interconnect communication devices (end nodes) by guiding signal energy in a particular direction or directions through a transmission medium such as copper, air, or glass. A transmission system will have at least one transmitting device, a transmission medium, and a receiving device. The transmitting communication device is capable of converting an information signal into a form of electrical, electromagnetic wave (radio), or optical signal that allows the information to be transferred through the transmission medium. The receiving communication device converts the transmitted signal into another form that can be used by the device or other devices that are connected to it. Transmission systems can be unidirectional (one direction) or they can be bi-directional (two directions). The basic types of transmission mediums include copper wire, coaxial cable, free space/air, fiber optic cable, and mechanical transmission line. Copper and coaxial wire is primarily used for low to moderate frequency transmission over a few miles. Free space/air systems can transmit hundreds of miles but have limited bandwidth and are susceptible to noise interference. Fiber optic cable is capable of carrying high-speed data signals (as light pulses) over thousands of miles. Mechanical (acoustic wave) transmission lines transmit over very short distances (only a few millimeters) and are used for signal filtering components. Different types of transmission lines have varying performance characteristics and may be susceptible to interference during signal transmission. These characteristics include the available frequency bandwidth (frequency response), how much signal leakage that occurs (cross talk), and the susceptibility of absorbing other signals (signal ingress). The construction of the transmission line itself may cause distortions in the transmitted signal. This includes unterminated line splices (bridge tap reflections), poor line splices, and line resistance (signal attenuation). Other characteristics, such as varying delays to different frequency ranges, may cause group dispersion (smearing) of the desired signal. To allow devices to communicate with each other over a transmission line, carrier systems specify the signal types and levels along with specific protocol controls (communication rules). These carrier systems are often specific to the transmission medium such as copper or fiber. Some of the more popular carrier systems include plain old telephone service (POTS), digital signaling carrier (DSx), digital subscriber line (DSL), and optical carrier (OCx). To coordinate the transmission line, signaling messages are sent between communication devices. Some of these control messages are sent along with the data on the transmission line (called in-band signaling) and others are sent through another path or network (called out-of-band signaling). In some cases, a transmission path may only be a portion of a path (a logical path) through a transmission line. The length of a transmission may be extended through the use of amplifiers or repeaters. Most telecommunication customers are served by copper cable (twisted pair or coax) terminated by the local telephone company in a telephone network interface box, called a network termination (NT). The NT is normally located on the side of the building. The network termination isolates the network from the wiring inside the building. From the NT, the “inside wiring” extends the telephone cable to all internal wall and floor jacks. To reduce the number of copper pairs, telephone systems use a hybrid transmission system to allow both transmission and reception on a single pair of copper wires. By combining both transmit and receive audio signals using a special hybrid combiner, only one pair of wires is

required to operate a standard home telephone. These two lines are routinely referred to as “tip and ring.” This single pair of wires also provides dial tone, dialing pulses or tones, ringing (high voltage signal), and a talk path. Most of the information that is transferred in voice conversation occurs at frequencies below 3,300 cycles per second (Hertz or Hz) and above 300 Hz. This allows telephone systems to restrict the audio frequency range for voice grade circuits from 300Hz to 3300Hz. Using a restricted frequency range reduces the transmission line and system switching performance requirements. The limiting of the audio frequency range is accomplished with the use of devices known as band-pass filters. Band-pass filters strongly attenuate signal frequencies above and below specific frequencies. It is possible to send digital information through the hybrid network through the use of a modulator/demodulator (MoDem). The MoDem converts digital signals into analog tones that can be transmitted on standard telephone lines. Telephone transmission lines can be divided into access lines (local loops) and interconnection lines (trunks). Often referred to as 1FB’s or B1’s, local loops refer to all two-wire voice grade connections between a residence or place of business and the telephone company’s serving end office (e.g., where the dial tone originates). Interconnection trunks refer to high capacity groups of circuits connecting switching sites such as end offices or other switching centers.

Telecommunication transmission technology is often lumped into two categories: analog transmission or digital transmission. Figure 4.1 depicts a basic telecommunications transmission system that transfers digital information from one source to an information receiver. The information source is supplied to an end-node that converts the information to a form that can be transmitted through the transmission medium (air, copper, or fiber). The receiving end-node converts the transmission signal into a form that is compatible for the receiver of the information.

Figure 4.1: Basic Transmission System

Analog Transmission
Analog transmission is a process of transferring signals between end-nodes than can have many different signal levels and frequencies. Because analog signals can continuously change to many different levels (voltages) at changing rates (frequencies), the transfer of analog signals (such as an audio signal) requires the transmission medium to have similar transfer characteristics to all parts (levels and frequencies) of the transmission signal. Analog transmission systems must be

robust to transfer the signal unaltered for specific voltage levels and frequency components (e.g., high and low frequency). Figure 4.2 shows an analog transmission system. This diagram shows that an audio acoustic (sound) signal is converted by a microphone to an audio electrical signal prior to transmission on a copper line. This audio electrical signal is amplified by an end-node to increase the signal level for transmission on a copper wire. This amplification is necessary to overcome the transmission loss of the copper wire. As the signal progresses down the copper wire, some of the signal energy is converted to heat reducing the signal level. Another amplifier (the receiving end-node) is located at the receiving end to increase the signal to a level suitable for the information receiver (audio speaker).

Figure 4.2: Analog Transmission System

Digital Transmission
Digital transmission is the process of transferring information from node to node in a form that can only have specific levels (usually logic 1 and logic 0). Digital signals have a limited number of different levels (voltages) that represent digital information. Transferring digital signals (such as a computer’s data signal) only requires the transmission medium to transfer two levels without precisely (linearly) transferring levels in between the two levels. Figure 4.3 shows a digital transmission system. This diagram shows a computer that is sending digital data (one equal to +5 volts and zero equal to 0 volts) to an end-node. The end-node is a channel service unit (CSU) and digital service unit (DSU) that converts the levels from the computer to levels suitable for the copper wire transmission medium (logic 1 = +5V and logic 0 = 5V). As the digital signal transfers down the copper wire, some of the energy is converted to heat and some of the frequency components are attenuated resulting in a slightly distorted (rounded) digital pulse arriving at the receiving end-node. Because the receiving CSU/DSU only needs to sense two levels, it is able to re-create the original undistorted digital signal (also known as digital signal regeneration).

Figure 4.3: Digital Transmission System

Repeaters are devices or circuits that are located between transmitting and receiving devices to improve the quality the signal that is delivered between them. A repeater obtains some or all of the signal from the transmitter, amplifies and may adjust (change a frequency) or filter the signal, and retransmits the signal to the receiver(s). Repeaters can be analog or digital. Analog repeaters amplify the received signal for retransmission. Analog repeaters amplify both the desired signal and any noise that is added to the communication lines. This limits the maximum number of analog repeaters that can be used and this limits the maximum distance for analog communication lines. Digital repeaters can receive and recreate digital signals. Digital repeaters are also called regenerative repeaters. The regenerative process allows digital signals to be transferred at great distances with minimal errors at the receiving end.

Data Modems
Data modems are devices that convert signals between analog and digital formats for transfer to other lines. Data modems are used to transfer data signals over conventional analog telephone lines. The term modem also may refer to a device or circuit that converts analog signals from one frequency band to another. A point-to-point analog data circuit requires a modem at each end to transfer digital signals. The type of modems used on each end must be compatible due to encoding and decoding processes. Analog communication lines are restricted to audio bandwidth of 300 Hz to 3300 HZ. To communicate digital data and control signals, the modems vary the frequency of the carrier in each direction based on an agreed process for encoding data bits. Figure 4.5 shows how a data modem converts digital information into analog signals that can be transmitted on an analog communications network. In this example, the data signal comes from the computer (called the data terminal equipment (DTE)), via an RS-232 serial data interface. The RS-232 data interface uses pre-defined signaling commands and data transmission rates to communicate with the data modem. The modem performs a digital-to-analog conversion and from the line to the DTE, an analog-to-digital conversion.

Figure 4.5: Data Modem

Copper Wire
Twisted pair copper wire is the most utilized telecommunications medium and thus has the largest installed base in the worldwide telecommunications infrastructure by far. It is relatively inexpensive, easy to install, and, locally available in quantities. Practically every residential telephone worldwide connects to the local telephone company via a twisted pair jack or block located on an inside wall or baseboard in the residence. Because of these factors, intensive research and development (R&D) funding continues to be allocated for the purpose of extending the usefulness of twisted pair copper wire. By enhancing its ability to carry information faster and farther, R&D will continue as long as the resulting enhancements meet, or exceed, many of the requirements placed on the industry by customer demands. Twisted pair copper wire will remain a logical, cost-effective medium for providing many commercial services for some time to come. Figure 4.6 illustrates the transmission of electrical energy (electricity) via copper wire. Note that the electricity is conducted via the outer surface of the wire, not the inner. Consequently the greater the outer surface, the more electricity that can be conducted. In other words, the larger the wire diameter the more outer surface there is to conduct electricity. Figure 4.6: Electrical Transmission through Copper Wire In the telecommunications industry copper wire is normally referred to as twisted pair. Through twisting wire into pairs, electrical radiation (eddy fields or cross talk) is reduced. This reduction is both radiation off the pair and the susceptibility to radiation from other pairs and sources. It has been found that the tighter the twist (this is still in R&D) the less interference (cross talk) and the higher the speed at given error rate. Twisted pair cables come in a variety of sizes and jackets (outer covering). Twisted pair cables are jacketed and may contain from 2-pair to several hundred pairs. Twisted pair wire comes jacketed in PVC or plenum-rated. “Plenum” is the name given to the non-toxic PVC-like jacket that is authorized by local fire ordinances for use in ceilings and walls considered to be “airreturn”. Twisted pair cables are produced as either shielded twisted pair (STP) or unshielded twisted pair (UTP). When a twisted pair installation is to occur in an area that has abnormally high levels of electromagnetic energy, STP is recommended. In most office settings UTP is the standard, However, even in such seemingly neutral environments there can be problems such as fluorescent lighting fixtures and parallel runs with electrical wiring. A good installation plan prepared by a certified wiring engineer reduces the likelihood of such problems and also enhances the probability that the final infrastructure will operate at the desired performance (to allow the desired data transmission speed).

In office or campus environments UTP and/or STP provide the wiring infrastructure for LAN’s, some host-based data applications, video, and voice. Central office lines are normally delivered as twisted pairs to the client telephone or computer systems. Such lines range from single station analog lines up to and including T-1 (US, 1.544Mbps) or E-1 (Europe, 2.048Mbps). Of course this includes the intermediate digital service known as digital subscriber line (DSL). In addition, twisted pair cable that is to be directly buried in the ground has a special construction to prevent water seepage. Cables to be run overhead outside are constructed with an extra steel cable known as a strength member. The strength member supports the weight of the cable between the poles from which the cable is suspended. This prevents the cable from sagging and ultimately rendering it useless. Finally, the US government contracts for “tap-proof” cable that has a special outer conductor, located under the outer covering, that is capable of conducting as much as 2,000 volts. Figure 4.7 lists the types of twisted pair cables and their rated capacities and maximum links. Copper wire cable is “Category” rated. This table shows that the category varies based on the design of the cable (shielded or unshielded), size (gauge) of the wire, and the types of insulation material used.

Figure 4.7: Copper Wire Pair Information Capacity

Coaxial Cable (Coax)
Coaxial cable is a transmission line that is constructed from a center conductor that is completely surrounded by shield conductor. The center conductor and the shield conductor are separated by an insulation material. The shield can be inter-woven strands of wire or metal foil. Because the center conductor is complete surrounded by the shield, in an ideal coaxial cable, all the transmitted energy is contained within the cable. Figure 4.8 shows a cross sectional view of a coaxial cable. This diagram shows a center conductor that is surrounded by an insulator (dielectric). The insulator is surrounded by the shield. This diagram shows that during transmission, electric fields extend perpendicular from the center conductor to the shield and magnetic fields form a circular pattern around the center conductor.

Figure 4.8: Cross Sectional View of Coax Cable Coaxial cable is best known as the medium for cable television. It was primarily chosen because of its durability, wide frequency bandwidth capacity (often up to 1 GHz bandwidth), and less rigid length restrictions. Coax (as it is normally called) is often used in local area networks (LAN) to transport high-speed data signals with relatively high security (low signal leakage). Twinax is a derivative of coax and is constructed as noted above with the exception that twinax uses two center conductors. Each center conductor is individually insulated, but, as with coax, each references the single shield for ground. The first local area networks (LAN’s) were almost exclusively centered around coax as the network medium of choice. It became the standard for the early LAN’s. There are three common types of coax that are, or have been, in use extensively with many computer systems: thicknet, thinnet, and twinax. Thicknet is often associated with the first Ethernet LAN’s and with high-speed bus cables used between mainframe computers and their peripherals. It is bulky and difficult to install but provides high speed and capacity where it is most critical. With respect to LAN’s, use of thicknet provides extra protection from electrical interference that may be encountered (e.g., such as on factory floors near assembly equipment). For most LAN’s installed in the early 2000’s, UTP or STP is used. Coax is used in cable television systems and for interconnection trunks at telephone company switching centers. This provides for relatively high data transmission capacity (e.g., for DS3 45 Mbps transmission). Figure 4.9 lists the most frequently encountered types of copper and coax line and their approximate information transmission capabilities.

Figure 4.9: Copper and Coax Cable Information Capacity

Power Line Carrier (PLC)
Power line carrier (PLC) is a transmission carrier wave signal that is simultaneously transmitted on electrical power lines. A power line carrier signal is above the standard 60 Hz powerline power frequency (50 Hz in Europe). Figure 4.10 shows three types of communication systems that use an electrical power distribution system to simultaneously carry information signals along with electrical power signals. In this example, the high voltage portion of the transmission system is modified to include communication transceivers that can withstand the high voltages while coupling/transferring their information to other receivers that are connected to the high voltage lines. This type of communication could be used to monitor and control power distribution equipment such as relays and transformers. This example also shows a power line distribution system that locates a communication node (radio or fiber hub) near a transformer and provides a data signal to homes connected to a transformer. This system could allow customers to obtain Internet access or digital telephone service by plugging the computer or special telephone into a standard power socket. The diagram also shows how a consumer may use the electrical wiring in their home as a distribution system for data (e.g. Ethernet) communication.

Figure 4.10: Power Line Data Transmission System

Free Space/Air
Transmission in free space and air can be accomplished by radio or light signals. Free space/air transmission is the transfer of signal energy through an unobstructed medium. Free space transmission occurs in an ideal medium (vacuum) that is free of objects or particles that may disrupt the transmission of signal energy. Transmission of signals in air has similar characteristics as free space transmission. However, particles in the air result in signal scattering and absorption during transmission. Free space transmission systems require a transducer to convert signal energy of one form into electromagnetic or optical energy for transmission. The transducer must also focus the energy so it may launch the energy in the desired direction. When air is the medium, particles in the air (such as water) may absorb or redirect (scatter) the transmitted signal. Figure 4.11 shows two types of free space transmission systems: radio and optical. The microwave transmission system shows that some of the electromagnetic energy is absorbed by the water particles in the air. The optical transmission system uses a laser and photo-detector. The optical transmission system shows that some of the optical energy is scattered in other directions as it passes through smog and water particles.

Figure 4.11: Free Space Transmission System In 1951, microwave radio transmission through free space became the backbone of the telecommunications infrastructure. Point-to-point microwave transmission systems have the data transmission capacity of hundreds of megabits per second. Although the extensive deployment of fiber optic cable has removed some of need for microwave radio systems, microwave radio is still used in places that are hard to reach or not cost effectively served by fiber cable. In addition, microwave radio free space technology is the basis for satellite communications. In the commercial broadcast industry satellites are fed from terrestrial sites called “mother stations.” The mother station transmits up to the satellite on multiple frequencies called “uplinks”. Transceivers on the satellite, referred to as “transponders”, retransmit the signals back down to earth. These signals are known as “downlinks”. On the ground, satellite dishes (focusing antennas) receive the downlink signals and a radio receiver converts these signals back into television images and sound. Home Box Office (HBO) and ShowTime are examples of commercial broadcast companies that use satellite almost exclusively for distribution. In developing countries and regions where the telecommunications infrastructure is of poor quality, radio, microwave, and satellite have been used to solve connectivity requirements in short order. Data radio and very small aperture satellite (VSAT) systems have allowed banks and other information-dependent companies to reliably connect to branch offices for the online exchange of information. Since the mid-1980’s data radio has played a major role in the telecommunications industry in developing countries where the copper wire infrastructure is generally of substandard quality. Microwave transmission systems transfer signal energy through an unobstructed medium (no blocking buildings or hills) between two or more points. In 1951, microwave radio transmission

systems became the backbone of the telecommunications infrastructure. Microwave systems require a transducer to convert signal energy of one form into electromagnetic energy for transmission. The transducer must also focus the energy (using an antenna dish) so it may launch the energy in the desired direction. Some of the electromagnetic energy that is transmitted by microwave systems is absorbed by the water particles in the air. Although the extensive deployment of fiber optic cable has removed some of need for microwave radio systems, microwave radio is still used in places that are hard to reach or not cost effectively served by fiber cable such as in developing countries. Figure 4.12 shows a terrestrial microwave system-connecting IXC switches in Philadelphia and New York City. The microwave signals are moved between the two switching offices through a series of relay microwave systems located approximately 30 miles apart. Microwave is a line-ofsight technology that must take the earth’s curvature into consideration. Also note that microwave towers are not limited to only facing one or two directions. A single tower can be associated with several other towers by positioning and aiming additional transceiver antennas at other microwave antennas on other towers.

Figure 4.12: Long Haul Microwave Modern optical transmission systems use infrared, and other laser optical signals to carry large amounts of information. Free space infrared and laser communication systems have been limited to span small distances of a few miles due to interference of the particles in air. Infrared systems have gained popularity because of their high bandwidth and ease of installation. Optical systems usually do not require government licenses or other authorization to use. These optical systems can be found connecting buildings on a campus and as supplements to wired LAN’s within an office or plant.

Fiber Optic Cable
Fiber optic cable is a strand of glass or plastic that is used to transfer optical energy between points. The size of most fibers is from 10 to 200 microns (1/100th to 1/5th of a mm). Optical fibers are typically used in a unidirectional mode (e.g., data moves in only one direction). Because of this, every transmission system requires at least two fibers (one for transmission and one for reception). For most fiber systems, the transmitting end-node uses a light amplification through stimulated emission of radiation (LASER) device to convert digital information into pulsed light signals (amplitude modulation). The light signals travel down the fiber strand by bouncing (reflecting) off the sides of the fiber (called the cladding) until they reach the end of the fiber. The end of the fiber is connected to a photo-detector that converts these light pulses back into their electrical signal form. Optical fibers are often characterized by either single mode or multimode transmission. Single mode of fiber transmission only allows a specific narrow wavelength of light to pass through the

fiber. Multimode fiber transmission allows a much wider wavelength of light to pass through the fiber by gradually bending different wavelengths back towards the center of the fiber. Single mode fiber strands are very narrow with a fiber diameter of 9-10 microns (1/100 of a mm). Multimode fibers are much wider as they can have a fiber diameter of 50-125 microns). Figure 4.13 shows single mode and multimode fiber lines. This diagram shows that multimode fibers have a relatively wide transmission channel that allows signals with different wavelengths to bend back into the center of the fiber strand as they propagate down the fiber. The diagram also shows that single mode fiber has a much smaller transmission channel that only allows a specific wavelength to transfer down the fiber strand.

Figure 4.13: Single and Multimode Fiber Lines Because of the relatively wide frequency bandwidth (and high data-transmission rate), multimode fibers are predominantly used for high-speed short runs such as those occurring within a building or around a campus. Typical multimode fiber runs are less than a mile, but can be several miles (2 – 5). Because of this, it is often referred to a “short haul” fiber solution. Single mode is a “long haul” fiber solution (50 – 75 mile runs). Single-mode fiber has been used by telephone companies and long distance carriers for several years to off-load expanding requirements from traditional terrestrial microwave. Single-mode fiber can transmit much further than multimode fibers. Single-mode applications use a diode laser as the light source, while multimode uses a light emitting diode (LED). Use of a LASER ensures a high energy at a very narrow optical bandwidth as compared to the LED that has optical energy distributed over a wide optical bandwidth. Single mode fibers use a narrow glass filament with a diameter of approximately 10 microns compared to multimode where the diameter ranges from 50 to 125 microns. The narrow channel of the single mode fiber minimizes the bending of the wave and this results in less dispersion (less smearing of the pulses) over distance. Figure 4.14 shows a fiberoptic communication system that is composed of two end-nodes and a fiber optic cable transmission medium. This diagram shows two optical network units (ONUs) that connect data networks together using fiber cable. This diagram shows that two fiber strands are needed: one for transmitting and one for receiving.

Figure 4.14: Fiber-Optic Cable Transmission System

To increase the capacity of fiber systems, multiple optical signals of different wavelengths are combined on a single strand of fiber. This is called wave division multiplexing (WDM). When 40 (or more) optical signals are combined on a single fiber strand, this is called dense wave division multiplexing (DWDM). Figure 4.15 shows how wave division multiplexing over fiber operates. This diagram shows that there are several lasers operating at different optical wavelengths (different colors/frequencies). Each laser converts an electrical signal into a pulsed light signal. These optical signals (optical carriers) are combined by an optical multiplexer (lens) for transmission through the optical fiber. At the receiving end, the different optical carriers are separated by an optical demultiplexer (lens) and each optical carrier is sent to a photo-detector. The photo-detector converts the optical signal back into its original electrical form.

Figure 4.15: Wave Division Multiplexing

Transmission Medium Limitations
Some of the limitations of transmission lines that reduce their ability to transfer analog and digital information include limited frequency response of the transmission lines, crosstalk, noise from external sources that cause distortion, non-terminated tap lines, and signal attenuation that results from line splices and line resistance.

Plain Old Telephone Service (POTS)
Plain old telephone service (POTS) is a transmission system that is used to provide basic telephone service. It is the common term used for residential telephone service. Between the late 1800’s through the 1990’s, telephone transmission had remained basically the same. Acoustic energy from the customer was converted to electrical signal by a microphone in a handset. This electrical energy was applied through a hybrid electrical device to the telephone line through the speaker in the handset. A telephone hybrid device (often called a “magic” device by telephone personnel) transferred energy from the microphone to the telephone 2-wire line while extracting most the remote microphone energy and applying it to the speaker. At the same time at the other end of the connection, the same process was occurring. Figure 4.23 shows how a typical analog telephone transmission line operates. In this diagram, audio from customer #1 is converted to electrical energy by microphone #1. This signal is applied to the telephone line via the hybrid adapter #1. A portion of this signal is applied to the handset speaker to produce sidetone (so the customer can faintly hear what they are saying). This audio signal travels down the telephone line to hybrid #2. Hybrid #2 applies this signal to speaker #2 so customer #2 can hear the audio from customer #1. When customer #2 begins to speak, microphone number #2 converts the audio to an electrical signal. This signal travels down the line to hybrid #1. Hybrid #1 subtracts the energy from microphone #1 (the combination of both signals are actually on the line) and applies the different (audio from customer #2) to the speaker #1.

Figure 4.23: Hybrid Telephone Operation

Integrated Services Digital Network (ISDN)
Integrated services digital network (ISDN) is a structured all digital telephone network system that was designed 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. There are two key user interfaces defined for ISDN networks: basic rate interface (BRI) and primary rate interface (PRI). The basic rate interface (BRI) is the smallest transmission system (or interface) available through ISDN. BRI provides for two 64 kbps bearer channels (B channels) and a 16 kbps signaling (data) channel (D channel). This configuration is also is also referred to as 2B+D. The primary rate interface (PRI) is a standard high-speed data communications interface that is used in the ISDN system. This interface provides a standard data rates for T1 1.544 Mbps and E1 2.048 Mbps. The interface can be divided into combinations of 384 kbps (H) channels, 64 kbps (B) channels and includes at least one 64 kbps (D) control channel.

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. Digital subscriber line (DSL) was first used in the 1960s to describe the T-1 circuits that were extended to the customer premises. Later the same term was used to describe ISDN basic rate interface (BRI) (2B+D, 144 Kbps) and primary rates interface (PRI) (23B+D, 1.544 Mbps). There are several different digital subscriber line technologies. Each of these DSL technologies usually has a prefix to indicate the specific variant of DSL technology. Hence, the “x” in xDSL indicates that there are many forms of xDSL technology. DSL transmission allows high-speed data transmission over existing twisted pair telephone wires. This has the potential providing high-speed data services without the burden of installing new transmission lines (e.g., for Internet access). DSL service dramatically evolved in the mid 1990s due to the availability of new modulation technology and low cost electronic circuits that can do advanced signal processing (e.g., echo canceling and multiple channel demodulation). This has increased the data transmission capability of twisted pair copper wire to over 50 Mbps. The data transmission capability of a DSL system varies based on the distance of the cable, type of cable used, and modulation technology. There are several different DSL technologies. Each of the DSL technologies mixes different types of transmission technologies to satisfy a specific business need. Some DSL systems allow simultaneous digital and analog transmission and are compatible with analog POTS systems.

Figure 4.24 shows a basic DSL system. This diagram shows that the key to DSL technologies is a more efficient use of the 1 MHz of bandwidth available on a single pair of copper telephone lines. A DSL system consists of compatible modems on each end of the local loop. For some systems, the DSL system allows for multiple types of transmission on a single copper pair. This includes analog or ISDN telephone (e.g., POTS) and digital communications (ADSL or VDSL). This diagram shows that there are basic trade offs for DSL systems. Generally, the longer the distance of the copper line, the lower the data rate. Distances of less than 1,000 feet can achieve data rates of over 50 Mbps.

Figure 4.24: Basic Digital Subscriber Line (DSL) System The first digital subscriber lines (DSLs) were developed due to the need for cost effective quality communication over copper wire. The first digital transmission system was the T1 line. This system had a maximum distance of approximately 6,000 feet prior to needing repeaters. The T1 digital transmission system used a very complex form of digital transmission. A new highspeed digital subscriber line technology was developed to replace T1 transmission technology. HDSL systems increased the distance that high-speed digital signals could be transmitted without the user of a repeater/amplifier. The HDSL system did require 2 (or 3) pairs of wires to allow simultaneous (send and receive) up to 2 Mbps of data transmission. To conserve the number of copper pairs for data transmission, symmetrical digital subscriber line (SDSL) technology was developed. Although SDSL systems offered lower data rates than HDSL, only 2 wire pairs were required. Since SDSL was developed, the HDSL system has evolved to a 2nd generation (HDSL2) that allows the use of 2 wire pair for duplex transmission with reduced emissions (lower egress). New efficient modulation technology used by ADSL systems dramatically increased the data transmission rates from the central office to the customer to over 6 Mbps (some ADSL systems to 8 Mbps). To take advantage of integrated services digital network (ISDN) equipment and efficiency, an offshoot of ISDN technology that was adapted for the local loop was developed. This technology called ISDN digital subscriber line (IDSL). Asymmetric digital subscriber line (ADSL) systems evolved to rate adaptive digital subscriber line (RADSL) allow the data rate to be automatically or manually changed by the service provider. To simplify the installation of consumer based DSL equipment, and low data transmission offshoot of ADSL developed that is called ADSL-Lite. Using similar technology as the ADSL system, very highspeed digital subscriber line (VDSL) was created to provide up to 52 Mbps data transfer rates over very short distances. Figure 4.25 shows the evolution of DSL systems. This diagram shows that high-speed digital subscriber line technology has been readily available since the 1970s. In the late 1990’s, the addition of advanced signal processing technology allowed DSL technology to rapidly increase transmission speed to over 50 Mbps in short distances.

Figure 4.25: Evolution of DSL

High Bit Rate Digital Subscriber Line (HDSL)
High bit rate digital subscriber line is an all digital transmission technology that is used on 2 or 3 pairs of copper wires that can deliver T1 or E1 data transmission speeds. HDSL is a symmetrical service. Figure 4.26 shows that the first application for HDSL used two pairs (and sometimes 3 pairs) of copper wire. Each circuit has an HDSL Termination Unit (HTU) on each end; an HTU-C (central office) and HTU-R (remote). This example shows that each pair of HDSL wires carries 784 kbps full duplex (simultaneous send and receive) data transmission. To carry the equivalent of a T1 line, two pairs of lines are used. It is also possible to carry an E1 line by using 3 pairs of copper wire. Although the framed transport for HDSL is different than for a T1 or E1 line, the HTU-C and HTU-R convert the protocols to standard T1 lines.

Figure 4.26: High bit rate Digital Subscribe Line (HDSL) System

Asymmetric Digital Subscriber Line (ADSL)
Asymmetrical digital subscriber line (ADSL) is a communication system that transfers both analog and digital information on a copper wire pair. The analog information can be a standard POTS or ISDN signal. The maximum downstream digital transmission rate (data rate to the end user) can vary from 1.5 Mbps to 9 Mbps downstream and the maximum upstream digital transmission rate (from the customer to the network) varies from 16 kbps to approximately 800 kbps. The data

transmission rate varies depending on distance, line distortion and settings from the ADSL service provider. Figure 4.27 shows that a typical ADSL system can allow a single copper access line (twisted pair) to be connected to different networks. These include the public switched telephone network (PSTN) and the data communications network (usually the Internet or media server). The ability of ADSL systems to combine and separate low frequency signal (POTS or IDSN) is made possible through the use of a splitter. The splitter is composed of two frequency filters; one for low pass and one for high pass. The DSL modems are ADSL transceiver unit at the central office (ATU-C) and the ADSL transceiver unit at the remote home or business (ATU-R). The digital subscriber line access module (DSLAM) is connected to the access line via the main distribution frame (MDF). The MDF is the termination point of copper access lines that connect end users to the central office.

Figure 4.27: Asymetric Digital Subscriber Line (ADSL) System

ADSL-Lite is a limited version of the standard ADSL transmission system. This limited version of ADSL allows for a simpler filter installation that can often be performed by the end user. The limitation of ADSL-Lite is a reduced data transmission rate of 1 Mbps instead of a maximum rate of 8 Mbps. Figure 4.28 shows that an ADSL-lite system is similar to the ADSL network with the primary difference in how the end user equipment is connected to the telephone network. The ADSL-lite system does not require a splitter for the home or business. Instead, the end user can install microfilters between the telephone line and standard telephones. These microfilters block the high speed data signal from interfering with standard telephone equipment. The ADSL-Lite end user modem contains a filter to block out the analog signals.

Figure 4.28: ADSL Lite System

Very High Bit Rate Digital Subscriber Line (VDSL)
Very high bit rate Digital Subscriber Line (VDSL) is a communication system that transfers both analog and digital information on a copper wire pair. The analog information can be a standard POTS or ISDN signal and the typical downstream digital transmission rate (data rate to the end user) can vary from 13 Mbps to 52 Mbps downstream and the maximum upstream digital transmission rate (from the customer to the network) can be 26 Mbps. The data transmission rate varies depending on distance, line distortion and settings from the VDSL service provider. The maximum practical distance limitation for VDSL transmission is approximately 4,500 feet (~1,500 meters). However, to achieve 52 Mbps, the maximum transmission length is approximately 1,000 feet (~300 meters). Figure 4.29 shows how a VDSL system is commonly used with a fiber distribution network that reaches a neighborhood or small group of buildings. The fiber terminates in an optical network unit (ONU). The ONU converts the optical signal into an electrical signal that can be used by the VDSL modem in the DSLAM. The DSL modem signal is supplied to a splitter that combines the analog and digital signal to copper access line. The splitter is actually attached to the last few hundred feet of the copper access line. The figure shows that the analog POTS signal from the local telephone company may still travel thousands of feet back to the central office. At the customers’ premises, the VDSL signal arrives to a splitter that separates the analog signal from the high-speed digital VDSL signal. Because VDSL has a much higher data transfer rate, the CPE may include a digital video set top box that allows for digital television.

Figure 4.29: Very high bit rate Digital Subscriber Line (VDSL) System

ISDN Digital Subscriber Line (IDSL)
ISDN Digital Subscriber Line (IDSL) is a hybrid of ISDN and DSL technologies. It uses the same data formatting as ISDN devices on the copper wire pair and delivers up to 144 kilobits per second bandwidth through two 64 kbps channels and one 16 kbps channel. The key difference for IDSL systems is that the IDSL system only uses the 64 kbps DS0 channels and the ISDN control channel (D channel) is ignored. The IDSL system effectively multiplies the number of channels on a single copper pair by 2x. The ability to avoid using ISDN signaling is very important as software upgrades for switching systems, to allow ISDN operation can cost more than $500,000 per switch.

Integrated Digital Loop Carrier (IDLC)
Integrated digital loop carrier (DLC) is a digital transmission technology that is used between the central office and groups of customers. The IDLC system is composed of two primary parts: an integrated digital terminal (IDT) and a remote digital terminal (RDT). The IDT concentrates up to

96 lines on to a single 24 channel T1 line. It does this by assigning central office channels to time slots on the IDLC line (between the IDT and RDT) as needed. The RDT reverses the process by assigning a time slot to an access line. The RDT also changes the format of the time slot to the access technology of choice (e.g., ISDN or analog). The key advantages to the DLC carrier system is that some of the switching function is moved closer to the customer (in the RDT) and increased cost effective transmission through the increased sharing of local loop copper lines. Because the RDT in the DLC system acts as a repeater, this also extends the range of access lines from the central office to the end customer. Unfortunately, DLC systems are not transparent to 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. 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/demultiplexing, 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). Figure 4.31 shows how an integrated digital loop carrier (IDLC) system can be installed in a local telephone distribution network to allow a 24 channel T1 line to provide service to up to 96 telephone lines. This diagram shows that a switching system has been upgraded to include an IDT and a remote digital terminal (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 is a local switch that 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 4.31: Integrated Digital Loop Carrier (IDLC) System The most common multiplexer is the SLC (subscriber line carrier)-96. This system, routinely referred to as a “Slick96”, can deliver 96 voice circuits (the equivalent 4 T-1’s) to the customer site. This methodology overcomes distance problems associated with providing voice services to remote customers as well as free additional twisted pair for future use.

Chapter 5: Switching
Switching is the process of connecting two (or more) points together within a network or communication devices. These connections can be physically connected (mechanical switch) or connected logically (through software). The first telephone systems performed the switching of calls by human operators. The operators interconnected telephone lines by manually connecting cables at switchboards. Switching systems have evolved from manual switchboard systems (wires and plugs) to logical (digital) switches. The earlier types of manual switchboard systems were changed to automatic switching systems to eliminate the need for operators to setup every call. The first types of automatic switching systems used crossbar switches. Crossbar switches used mechanical arms to physically connect to wires (or busses) together. This has progressed to time slot interchange (TSI) switches. TSI switches logically interconnect communication lines through the temporary storage of data in memory time slots.

Circuit Switching
Circuit switching is a process of connecting two points in a communications network where the path (switching points) through the network remains fixed during the operation of a communications circuit. While a circuit switched connection is in operation, the capacity of the circuit remains constant regardless of the amount of content (e.g. voice or data signal) that is transferred during the circuit connection. Figure 5.1 shows how circuit switching is used for voice communication. In this example, a telephone is dialing a telephone that is connected to a distant switch. When the user dials the telephone, the dialed digits are captured and used to program the circuit switches between the two telephones. Each switch then has assigned input ports and output ports and each switch only adds a small amount of transfer time between ports. After the all the switching connections are made, an audio path can be connected between. Throughout the connection, this path will be maintained through the initial path (the same switch ports) without any changes.

Figure 5.1: Circuit Switching Operation

Circuit Switched Data
Circuit switched data is a data communication method that maintains a dedicated communications path between two communication devices regardless of the amount of data that is sent between the devices. This gives to communications equipment the exclusive use of the circuit that connects them, even when the circuit is momentarily idle. To establish a circuit-switched data connection, the address is sent first and a connection (possibly a virtual non-physical connection) path is established. After this path is setup, data is

continually transferred using this path until the path is disconnected by request from the sender or receiver of data. Figure 5.2 shows a circuit switched data system. In this figure, a computer is sending a data file through a circuit switched data communications network to a home office computer. To start the data file transfer, the computer sends the destination address (address to the data network. The destination address is used to program the switches between the points on which ports they will receive the data and which ports they will send the data. As soon as all the switching connections are made, the computer can start sending data to the office. Throughout the connection, this path will be maintained through the initial path (the same switch ports) without any changes.

Figure 5.2: Circuit Switched Data Operation

Packet Switching
Packet switching is the transfer of information between two points through the division of the data into small packets. The packets are routed (switched) through the network and reconnected at the other end to recreate the original data. Each data packet contains the address of its destination. This allows each packet to take a different route through the network to reach its destination. Figure 5.3 shows the basic operation that uses packet switching. In this example, a laptop computer is sending a file to a company’s remote computer that is connected to a packet data network. The laptop computer data communication software requests the destination address for the packets for the user to connect to the remote computer ( In this example, the source computer divides the data file into three parts and adds the packet address to each of the 3 data packets. The packets are sent through routers in the packet network that independently determine the best path at the time that will help the packet reach its destination (smart switches). This diagram shows the three packets take 3 different routes to reach their destination. When the 3 packets reach their destination, the remote computer reassembles the data packets into the original data file.

Figure 5.3: Packet Switched Data Operation

Packet Assembler And Disassembler (PAD)
The PAD divides or converts blocks of data (such as data files) to and from small packets of information. In the disassembly process, a PAD usually assigns sequential numbers to the packets as they are created to allow the reassembly PAD to identify the correct sequence of data packets to reproduce the original data signal. Figure 5.4 shows a packet assembler and disassembler (PAD) system operation. This diagram shows that a large file is to be sent over a packet data network. The large file is supplied to the PAD circuit that divides the data file into smaller packets. These packets are sent toward their destination through a data communications network. When they are received, they are reassembled into the original large data file by a packet assembler.

Figure 5.4: Packet Assembler and Disassembler (PAD) Operation

Jitter Buffer
The jitter buffer receives and adds small amounts of delay to packets so that all the packets appear to have been received without varying delays. Jitter buffers allow for the smoothing out of digital audio signals that experience variable transmission delay across a network (such as the Internet). Figure 5.5 shows how a jitter filter can remove the variable transmission delay for packets that experience variable transmission time through a packet switched network. This diagram shows that packets are delayed variable amounts (delay 1-3). The jitter filter receives the packets and stores the packets in memory until a specific start time. The jitter filter has a clock that provides specific start times for the transmission of the pulse. This fixes the amount of delay to an anticipated maximum amount.

Figure 5.5: Jitter Filter Operation

Switching Systems

Switching systems connect two (or more) points together. These connections can be physically connected (mechanical switch) or connected logically (through software). The types of switching systems that are still in common use today include crossbar, time slot interchange (TSI), and packet switching.

Crossbar switches use mechanical arms to physically connect to wires (or busses) together. These mechanical arms (“Crossbars”) connect horizontal and vertical bars together to connect input and output lines together. Magnets are used to open and close the crossbar switch contacts. Figure 5.6 shows a crossbar switching system. In this example, there is a matrix of lines (busses) where each input line can be connected to any output line. When a connection needs to be made, a mechanical switch connects one of the busses with the other busses. The disadvantage of this system is that the number of mechanical switches for connecting each input port to an output port exponentially increases with the number of ports that require connection. For example, a switch with 10 inputs and 10 output lines requires 100 switches. A switch that has 20 inputs and 20 outputs requires 400 switches.

Figure 5.6: Crossbar Switching

Time Slot Interchange (TSI)
Time slot switching (TSI) is a process of connecting incoming and outgoing digital lines together through the use of temporary memory locations. A computer controls the assignment of these temporary locations so that a portion of an incoming line can be stored in temporary memory and retrieved for insertion to an outgoing line. Figure 5.7 shows a TSI switching system. This diagram shows a simplified matrix switching system. Each input line (port) is connected to a multiplexer. The multiplexer places data from each port in time sequence (time slot) on a communications line (e.g., a T1 or E1 line). This time multiplexed signal is supplied to a matrix switching assembly. The matrix switching assembly core

has two memory parts: a section that holds the pulse coded modulation (PCM) data and Control Memory - CRAM that holds switching addresses data.

Figure 5.7: Time Slot Interchange (TSI) Switching System The time slots (voice channels) from the incoming multiplexed line is sent through switch S1 to be sequentially stored in the PCM data memory. The data is later retrieved by switch S2 and placed on a specific time slot on an outgoing line. The outgoing multiplexed line is supplied to a demultiplexer so each time slot is routed to an output port.

Optical Switching
Optical switching is the process of directly connecting optical signals between multiple ports or time periods. This is done on an optical communication line without the need to convert the optical signals to electrical form. The use of optical switches reduces the transmission time delay and jitter that is experienced in the conversion of optical to electrical and electrical to optical switching systems.

Packet Switches
Packet transmission is a mode of data transmission that divides messages or data into small increments (packets) that can be routed through a network. When the packets arrive at their destination, they are reassembled in the proper order to recreate the original message or data. Packet switching can be connection based or connectionless. For connection based switching, a path through the network is established during call initiation and packets are continuously routed through the same path. For connectionless switching, each packet is given a destination address and the switching points in the network (switching nodes) assist in routing the packet to its destination. Figure 5.8 shows two types of packet switching in a communications system. Diagram (a) shows that connection based packet switching sets up a communication circuit prior to transmitting packets that contain data. Diagram (b) shows connectionless packet switching. Connectionless packet switching requires intelligent switching nodes (routers) that can decode the destination address and select the forwarding route based on the results of the lookup in the routing table. This diagram shows that packets of data arrive at the switch. The routing switch extracts the destination address and possibly the type of message.

Figure 5.8: Packet Switches

Data Routing
The key technologies that have emerged to enable data networks include hubs, bridges, routers, gateways, and firewalls.

A hub is a communication device that distributes communication to several devices in a network. This is accomplished through the re-broadcasting of data that it has received from one (or more) of the devices connected to it. A hub generally is a simple device that re-distributes data messages to multiple receivers. However, hubs can include switching functionality and multi-point routing connectivity and other advanced system control functions. Hubs can be passive or active. Passive hubs simply re-direct (re-broadcast) data it receives. Active hubs both receive and regenerate the data it receives. Figure 5.9 shows an Ethernet hub. This diagram shows that one of the computers has sent a data message to the hub on its transmit lines. The hub receives the data from the device and rebroadcasts the information on all of its transmit lines, including the line that the data was received on. The hub’s receiver and transmit lines are reversed from the computers. This allows the computers that are connected to the hub to hear the information on their receive lines. The sending computer uses the echo of its own information as confirmation the hub has successfully received and retransmitted its information. This indicates that no collision has occurred with other computers that may have transmitted information at the same time.

Figure 5.9: Hub

A bridge is a data communication device that connects two or more segments of data communication networks by forwarding packets between them. The bridge operates as a physical connector and buffer between similar types of networks.

Bridges extend the reach of the LAN from one segment to another. Bridges have memory that allows them to store and forward packets. Bridges are protocol independent as the do not perform protocol adaptations. Bridges contain a packet address-forwarding table (routing table) that they use to determine if the packets should be forwarded between networks. The packet-forwarding table contained within the bridge can be initially programmed or learned by the bridge. A self-learning bridge can monitor packet traffic in the network to continually update its packet-forwarding table Bridges primarily operate at the physical layer and link layers of the OSI reference model. A bridge receives packets from one network, review the address of the packet to determine if it should be routed to the other network(s) it is connected to, and retransmits the packet following the standard protocol rules for the systems it is connected to. Figure 5.10 shows the basic operation of a bridge that is connecting 3 segments of a LAN network. Segment 1 of the LAN has addresses 101 through 103, segment 2 of the LAN has addresses 201 through 203, and segment 3 of the LAN has addresses 301 through 303. The table contained in the bridge indicates the address ranges that should be forwarded to specific ports. This diagram shows a packet that is received from LAN segment 3 that contains the address 102 will be forwarded to LAN segment 1. When a data packet from computer 303 contains the address 301, the bridge will receive the packet but the bridge will ignore (not forward) the packet.

Figure 5.10: Bridge

A router is a device that directs (routes) data from one path to another in a network. Routers base their switching information on one or more information parameters of the data messages. These parameters may include availability of a transmission path or communications channel, destination address contained within a packet, maximum allowable amount of transmission delay a packet can accept, along with other key parameters. Routers that connect data paths between different types of networks are sometimes called gateways. Routers provide some of the same functionality as network switches. Their primary function is to provide a path for each routable packet to its destination. When a router is initially installed into a network, it begins its life by requesting a data network address. Using this data network address, it sends messages to nearby routers and begins to store address connections of routers that are located around it. Routers regularly exchange their connection information (lists of devices it is connected to) with nearby routers to help them keep the latest packet routing information.

A router can make decisions on where to forward packets dependent on a variety of factors including the maximum distance or packet priority. Distance vector routing and link state routing allow the router to select paths that match the needs of the data that is being sent through it. Routers may also have fixed routing tables that are manually programmed by the network administrator. These static routing tables may be inflexible, however the use of static routing ensures other router’s that may have corrupt routing tables does not change the table. Figure 5.11 shows a how a router can dynamically forward packets toward their destination. This diagram shows that a router contains a routing table (database) that dynamically changes. This diagram shows a router with address 100 is connected to two other routers with addresses 800 and 900. Each of these routers periodically exchanges information allowing them to build routing tables that allow them to forward packets they receive. This diagram shows that when router 100 receives a packet for a device number 952, it will forward the packet to router 900. Router 900 will then receive that packet and forward it on to another router that will help that packet reach its destination.

Figure 5.11: Router

Gateways are devices that enable information to be exchanged between two dissimilar computer systems or data networks. A gateway reformats data and protocols in such a way that the two systems or networks can communicate. Gateways can convert packets between dissimilar networks. Figure 5.12 shows how a gateway can convert large packets from a FDDI into very small packets in an ATM network. Not only does the gateway have to divide the packets, it must also convert the addresses and control messages into formats that can be understood on both networks.

Figure 5.12: Gateway

A firewall is a device or software program that runs on a computer that provides protection from external network intruders by inhibiting the transfer of unauthorized packets and by allowing through packets that meet safe criteria. There are various processes that can be used by firewalls to determine which packets are authorized and packets that should be rejected (not forwarded). Because firewalls can use many different types of analysis to determine packets that will be rejected, they can be complicated to setup. If a firewall is not setup correctly, it can cause problems for users that are sending and expected return packets that may be blocked by the firewall. Because firewalls process and analyze information, this process requires additional time and this can slow down network data transfer and response time. Figure 5.13 shows how a firewall works. This diagram shows that a user with address 201 is communicating through a firewall with address 301 to an external computer that is connected to the Internet with address 401. When user 201 sends a packet to the Internet requesting a communications session with computer 401, the packet first passes through the firewall and the firewall notes that computer 201 has requested a communication session, what the port number is, and sequence number of the packet. When packets are received back from computer 401, they are actually addressed to the firewall 301. Firewall 301 analyzes the address and other information in the data packet and determines that it is an expected response to the session computer 201 has initiated. Other packets that are received by the firewall that do not contain the correct session and sequence number will be rejected.

Figure 5.13: Firewall Firewalls are also appropriate for small office and home office (SOHO) applications. There are low-cost software packages and hardware equipment that offer a moderate level of increased security. They cannot stop all hackers, but they will stop some of them.

Virtual Circuits

A virtual circuit or virtual channel (VC) is a logical connection between two communication ports in one or more communication networks. There are two types of VC’s: permanent virtual circuits (PVC’s) and switched virtual circuits (SVC’s). A SVC is an automatically and temporarily created virtual connection that is used for a communication session. A PVC is a virtual circuit is manually created for a continuous communication connection. To create a PVC, routing tables in switches are manually configured one time to provide a continuous connection of end points through a network. The ability to dynamically or manually create virtual connections through a network has created a new type of network referred to as value-added networks (VAN’s). Rather than purchase leased circuits between corporate locations some companies chose to contract with VAN’s to transport their data between their sites. With leased data circuits from each corporate site to the closest VAN pointof-presence (POP), usually in the same metropolitan area, a company could establish PVC’s between its sites through the VAN’s network. These function similar to leased circuits and often provid a monthly cost saving to the corporation, yet deliver practically the same service as leased data circuits.

Permanent Virtual Circuits (PVC)
A permanent virtual circuit (PVC) is a logical connection path that is manually created through a network that provides a continuous communication connection. After a permanent communications circuit is established, a data path (logical connection) is maintained. Figure 5.14 shows how a permanent virtual circuit (PVC) is used to allow the transfer of data through a communications network through a pre-established logical (virtual) path. In this example, a PVC is created by programming routing tables in 4 switches before any data is sent. These routing tables assign data transfer connections between input and output channel on each switch. For example, as data from the sending computer (portable computer) is sent into input channel 3 of the first switch, it is transferred to the output channel 5. This process will repeat for any data that is sent from the sending computer to the destination computer. This example also shows that the PVC path remains active even if the portable computer is disconnected for a period of time.

Figure 5.14: Permanent Virtual Circuit (PVC) Operation

Switched Virtual Circuits (SVC)
A switched virtual circuit (SVC) is an automatically and temporarily created logical connection that is used for a communication session. Figure 5.15 shows how a switched virtual circuit (SVC) uses an address provided by the user to establish a logical (virtual) path through a communications network. In this example, a SVC is created by using the destination address to determine the required programming of routing tables in switches within the data network. These routing tables assign data transfer connections

between input and output channel on each switch. For example, as data from the sending computer (portable computer) is sent into input channel 3 of the first switch, it is transferred to the output channel 4. This process will repeat for any data that is sent from the sending computer to the destination computer during the switched connection.

Figure 5.15: Switched Virtual Circuit (SVC) Operation

Chapter 6: Signaling
Signaling is the process of transferring control information such as connection addresses, call supervision codes, or other connection information between communication switching equipment and other communications equipment or systems. The basic functions of signaling include initiate a call or line connection (call setup), maintain a communication link, and to end a call or connection (call teardown). Signaling comes in two basic forms: in-band signaling and out-ofband signaling. The control of public telecommunications networks is a centralized system as call processing is coordinated through a controlled common channel signaling (CCS) network. The Internet uses distributed control as the switching information dynamically changes in packet switching centers (routers) throughout the Internet network.

In-Band Signaling
In-band signaling sends control messages in the same communication channel that is used for voice or data communication. During the period of in-band signaling, the voice or data communication is temporarily inhibited (muted) to allow the transfer of control messages. The types of in-band address signaling include dial pulse (DP), dual-tone multi-frequency (DTMF), MF (Multi-Frequency), audio signaling, and line control. In-band signaling is sometimes called blank and burst signaling. Dial pulse (DP) signaling senses and counts the changes in current flow, such as from a rotary dial telephone, to allow the user to send address information (dialed digits) to the telephone system. DTMF signaling is a means of transferring information from a user to the telephone network through the use of in-band audio tones. Each digit of information is assigned a simultaneous combination of one of a lower group of frequencies and one of a higher group of frequencies to represent each digit or character. Multifrequency (MF) signaling is a type of in-band address signaling method that represents decimal digits and auxiliary signals by pairs of frequencies from the following group: 700, 900, 1100, 1300, 1500, and 1700 Hz. These audio frequencies are used to indicate telephone address digits, precedence, control signals, such as line-busy or trunk-busy signals, and other required signals. On modern telephone systems, most in-band signaling only occurs between the end-user and his serving central office telephone switch. These signals travel over the same audio line as the voice or data call. Examples of other these in-band signaling messages include: Dial tone (the circuit is working) Busy tone (the circuit is unavailable) Fast busy tone (the system is busy) DTMF or pulse digits (send dialed digit information) Special functions such as # and * (activate other services) Telephone systems also can sense line condition as a signaling method. When the central office senses a grounding of the line (ground start) or a reduction in voltage (off-hook loop-start), it produces a dialtone signal (audio signaling) to inform the user service is available. Wink start is

another line activation signal that is used by the telephone switch to indicate to end-user telephone systems of a change in status. Wink signals are brief 140 msec interruptions of communication. Figure 6.1 shows how the basic process of in band signaling is used in analog cellular radio to deliver control messages sharing the same communication channel for voice and control signals. In this diagram, a radio base station desires to send a message to the mobile radio. The base initially sends a dotting sequence that indicates a synchronization word and message will follow. The mobile radio detects the dotting sequence. As a result, the mobile radio mutes the audio and begins to look for a synchronization word. The synchronization word is used to determine the exact start of the message. The mobile radio receives the message and on completion of the message, the mobile radio will then un-mute the audio and conversation continues. Because the sending of the message can be less than ⁄ second, the user may not even notice a message has been received.

Figure 6.1: In-Band Signaling

Sub Band Signaling
Sub-band signaling sends information by using a frequency band that is located within the communication channel but outside the normal communication channel (e.g. audio band) bandwidth to transfer signaling control messages. Figure 6.2 shows how sub-band digital and audio signals are combined with standard audio. This diagram shows that sub-band signaling is a unique signaling feature used by some radio systems is the sub-band digital audio signaling. In many radio broadcast and mobile communication systems, an audio bandpass filter blocks the audio channel’s lower range. It is possible to combine a sub-band digital signaling channel (a low speed digital signal) using the lower frequency range (below 300 Hz).

Figure 6.2: Sub-band Signaling

Out-of-Band Signaling
Out-of-band signaling is signaling that travels over a separate path from voice and data calls but carries control information about the calls such as call setup, call routing, caller-id, call tear-down, etc. For out-of-band signaling the telecommunication industry uses a standard called Common Channel Signaling and Control (CCS). The current version of CCS is known as Signaling System 7 (SS7). Through standardization all telephone companies implement SS7 and thus can interact smoothly with very few errors. Using SS7, switches can more effectively route calls and even query centralized databases for additional control information. The advent of SS7 has brought with it many new features such as caller-id. It has also been instrumental allowing for the phenomenal growth the industry has seen. Figure 6.3 shows how out-of-band signaling occurs on a telephone network. This diagram shows that a signaling control message are sent on different channels than voice or data information (e.g. voice trunks). This diagram shows that these signaling messages are sent on their own network (packet switching network). Because all the control signals are sent on a common packet network, this is called common channel signaling.

Figure 6.3: Out of Band Signaling Operation

Line Side
Line side connections are an interconnection line between the customer’s equipment and the last switch (end office) in the telephone network. The line side connection isolates the customer’s equipment from network signaling requirements. Line side connections and are usually low capacity (one channel) lines. There are many types of signaling used on line side connections. They include rotary dialing, dual tone multi-frequency (DTMF), flash request, and ring signals.

Rotary Dial
Rotary dialing is a process of sending digital digit information through the user of a spring-loaded mechanical switch that produces pulses as it rotates through 10 positions (1 through 9, and 0). As the rotary dial turns, a switch briefly interrupts the loop current. The number of pulses per rotation is counted to determine the number dialed. A time pause between rotary dials is used to determine when additional digits are dialed or when the caller is finished dialing. Figure 6.4 shows how a mechanical rotary switch can be used to gather dialed digit information. In this example, a spring-loaded rotary dial is turned to produces pulses that represent the numbers on the dial. As the rotary dial turns, a contact switch briefly interrupts the loop current. The line card in the telephone system counts number of pulses to determine the number dialed. This diagram shows that after a specified time pause occurs after a series of pulses, the counter resets and is ready to count another dialed digit.

Figure 6.4: Rotary Dial Operation

Dual Tone Multi-Frequency (DTMF)
DTMF signaling is a means of transferring information from a user to the telephone network through the use of in-band audio tones. Each digit of information is assigned a simultaneous combination of one of a lower group of frequencies and one of a higher group of frequencies to represent each digit or character. There are 8 tones that are capable of producing 16 combinations; 0-9, *, #, A-D. The letters A-D are normally used for non-traditional systems (such as the military telephone systems). Figure 6.5 shows how dual tone multi-frequency (DTMF) tones can be used to send dialing information from a telephone to a telephone system. There are 8 different frequencies that can be combined to represent 16 keys. The keys A-D are not usually included on standard telephone sets. To represent each button, two tones are combined. In this example, the button 3 is pressed, followed by a pause, then button 2 is depressed. Button 3 is represented by the combined tones 1477 Hz and 697 Hz. Button 2 is represented by the combined tones 1336 Hz and 697 Hz. To determine if the user is finished dialing, a timer is used. When the user has stopped dialing, the digits can be sent to the call processing section of the telephone system to initiate the call.

Figure 6.5: DTMF Dial Operation

Flash Request
A system special service request feature that is used to indicate that a subscriber has a desire to recall a service function or to activate a custom calling feature (such as a call transfer request). A flash feature service request can be created when the user initiates a short on-hook interval or through the sending of a special service request message. The short on-hook interval is created by a momentary operation of the telephone switch hook, during a prolonged off-hook period. The special service request message can be sent by a button on a telephone (such as a PBX telephone) or by pressing the SEND key on a mobile telephone. Figure 6.6 shows how the flash signal (special service request) can be sent to the telephone system to activate additional call processing features. This example shows a flash feature is sent on an analog line by momentarily opening the current loop connection. When the loop current sensing circuit senses a brief open (no current flow) period, it creates a flash message that is sent to the call processing section of the telephone system. For digital telephones, the flash message is sent via a signaling message on the digital channel. This diagram shows that on an ISDN line, the flash message is sent on the D (signaling) channel.

Figure 6.6: Flash Operation

Ring Signal
A ring signal is an alternating high voltage signal that is sent to a telephone set to alert the user (usually by creating a ring sound) of an incoming call. In North America, the ring cadence is usually 2 seconds on and 4 seconds off. The ring signal may also include messages that operate at low voltage on a different frequency that includes caller identification information. Figure 6.7 shows that the ring signal is a high voltage alternating current signal that alerts the user of an incoming call. This diagram shows that this ring signal includes a caller identification message and the data is transferred by FSK modulation between the AC ring periods.

Figure 6.7: Ring Operation

Trunk Side
Trunk side connections are used to interconnect telephone network switching systems to each other. Trunk side connections are usually high capacity lines that interconnect switching systems. The common types of signaling systems that are used for trunk side connections include multifrequency signaling (MF), E&M signaling, and SS7 signaling.

Multifrequency Signaling (MF)
Multifrequency (MF) signaling is a type of in-band address signaling method that represents decimal digits and auxiliary signals by pairs of frequencies from the following group: 700, 900, 1100, 1300, 1500 and 1700 Hz. These audio frequencies are used to indicate telephone address digits, precedence, control signals, such as line-busy or trunk-busy signals, and other required signals. Figure 6.8 shows how 5 different multi-frequency (MF) tones are used to send signaling (dialing digit) information on a communication line between end office (EO) switches. This diagram shows that the MF system uses 5 different frequencies that are combined to represent 10 keys; 700, 900, 1100, 1300, and 1500 Hz.

Figure 6.8: Multifrequency Signaling (MF) In this example, dialed digits are gathered from a telephone that is connected to the end office switch. A MF tone generator converts these digits to tones that are sent on a trunk line between end office switches. The receiving end office switch uses an MF receiver to convert the tones back to the digit information. This allows the receiving end office switch to determine which telephone (switch port) to connect to the communication trunk line.

E&M Signaling
Ear and Mouth (E&M) signaling is a method of communication trunk signaling over a path separate line from the transmission path. The two leads of this separate path are called the ear (receive) and the mouth (transmit). As a memory aid, the leads are analogous, direction wise, to an ear and a mouth. E&M signaling also is used in special services applications. (See also: loop signaling.)

Although E&M signaling is commonly refer to as “ear & mouth” or “recEive and transMit”, its origin actually comes from the term earth and magnet. Earth represents electrical ground and magnet represents the electromagnet used to generate tone. E&M signaling defines a trunk circuit side and a signaling unit side for each connection similar to the data circuit-terminating equipment (DCE) and data terminal equipment (DTE) reference type. Usually the PBX is the trunk circuit side and the telco, End Office (EO), or channel-bank is the signaling unit side. Figure 6.9 shows how E & M signaling uses separate lines to send control messages between switching systems. In this example, two PBX systems are connected together using voice lines and E & M signaling lines. When a PBX system desires to use a communication line, it changes the voltage of the M line to alert the other PBX’s E line. When the other PBX detects this change of state, additional control messages may follow to allow the switching units to communicate with each other and route calls to the correct extension or port.

Figure 6.9: E&M Signaling

Common Channel Signaling System 7 (SS7)
Common channel signaling system #7 (“SS7”) is the primary system used for the control of telephone systems. SS7 sends packets of control information between switching systems. Worldwide telephone networks are undergoing significant changes as methods of call processing and network management are altered to provide new services and to streamline operations. These changes are driven by user demand for enhanced services and the corresponding efforts of telephone operating companies to satisfy current and future needs. Enhanced services require bi-directional signaling capabilities, flexibility of call setup, and remote database access. Earlier signaling systems lacked the sophistication required to deliver much more than Plain Old Telephone Service(POTS). These traditional systems use dial pulses and multi-frequency (MF) tones to transmit call and circuit-related information such as dialed digits and circuit busy/idle states. The complexity of adding new functionality to traditional signaling systems meant that a new network signaling architecture was needed. SS7 was developed to satisfy the telephone operating companies’ requirements for an improvement to existing signaling systems. Common channel signaling (CCS) is a separate signaling system that separates content of telephone calls from the information used to set up the call (signaling information). When callprocessing information is separated from the communication channel, it is called “out-of-band” signaling. This signaling method uses one of the channels on a multi-channel network for the control, accounting, and management of traffic on all of the channels of the network. Because the public telephone network uses common channel signaling, intelligence in the network can be distributed to databases and information processing points throughout the network. A set of service development tools has been developed to allow companies to offer advanced intelligent network (AIN) services.

An SS7 network is composed of service switching points (SSPs), signaling transfer points (STPs), and service control points (SCPs). The SSP gathers the analog signaling information from the local line in the network (end point) and converts the information into an SS7 message. These messages are transferred into the SS7 network to STPs that transfer the packet closer to its destination. When special processing of the message is required (such as rerouting a call to a call forwarding number), the STP routes the message to a SCP. The SCP is a database that can use the incoming message to determine other numbers and features that are associated with this particular call. Figure 6.10 shows the basic structure of the SS7 control signaling system. The SS7 network is composed of its own data packet switches, and these switching facilities are called signal transfer points (STPs). In some cases, when advanced intelligent network services are provided, STPs may communicate with signal control points (SCPs) to process advanced telephone services. STPs are the telephone network switching point that route control messages to other switching points. SCPs are databases that allow messages to be processed as they pass through the network (such as calling card information or call forwarding information).

Figure 6.10: SS7 Common Channel Signaling In the SS7 protocol, an address, such as customer-dialed digits, does not contain explicit information to enable routing in a signaling network. It then will require the signaling connection control part (SCCP) translation function. This is a process in the SS7 system that uses a routing tables to convert an address (usually a telephone number) into the actual destination address (forwarding telephone number) or into the address of a service control point (database) that contains the customer data needed to process a call. Signaling links are logically organized by link type (“A” through “F”) according to their use in the SS7 signaling network. Figure 6.11 shows the relationship between the link names and the link location (type):

Figure 6.11: SS7 Signaling Link Types

Access Link (A Link)
An “A” (access) link connects a signaling end point (e.g., an SCP or SSP) to an STP. Only messages originating from or destined to the signaling end point are transmitted on an “A” link.

Bridge Link (B Link)
A “B” (bridge) link connects an STP to another STP. Typically, a quad of “B” links interconnect peer (or primary) STPs (e.g., the STPs from one network to the STPs of another network). The distinction between a “B” link and a “D” link is rather arbitrary. For this reason, such links may be referred to as “B/D” links.

Cross Link (C Link)
A “C” (cross) link connects STPs performing identical functions into a mated pair. A “C” link is used only when an STP has no other route available to a destination signaling point due to link failure(s). Note that SCPs may also be deployed in pairs to improve reliability; unlike STPs, however, mated SCPs are not interconnected by signaling links.

Diagonal Link (D Link)
A “D” (diagonal) link connects a secondary (e.g., local or regional) STP pair to a primary (e.g., inter-network gateway) STP pair in a quad-link configuration. Secondary STPs within the same network are connected via a quad of “D” links. The distinction between a “B” link and a “D” link is rather arbitrary. For this reason, such links may be referred to as “B/D” links.

Extended Link (E Link)
An “E” (extended) link connects an SSP to an alternate STP. “E” links provide an alternate signaling path if an SSP’s “home” STP cannot be reached via an “A” link. “E” links are not usually provisioned unless the benefit of a marginally higher degree of reliability justifies the added expense.

Fully Associated Link (F Link)
An “F” (fully associated) link connects two signaling end points (i.e., SSPs and SCPs). “F” links are not usually used in networks with STPs. In networks without STPs, “F” links directly connect signaling points.

Service Switching Points (SSP)

Service Switching Points (SSP) are telephone switches interconnected by SS7 links. The SSPs perform call processing on calls that originate, tandem, or terminate at that site. As part of this call processing, the SSP may generate SS7 messages to transfer call-related information to other SSPs, or to send a query to a Service Control Point for instructions on how to route a call.

Signal Transfer Points (STP)
Signal Transfer Points (STP) are switches that relay messages between network switches and databases. Their main function is to route SS7 messages to the correct outgoing signaling link, based on information contained in the SS7 message address fields.

Service Control Points (SCP)
Service Control Points (SCP) contain centralized network databases for providing enhanced services. The SCP accepts queries from an SSP and returns the requested information to the originator of the query. For example, enhanced 800 service uses an SCP database to determine the routing on 800 calls. When an 800 call is initiated by a user, the originating SSP sends a query to an 800 database requesting information to the SSP originating the query and the call proceeds.

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 [1]. 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 6.12 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 6.12: Advanced Intelligent Network (AIN)

Chapter 7: Protocols
Protocols are the precise set of rules and a syntax that govern the accurate transfer of information within a communications network. Protocols are used within a communication system to establish, carry out, and terminate communication circuits. Protocols are also used to coordinate billing and customer care systems, manage network devices, and any other process that requires coordinated communication and control. There are thousands of different protocols used in communications systems. Usually, protocols are grouped into families of protocols so they can serve specific types of networks and services. When interconnecting different networks, protocols need to be converted. Protocol conversion involves the translation of the protocols of one system to those of another to enable different types of equipment, such as data terminals and computers, to communicate. This is done by an inter-working function (IWF). An IWF system (such as a data bridge) adapts the communications between two different types of networks. Protocol conversion may be used to interconnect circuit switched or packet switched networks. Access protocols are the set of rules that workstations use to avoid collisions when sending information over shared network media. Access protocols are also known as the media access control (MAC) protocols. Handshaking protocols involve the sequence of events that occur between communication devices that negotiate the data transmission rules and ensure reliable data transmission. When data devices begin to communicate, they discover the capabilities and agree on a common set of protocols to use during data communications session. Line discipline is the sequence of events that must occur to control the reception of data, perform error detection and correction, and multiplexing of control information, if necessary. Session protocols control the end-to-end connectivity of a data communication session. Session protocols ensure all the data is received and in the correct order. Different protocols may be used in systems that provide similar functions. An example of this is token ring and Ethernet. Although these networks may actually use the same signaling system, they use incompatible protocols. To allow data to transfer between these networks, protocol converters are used. Protocol converts receive data and control messages, reformat data and convert control messages, and retransmit the data using the new protocol rules.

Protocol Stack
A protocol stack is a hierarchical structure of information processing functions that are logically separated into layers that theoretically only interact with higher and lower functional layers. The use of a protocol stack allows software programs and devices to be independently created (such as from different manufacturers) that only provide parts of the overall operation. Protocol stacks can be proprietary (owned exclusively by a company or group of companies) or protocol stacks can be created as an industry standard.

Open Systems Interconnection (OSI)
A generic form of communication protocol layers is open systems interconnection (OSI). OSI layers were developed by the International Standards Organization (ISC)) and the CCHT to facilitate the open inter-connection of computers and data terminals to their applications, regardless of their type or manufacturer. The protocols specify seven layers: physical, link,

network, transport, session, presentation, and application. Each layer performs specific functions for data exchange and is independent of the other layers. Figure 7.1 shows the seven layers of the open systems interconnection (OSI) model and how they interact with each other. This example shows how an email application can use the OSI model to allow communication between an email client (user that is checking email) to an email server (computer providing the email information) independent of who controls each layer, provided the interfaces between each layer are specifically defined. This diagram shows that the application layer is the interface to the user that permits the user to request delivery of their email. The application layer presents this request to the transport layer as a data file. The data file is divided up into smaller blocks of data and presented to the session layer. The session layer determines a new session is required (communication link) between the client and the server and this session information is passed on to the transport layer that will oversee the transfer of data during the session. The transport layer sends the destination address of the email server to the network layer. The network layer sends this information to the data link layer that establishes and maintains a data link connection to the network. The data link layer sends information to the physical layer that converts to data signals to either radio, electrical, or optical formats suitable for transmission.

Figure 7.1: Open System Interconnection (OSI) Layers

Protocol Suites
Protocol suite is a combination of network protocols that have designed to interoperate with each other to provide a common data communication language that is used on a network or communication system. One of the more common protocol suites is Internet protocol. The Internet protocol suite is a combination of network protocols that have designed to interoperate with each other to provide a common data communication language that is used on the Internet. The layers of protocol suite include physical layer, network (or routing) layer, transport (or session) layer, and application layer. The Internet protocol suite is overseen by the Internet Engineering Task Force (IETF). Key protocols included in the Internet Protocol Suite include Internet Protocol (IP), Transaction Capabilities Protocol (TCP), and User Datagram Protocol (UDP). There are many other protocols that are part of the Internet Protocol suite.

Parameter Negotiation (PN)
Parameter negotiation is the process of requesting and agreeing on the preferred characteristics for a communication session. Parameter negotiation is necessary to allow communication sessions to use the same protocol language and characteristic settings (such as data transmission rate). Figure 7.2 shows how two data communication devices negotiate for data transmission rates and protocols selection in a data network using the preferences assigned by a user along with the options determined by equipment availability. In this example, data terminal 1 sends a connection request message to data terminal 2. This connection request indicates that the data terminal prefers to use a 56 kbps data transmission rate because it has enough bandwidth. Unfortunately, the data terminal cannot accept the request for 56 kbps because it’s access bandwidth is low speed (28 kbps). The receiving data terminal sends back a request to use 28.8 kbps data transmission rate. When the originating data terminal receives this request, it accepts the request because has that data transmission rate and protocol capability available. It then confirms the request and both devices use a data transmission rate of 28.8 kbps.

Figure 7.2: Parameter Negotiation (PN) Operation

Internet Protocol (IP)
Internet protocol (IP) is a low-level network protocol that is used for the addressing and routing of packets through data networks. IP is the common language of the Internet. The IP protocol only has routing information and no data confirmation rules. To ensure reliable data transfer using IP protocols, higher level protocols such as transmission control protocol (TCP) are used. The basic IP protocol defines the packet datagram that hold packet delivery addressing, type of service specification, dividing and re-assembly of long data files and data security. IP protocol structure is usually combined with high-level transmission control protocols such as transaction control protocol (TCP/IP) or user datagram protocol (UDP/IP).

User Datagram Protocol (UDP)
UDP is a high-level communication protocol that coordinates the one-way transmission of data in a packet data network. The UDP protocol coordinates the division of files or blocks of data information into packets and adds sequence information to the packets that are transmitted during a communication session using Internet protocol (IP) addressing. This allows the receiving

end to receive and re-sequence the packets to recreate the original data file or block of data that was transmitted. UDP adds a small amount of overhead (control data) to each packet relative to other high-level protocols such as TCP. However, UDP does not provide any guarantees to data delivery through the network. UDP protocol is defined in request for comments 768 (RFC 768). Figure 7.3 shows how user datagram protocol (UDP) operates to efficiently send data through a packet network. This diagram shows that the UDP system first packetizes (divides) the sender’s data into smaller packets of data (maximum 1500 bytes). Each of these packets starts with an IP header that contains the destination address of the packet. The UDP system then adds a second header (the UDP control header) that includes a destination port. The packets are sent through the system where they may be received or lost in transmission. Because the UDP protocol does not contain any guarantee of delivery, it is up to the user on how to handle lost packets of data.

Figure 7.3: User Datagram Protocol (UDP) Operation

Transmission Control Protocol (TCP)
Transmission control protocol (TCP) is a session layer protocol that coordinates the transmission, reception, and retransmission of packets in a data network to ensure reliable (confirmed) communication. The TCP protocol coordinates the division of data information into packets, adds sequence and flow control information to the packets, and coordinates the confirmation and retransmission of packets that are lost during a communication session. TCP utilizes Internet Protocol (IP) as the network layer protocol. Figure 7.4 shows how transaction control protocol (TCP) operates to reliably send data through a packet network. This diagram shows that the TCP system receives the data from a specific communication port (port number). The TCP system then packetizes (divides) the sender’s data into smaller packets of data (maximum 1500 bytes). Each of these packets starts with an IP header that contains the destination address of the packet. The TCP system then adds a second header (the TCP control header) that includes a sequence number along with other flow control information. The packets are sent through the system where they may be received at different time periods. The sequence numbers can be used to reorder the packets. The TCP protocol also includes a window size that indicates to the receiving device how many packets it can receive before it must acknowledge their receipt. This window defines how much data the sending device must keep in temporary memory to enable the retransmission of a packet in the event that a packet is lost in transmission. If a packet is lost, the receiving device requests the transmitting device to re-send the packet with a specific sequence number.

Figure 7.4: Transmission Control Protocol (TCP) Operation

Real Time Transport Protocol (RTP)
Real time transport protocol (RTP) is a packet based communication protocol that adds timing and sequence information to each packet to allow the reassembly of packets to reproduce real time audio and video information. RTP is the transport used in VoIP environments. Figure 7.5 shows how real time transmission control protocol (RTP) operates to send real time data through a packet network that may have variable transmission delays. This diagram shows that an RTP system requires that a real time signal (e.g. audio signal) be converted to digital form (digital audio) prior to transmission. This digital signal is divided into small packets. The RTP protocol is a high-level protocol and each packet of data each of the transmitted packets starts with an IP header that contains the destination address of the packet. An additional flow control protocol header is added (usually UDP protocol header) to identify the specific port the data will be routed to at it’s destination. The RTP system then adds a third header (the RTP control header). The RTP system uses a precise clock to add time stamp information to each packet along with other signal recreation control information. Because the packets may have different types of compression and their recreation time can dramatically vary, the RTP protocol header uses the time stamp and other information to decode the and recreate the data packet.

Figure 7.5: Real Time Transport Protocol (RTP) Operation

Chapter 8: Networks
Networks are a series of points interconnected by communications channels. Networks can have different interconnection configurations. Data networks are designed specifically for the transmission of data information while other networks are better suited to transfer voice. Networks are composed of components that process and transfer information. Network elements or components can be standardized and combined to form larger networks that can be interconnected.

Network Configuration
Networks can have various configurations including star networks, ring networks, and bus networks.

Star Network
A star network interconnects each communication device (typically computers) to a central node in the network. At the center of the star network is a hub or concentrator that routes data to any of the point (nodes) that are connected to it. In a star network, each node requires its own individual cable. Figure 8.1 shows a star topology network. Each node in this network is connected directly to a central node. Each station must communicate with or through the central node (usually a hub) to reach other stations in the network.

Figure 8.1: Star Network Configuration

Ring Network
A ring network connects each communication device (typically computers) to a neighboring computer and these interconnected computers form a ring where the last computer in the string is connected to the first computer in the network. The data transmission process in a ring network involves computers passing all network data from its neighboring computer to the next

neighboring computer. When a computer in the ring network receives data, it looks for data information designated for its address and removes (does not retransmit) the data. Figure 8.2 shows a ring network. In this diagram, each data communication device receives information from one computer in the ring and sends (and possibly forwards received information) information to another computer in the ring.

Figure 8.2: Ring Network Configuration

Bus Network
A bus network connects each communication device (typically computers) to a common data transmission bus. Each computer that is connected to the bus network uses a transmission control process to sense availability and contention (simultaneous access) on the network. Figure 8.3 shows a bus network. This diagram shows that data communication devices can hear each other on the common (shared) bus.

Figure 8.3: Bus Network Configuration

Network Technologies
Networks are systems that transfer information or data between network access points (nodes) through data switching, system control, and interconnection transmission lines. Some of the more common network technologies used in telecommunication systems include ISDN, X.25 packet, frame relay, ATM, passive optical networks (PONs), local area data networks, and the Internet. These network technologies are the building blocks for communication systems such as PSTN, private telephone systems, mobile telephone, and satellite systems.

Integrated Services Digital 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 for North America and 2.048 Mbps and is divided into 30B + 2D for the rest of the world. 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. Figure 8.4 shows 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. Network termination 1 (NT1) equipment devices can directly connect to the NT1 connection. Devices that require other standards (such as POTS or data modems) require a terminal adapter (TA). This example shows that the NT2 interface works with the NT1 interface to allow the application layers (terminal intelligence) to communicate with the ISDN termination equipment.

Figure 8.4: Integrated Services Digital Network (ISDN) System

X.25 Packet
X.25 packet is an international standard for reliable data communications through the use of a packet-data switching network. The X.25 standard specifies the protocol between the data device (such as a computer) and the network such as a public packet data network (PDN). The X.25 system is a connection based packet switching system. X.25 packet data switches are initially programmed to create a logical path (virtual connection) from the entry point to the exit point before data transmission begins. X.25 systems are used to ensure reliable data transmission as it uses advanced error protection and retransmission processes. To provide this reliable transmission of packets of data, each link in the packet data network receives, checks, requests retransmission if necessary, and forwards the data onto the next link. The key components of a X.25 system are packet assembler and disassemblers (PAD) and packet nodes (packet switching points). The PAD divides or converts blocks of data (such as data

files) to and from small packets of information. In the disassembly process, a PAD usually assigns sequential numbers to the packets as they are created to allow the reassembly PAD to identify the correct sequence of data packets to reproduce the original data signal. The ITU specification for a X.25 system PAD is X.3. A packet node is a packet switch in an X.25 network. The packet node receives and forwards packets of data. The packet switch receives the packet of data, reads its address, searches in its database for its forwarding address, and sends the packet toward its next destination. X.25 systems are public data network (PDN) or private data systems. The X.25 specification only defines the communication with the X.25 network. Communication within the X.25 network is often implementation specific (company proprietary). To interconnect X.25 systems together, the X.75 specification is used. Because of the error checking and retransmission process used in the X.25 system, packet transmission time is generally longer than in newer packet switching systems such as frame relay. In a packet network, packet switches are networked together over a wide area (normally a country or continent). Packet switches are connected to each other via dedicated high-speed communication lines. Each switch is configured to have at least two leased circuits to at least two different switches. The local switch is in turn connected to local hosts via dedicated, leased lines and to multiple modems (modem banks) to allow local dial up access. The switches are constantly programmed with remote host addresses and the least cost routes to those devices. Figure 8.5 shows a X.25 packet data network. This diagram shows bank teller machine in Rome is connected to a bank processing system in London. The X.25 system is setup so a virtual path is created through the X.25 network so data can reliably pass through each packet node to reach its previously established destination. This diagram shows that a virtual connection is made through a packet node in Paris. Each packet that is sent is validated over each link until it reaches its destination.

Figure 8.5: X.25 Packet Data

Frame Relay
Frame relay is a packet-switching technology provides dynamic bandwidth assignment. Implementations of frame relay in 2002 allowed for dynamic bandwidth allocation up to 45 Mbps. Frame relay systems offer dynamic data transmission rates through the use of varying frame sizes. The frame relay system is a connection based switching system. Switches are initially programmed to create a logical path (virtual connection) from the entry point to the exit point. Frame relay systems are a simple bearer (transport only) technology and do not offer advanced error protection or retransmission. This reduces the time for packet switching (reduced transmission delay time). It is up to the sender and receiver of frame relay data to ensure the integrity of the data. When used in systems that have good digital communication systems, frame relay provides reliable data communication service. The first frame relay standard I.122 was defined in 1988 by the International Telecommunications Union (ITU). The current frame relay specification standards include the ITU I.233 and American National Standards Institute (ANSI) T1.606.

The key components of a frame relay system include frame relay access device (FRAD) frame relay network devices (FRND) and frame relay switches. The FRAD converts end user data into protocol data unit (PDU) variable length packets. The FRAD communicates to the FRND over an access line (e.g., fractional T1/E1 or ISDN line). The FRND is a packet switch that also operates as a gateway to the frame relay network. The FRND passes frames it receives from the FRAD to other frame relay switches that forward packets toward their destination network. Frame relay switches have buffer memory that allows them to hold and prioritize packets before they are retransmitted. Packet switches can selectively discard packets if network congestion occurs. The FRAD and FRND provide information about the priority of the frames (e.g., non-essential discard eligibility) and status of the system (e.g., network congestion notification). The frame relay system uses a discard eligibility (DE) flag system to indicate the essential nature of the packet’s data. The DE flag(s) allow systems to selectively discard data packets or frames that are non-essential. This process allows some data transmission systems to send more data than is agreed to (dynamic bandwidth). If the network is not congested, it may allow the extra packets of data to reach their destination. Congestion notification is a control flag signaling system that is used to indicate status of network congestion in a data network. Congestion notification allows data communication devices that are connected to the data network to send or delay the sending of data dependent on the status of the network. The frame relay system uses both forward and backward congestion notification. Forward explicit congestion notification (FECN) indicates to upstream switching devices that data that is being transmitted through congested switches and it is likely that some of the remaining packets may be discarded. The upstream switch can then change the data discard priority level accordingly. Backward explicit congestion notification (BECN) indicates to the sending (downstream) switching devices that congestion is occurring and packets that are received may be discarded. The sending switch can then change the priority of packet discarding and send and indication to other switches indicating network congestion. This should eventually reduce the amount of data endusers are sending into the network. The frame service provider usually agrees to provide the frame relay service at certain data transmission rate (service level). The frame relay system may provide a committed information rate (CIR) and a maximum burst information rate (BIR). A packet-switching technology provides dynamic bandwidth assignment. Frame relay systems are a simple bearer (transport only) technology and do not offer advanced error protection or retransmission. Frame relay were developed in the 1980s as a result of improved digital network transmission quality that reduced the need for error protection. Frame relay systems offer dynamic data transmission rates through the use of varying frame sizes. Figure 8.6 shows a frame relay network. This diagram shows a local area network (LAN) in San Francisco is connected to a LAN in New York. A virtual path is created through the frame relay network so data can rapidly pass through each frame relay switch as its path is previously established. When data is to be transferred through the LAN (e.g., a large image file), the data file passes through a FRAD that is the gateway to the frame relay network. The FRAD divides the data file from the LAN into variable length data frames. The FRAD sends and receives control commands to the frame relay network that allows the FRAD to know when and if additional data frames can be sent.

Figure 8.6: Frame Relay Network

Asynchronous Transfer Mode (ATM)
Asynchronous transfer mode (ATM) is a packet data transmission and switching system that transfers information by dividing all types of data into small fixed length packets of data (53 byte cells). The ATM system uses high-speed transmission (usually 155 Mbps or above) and is a connection-based system. When an ATM circuit is established, a patch through multiple switches is setup and remains in place until the connection is completed. ATM service was developed to allow one communication medium (high-speed packet data) to provide for voice, data, and video service. As of the 1990’s, ATM has become a standard for high-speed digital backbone networks. ATM networks are widely used by large telecommunications service providers to interconnect their network parts (e.g., DSLAMs and Routers). ATM aggregators operate networks that consolidate data traffic from multiple feeders (such as DSL lines and ISP links) to transport different types of media (voice, data, and video). The ATM switch rapidly transfers and routes packets to the pre-designated destinations. To transfer packets to their destination, each ATM switch maintains a database (called a routing table). The routing table instructs the ATM switch to which channel to transfer the incoming packet to and what priority should be given to the packet. The routing table is updated each time a connection is setup and disconnected. This allows the ATM switch to forward packets to the next ATM switch or destination point without spending much processing time. The ATM switch also may prioritize or discard packets that it receives based on network availability (congestion). The ATM switch determines the prioritization and discard options by the type of channels and packets within the channels that are being switched by the ATM switch. Figure 8.7, shows a functional diagram of an ATM packet switching system. This diagram shows that there are three signal sources going through an ATM network to different destinations. The audio signal source (signal 1) is a 64 kbps voice circuit. The data from the voice circuit is divided into short packets and sent to the ATM switch 1. ATM switch 1 looks in its routing table and determines the packet is destined for ATM switch 4 and ATM switch 4 adapts (slows down the transmission speed) and routes it to it destination voice circuit. The routing from ATM switch 1 to ATM switch 4 is accomplished by assigning the ATM packet a virtual circuit identifier (VCI) that ATM switch can understand (the packet routing address). This VCI code remains for the duration of the communication. The second signal source is a 384 kbps Internet session. ATM switch 1 determines the destination of these packets is ATM switch 4 through ATM switch 3. The third signal source is a 1 Mbps digital video signal from a digital video camera. ATM switch 1 determines this signal is destined for ATM switch 4 for a digital television. In this case, the communication path is through ATM switches 1, 2, and 4.

Figure 8.7: Asynchronous Transfer Mode (ATM) System

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 8.8 shows an ATM passive optical network (APON) system that locates optical network units (ONUs) near residential and business locations. This passive optical network routes different optical signals (different wavelengths) to different areas in the network by using optical splitters instead of switching devices. In this example, the optical distribution system 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 8.8: Passive Optical Network (PON) System

Data Networks
Data networks are primarily designed to transfer data from one point to one or more points (multipoint). Data networks may be composed of a variety of communication systems including: circuit switches, leased lines, and packet switching networks. There are predominately two types of data networks, broadcast and point-to-point. Broadcast networks allow a single information packet to be addressed to all destinations on the network at once. In this way, every single machine receives the packet and is capable of processing it. Certain types of broadcast networks can address a specific subset of the machines on the network. This is often called multicasting. Point-to-point networks consist of many individual machines that typically communicate in pairs. These types of networks must be able to support a large number of connections and the packets must travel between end-points by traversing through a number of intermediate machines. Data networks can be characterized as premises distribution networks (PDNs), Local area networks (LANs), metropolitan area networks (MANs), wide area networks (WANs), and wireless data networks (WDNs). Figure 8.9 shows the basic types of data networks. This diagram shows several types of local area networks (LANs) including Ethernet, Token Ring and FDDI. It also shows that small networks can be interconnected to form wide area networks. Data networks can be private networks or public networks. It is also possible to encrypt (protect) data information that is transmitted on public networks to form virtual networks.

Figure 8.9: Data Networks Some of the more popular data communication systems used include Ethernet, token ring, fiber distributed data interface (FDDI), asynchronous transfer mode 25 (ATM 25), phone line networking (HomePNA), USB, and Firewire.

Ethernet is a packet-switching transmission protocol that is primarily used in LANs. Ethernet is often characterized by its data transmission rate and type of transmission medium (e.g., twisted pair is T and fiber is F). Ethernet can be provided on twisted pair, coaxial cable, wireless, or fiber cable. In 2001, the common wired connections for Ethernet was 10 Mbps or 100 Mbps. 100 Mbps Ethernet (100 BaseT) systems are also called “Fast Ethernet.” Ethernet systems that can transmit at 1 Gbps (1 Gbps = 1 thousand Mbps) or more, are called “Gigabit Ethernet (GE).” Wireless Ethernet have data transmission rates that are usually limited from 2 Mbps to 11 Mbps. Wired Ethernet conforms to IEEE 802.3 standards and wireless Ethernet conforms to 802.11. IEEE 802.3 standard and uses carrier sense multiple access with collision detection (CSMA/CD) media access control (MAC). Because Ethernet systems can use different cabling systems (e.g., twisted pair and coax), network interface cards (NICs) must contain a connector that is compatible with the cabling systems. Some NIC cards come with multiple connectors. The different types of connectors include: – DB-15 AUI connector for thicknet, 10Base5 – BNC coaxial connector for thinnet, 10Base2 – RJ-45 for twisted pair, 10BaseT or 100BaseT. Figure 8.10 shows a typical Ethernet system. This diagram shows that this Ethernet system has 2 data rates; 100 Mbps and 10 Mbps. This system interconnects two different data rate Ethernet systems using a bridge. The bridge temporary stores (buffers) and retransmits Ethernet data frames when it recognizes the Ethernet addresses that are associated with the networks that it is connected to. It also connects the Ethernet systems to outside networks using a router. The router converts the Ethernet device address to other addresses (such as IP) that can be routed to other networks.

Figure 8.10: Ethernet System

Token Ring
Token ring is a LAN system developed by IBM that passes a token to each computer connected to the network. Holding of the token permits the computer to transmit data. The token ring specification is IEEE 802.5 and token ring data transmission speed range from 4 Mbps or 16 Mbps. 100 Mbps and higher token ring speeds are in development. Token ring networks are non-contention based systems, as each computer connected via the token ring network must have received and hold a token before it can transmit. This ensures computers will not transmit data at the same time. Token ring systems provide an efficient control system when many computers are interconnected with each other. This is the reason token ring systems will not see data traffic degradation when many new users are added compared to Ethernet systems. However, passing tokens does add overhead (additional control messages) that reduces the overall data transmission bandwidth of the system. Figure 8.11 shows a typical token ring LAN. This diagram shows that the network is logically setup in a ring and each computer in the token ring network must receive a token before it can transmit. Since the token is relatively small compared to the packets of data that are sent, the token can rapidly move from computer to computer. When a computer receives a token, it can transmit data for a limited amount of time before it is required to forward the token.

Figure 8.11: Token Ring System

Fiber Distributed Data Interface (FDDI)
Fiber distributed data interface (FDDI) is a computer network protocol that uses fiber optic cable as the transmission medium to provide high-speed data transmission service to LANs. FDDI is a token protocol. The basic transmission rate of FDDI is 100 Mbps. FDDI is commonly used as a backbone network that interconnects several LANs within a company. The FDDI specification is IEEE 802.2 and FDDI data transmission speed range from 100 Mbps to over 1 Gbps. FDDI is a LAN architecture that is based on redundant fiber rings that transmit in opposite directions. One of the rings is the primary ring and the other ring is the secondary ring. When the primary ring ceases to be operational (such as a cut cable) the network reconfigures itself (called “self-healing”) and it reconfigures the secondary ring as the primary ring. Both single mode fiber and multimode fiber cable systems can be used with FDDI. Multimode fibers have a wider optical bandwidth transmission capability. However, this introduces distortion and limits the maximum distance for multimode fiber systems to about 2 kilometers. Single mode fiber systems have maximum range of approximately 60 km. FDDI is a token passing architecture differing from token ring in that while a station has a token it can transmit as many frames as possible before the token expires. Because of this, there can be multiple frames on the ring at any time. The interconnection devices in a FDDI network include a dual attached concentrator (DAC) and dual attached station (DAS). These devices remove and insert data to the FDDI ring. Each of these devices has dual transmission capability. If the fiber ring is cut, they can automatically redirect data onto its other channel (the secondary ring). The DAC is a concentrator the converts the optical data on the FDDI system into another format that can be used to connect to other data networks. This allows one FDDI network node to connect to many other data communication devices. Figure 8.12 shows FDDI system that uses dual rings that transmit data in opposite directions. This diagram shows one dual attached station (DAS) and a dual attached concentrator (DAC).

The DAS receives and forwards the token to the mainframe computer. The DAC receives and token and coordinates its distribution to multiple data devices that are connected to it.

Figure 8.12: Fiber Distributed Data Interface (FDDI)

Network Management
Network management is set of procedures, equipment, and operations that keep a telecommunications network operating near maximum efficiency despite unusual loads or equipment failures. Network managers should be able to monitor, configure, and operate their network equipment from distant communication locations using a set of network management protocols. A key network management protocol is simple network management protocol (SNMP). SNMP is an industry standard communication protocol that is used to manage multiple types of network equipment (most vendors comply at some level). By conforming to this protocol, equipment assemblies that are produced by different manufacturers can be managed by a single network management program. While many vendors supply proprietary configuration and administration software for their products, many support diagnostic and maintenance features through the use of SNMP.

Chapter 9: Systems
Systems are either common to all users or privately leased by a customer for some specific application. Telecommunications networks serve as interconnection points between end-users. Until the late 1990’s, technology, cost and regulatory limitations have restricted most systems to offering specific types of services such as voice, data, or video transmission. The traditional categories of systems include local exchange networks (LECs), private telephone systems, interexchange telephone networks (long distance), cable television systems, data networks, and wireless networks.

Public Switched Telephone Networks (PSTNs)
Local exchange carriers (LECs) or post and telegraph and telecommunications (PTT) companies provide telephone services directly to residential and business customers located within a localized geographic area. Typically, these telephone companies provide services via copper lines that extend from a local carrier’s switching facilities to the end customer’s premises equipment (CPE). This is referred to local loop. Until the early 1990’s, most countries had a single company that provided local telephone services. This company was either owned or highly regulated by the government. To increase competition and reduce telephone service prices to consumers, some governments have begun to allow other companies to provide basic (local) telephone service. These competitive local exchange company (CLEC) or competitive access providers (CAPs) provide alternative connections to the public switched telephone networks (PSTN). The established telephone companies are now called the incumbent local exchange carriers (ILECs), Traditional incumbent local exchange company (ILEC) may be connected with one or more competitive local exchange companies (CLECs) who provide local telephone service in a defined geographic area. End customers (the houses) in the geographic area are connected to the End Office (EO) switching center by copper wire (local loop). The local loop length is approximately 1k to 10k feet from the EO. Rarely will the distance of the local loop exceed 20k feet. The local loop is the connection (wireless or wired) between a customer’s telephone or data equipment and a LEC or other telephone service provider. Traditionally, the local loop (also called “outside plant”) has been composed of copper wires that extend from the EO switch. The EO is the last switching office in the telephone network that connects customers to the telephone network. The EO switch cables meet the copper (or other types of lines) at the main distribution frame (MDF). The MDF is a wiring rack that allows technicians to splice the local loop lines with the lines from the switching system. Local loop lines leave the MDF in bundles (possibly thousands of wires in each bundle) and arrive in other junction points such as local distribution frames (LDF). The LDF allows the connection of the final connection (the “drop”) to the business or residence. At the entry to the customer’s location, there is often a network termination (NT) device that isolates the telephone network from the wiring inside the customers building. Figure 9.1 depicts a traditional local loop distribution system. This diagram shows a central office (CO) building that contains an EO switch. The EO switch is connected to the MDF splice box. The MDF connects the switch to bundles of cables in the “outside plant” distribution network. These bundles of cables periodically are connected to local distribution frames (LDFs). The LDFs allow connection of the final cable (called the “drop”) that connects to the house or building. A NT block isolates the inside wiring from the telephone system. Twisted pair wiring is usually looped through the home or building to provide several telephone connection points, or jacks, so telephones can connect to the telephone system.

Figure 9.1: Local Loop The end office interconnects calls between local customers. Each end office switch can usually supply service up to 10,000 customers. In larger areas (such as a city), established LECs may have several EO switches. The EO switches are interconnected using a higher-level tandem switch. If is a significant amount of calls regularly processed between end offices, they may be directly connected via high-speed communication lines (trunks). Local telephone service areas are connected to each other by inter-exchange switches. These higher level (meaning above tandem) are often operated by long-distance service providers called inter-exchange carriers (IXCs). The IXCs interconnect their long distance switches to the local network through a point of presence (POP). The POP is the toll center that allows separation of billing for local and long distance service. There are several other types of competitive access companies that are starting to provide local telephone service. These include cable television companies, Internet telephony service providers (ITSPs) and wireless local loop (WLL) providers. Public telephone networks are unrestricted dialing telephone networks that are available for public use to interconnect communication devices. Public telephone networks within countries and regions are a standard integrated system of transmission and switching facilities, signaling processors, and associated operations support systems that allow communication devices to communicate with each other when they operate. Figure 9.2 shows a basic overview of the Public Switched Telephone Network (PSTN) as deployed in a typical metropolitan area. PSTN customers connect to the end-office (EO) for telecommunications services. The EO processes the customer service request locally or passes it off to the appropriate end or tandem office. As Different levels of switches interconnect the parts of the PSTN system, lower-level switches are used to connect end-users (telephones) directly to other end-users in a specific geographic area. Higher-level switches are used to interconnect lower level switches.

Figure 9.2: Public Switched Telephone Network (PSTN) Switches send control messages to each other through a separate control-signaling network called signaling system number 7 (SS7). The SS7 network is composed of signaling transfer

points (STPs) and service control point (SCP) databases. A STP is used to route packets of control messages through the network. SCP’s are databases that are used by the network to process or reroute calls through the network (such as 800 number toll free call routing). SS7 also provides for the newer features such as incoming call identification and automatic call rerouting used by some service companies that provide 24/7, worldwide dial-in support. Switching systems are assemblies of equipment that setup, maintain, and disconnect connections between multiple communication lines. Switching systems are often classified by the type of network they are part of (e.g., packet or circuit switched) and the methods that are used to control the switches. The term “switch” is sometimes used as a short name for switching system. Public telephone switching systems have many switches within their network. A typical switch can handle up to 10,000 communication lines each. Early switches used mechanical levers (crossbars) to interconnect lines. Modern switches use computer systems to dynamically setup, maintain, and disconnect communication paths through one or more switches. True computer-based switching came about through the introduction of the electronic switching systems (ESS’s). ESS EOs did not require a physical connection between incoming and outgoing circuits. Paths between the circuits consisted of temporary memory locations that allowed for the temporary storage of traffic. For an ESS system, a computer controls the assignment, storage, and retrieval of memory locations so that a portion of an incoming line (time slot) could be stored in temporary memory and retrieved for insertion to an outgoing line. This is called a time slot interchange (TSI) memory matrix. The switch control system maps specific time slots on an incoming communication line (e.g., DS3) to specific time slots on an outgoing communication line. The public telephone network switching system architecture uses a distributed switching system that has a hierarchy of switching levels. Distributed switching systems connect calls through the nearest switching system. With distributed network architecture, the call processing requirements are distributed to multiple points. Using a multilevel hierarchy structure for switching systems allows switching to occur at lower levels of switching unless the telephone call must pass between multiple switches. At that point, the call is passed up to a higher-level switch for transfer to more distant locations. In conjunction with distributed network architecture, the ability to perform “dynamic routing” furthers the network’s resiliency to faults. Sometimes called “adaptive routing”, dynamic routing automatically re-routes communication paths or circuits as the network traffic levels (e.g., levels of congestion) change or as paths go in or out of service. A key part of public telephone networks is system reliability. As a result, in the event of equipment failure in such a network, backup (redundant) equipment must provide for continued service. Although this increases the reliability of switching systems, it also increases the system cost (for additional backup equipment) and complexity (recovery management systems). Public telephone switching systems use EO telephone switches to connect the telephone network to end customers. These switches serve as an end node switch that and provides local dial and access to local and long distance services. Switches that are used to interconnect switches to each other are called tandem switches. Some systems use mini-switches called remote digital terminals that are located near the EO switch. These mini-switches act as concentrator lines of voice channels between the end customers and the EO switching system. Concentrators grouping multiple communication lines into more efficient trunked (multi-channel) lines. A numbering plan is a system that identifies communication points within a communications network through the structured use of numbers. The structure of the numbers is divided to indicate specific regions or groups of users. It is important that all users connected to a telephone

network agree on a specific numbering plan to be able to identify and route calls from one point to another. Telephone numbering plans throughout the world and systems vary dramatically. In some countries, it is possible to dial using 5 digits and others require 10 digits. To uniquely identify every device that is connected to public telephone networks, the Comite Consultatif Internationale de Telegraphique et Telehonique (CCITT) devised a world numbering plan that provides codes for telephone access to each country. These are called country codes. Coupled with the national telephone number assigned to each subscriber in a country, the country code telephone makes that subscribers number unique worldwide. The International Telecommunications Union (ITU) administers the World Numbering Plan standard E.164 publishes any new standards or modifications to existing standards on the Internet. Each country defines its public telephone network numbering plans. The United States and Canada adopted the North American Numbering Plan (NANP) that allows the two countries to appear as one when dialing internally. Each country has a country code prescribed by the World Numbering Plan so they are accessed internationally as separate entities. The NANP is based on 10 digit numbering (NXX-NXX-XXXX). The number consists of a 3-digit area code, a 3-digit central office code, and a 4-digit line number. The first three digits (NXX) are the Numbering Plan Area (NPA) or area code. It is this 3-digit code that designates one of the numbering plan areas in the North American Numbering Plan for direct distance dialing. Originally, the format was N0/1X, where N is any digit 2 through 9 and X is any digit. From 1995 on, the acceptable format is NXX. With the massive requirement for telephone numbers generated by Internet access, fax machines, and cellular telephones, new area codes are being placed in service at an all time high rate. This is causing the telecommunications industry and standards bodies in North America to consider the implementation of “number portability”. When this occurs each subscriber will be assigned telephone numbers permanently (e.g., all subscribers in North America will dial ten digits to make a local call and take their number with them when they move.). Some of the key technologies behind the operation of the public telephone network include interconnection lines, network common control signaling, and intelligent call processing. Several types of interconnection systems are used to provide access to different services and systems available through the PSTN. To coordinate the overall operation of the PSTN, a standard common control signaling (CCS) system is used. Intelligent call processing combines these interconnection lines and common control signaling to provide for advanced services such as call forwarding, telephone number portability, and prepaid services.

Public Telephone System Interconnection
There are many types of interconnection options available to connect public telephone systems to other public telephone networks or private telephone networks. The type of connection selected depends on the type of private system, telecommunications regulations, and the needs of the company that uses the private telephone system (e.g., advanced calling features). In addition to standard telephone system connection types, there are also private-line connections that may be used to link private branch exchange PBX systems together. There are two types of connections used between switching systems: line side and trunk side. Line side connections are an interconnection line between the customer’s equipment and the last switch EO in the telephone network. The line side connection isolates the customer’s equipment from network signaling requirements. Line side connections are usually low capacity (one channel) lines. Trunk side connections are used to interconnect telephone network switching systems to each other. Trunk side connections are usually high capacity lines.

POTS (dial) Line Connections

POTS dial lines are 2-wire, basic line-side connections from an EO with limited signaling capability. Because dial lines are line-side connections, call setup time may be longer than those connections that employ trunk-side supervision.

Direct Inward Dialing (DID) Connections
Direct inward dialing (DID) connections are trunk-side (network side) EO connections. The network signaling on these 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 oneway, trunk-side connections. Until recently, most DID trunks were equipped with either dial pulse (DP) or dual tone multifrequency (DTMF) address pulsing. While many wireless carriers would have preferred to use multifrequency (MF) address pulsing, a number of LEC’s prohibited the use of MF on DID trunks.

Type 1 Connections
Type 1 connections are trunk-side connections to an EO. The EO uses a trunk-side signaling protocol in conjunction with a feature known as Trunk With Line Treatment (TWLT). This connection was originally described in technical advisory 76 published by AT&T in 1981. This interconnection was developed because dial line and DID connections did not provide enough signaling information to allow the connection of public telephone networks to other types of networks (such as wireless and PBX networks). The switch must be equipped to provide TWLT, or its equivalent to offer Type 1 service. As a result, type 1 is not universally available. The TWLT feature allows the EO to combine some line-side and trunk-side features. For example, while trunk-side signaling protocols are used, the calls are recorded for billing purposes as if they were made by a line-side connection. Type 1 connections are usually used as 2-way trunks. Two-way trunks are always 4-wire circuits, meaning they have separate transmit and receive paths, and almost always use MF address pulsing and supervision. The address pulsing normally uses wink-start control. One-way Type 1 connections can be provided on a 2-wire basis using E&M supervision or reverse battery like the DID connection.

Integrated Services Digital Network - Basic Rate Interface Connections (ISDN-BRI)
ISDN-BRI connection provides two bearer channels, each using a 64 kbps digital channel, as well as a 16 kbps data link for signaling messages. This 144 kbps combination is referred to as 2B+D, which signifies two bearer channels and one data channel. The bearer channels provide the voice portion while the data channel is used to transfer SS7 signaling messages. EO switches must have an ISDN-BRI interface and software installed to supply this connection.

Integrated Services Digital Network - Primary Rate Interface Connections
Another variation of Type 1 is the Integrated Services Digital Network - Primary Rate Interface (ISDN-PRI). It has capabilities similar to the ISDN-BRI but employs 23 bearer channels and one signaling channel, or a 23B+D configuration. The ISDN-PRI interconnection is provided using a standard DS1-level interface that would normally provide the equivalent of 24 voice channels. It offers the same calling capabilities as noted for the Type 1 and ISDN-BRI connections.

Type 2A Connections
Type 2A connections are true trunk-side connections that employ trunk-side signaling protocols. Typically, they are two-way connections that are 4-wire circuits using E&M supervision with

multifrequency (MF) address pulsing. The address pulsing is almost always under wink-start control. Type 2A connections allow the other public or private telephone network switching systems to connect to the PSTN and operate like any other EO. Type 2A connections may restrict calls to specific NXX (exchange) codes and access to operator services (phone number directories, emergency calls, freephone/toll free) may not be permitted. For some interconnections, additional connections (such as a type 1) may be used to supplement the type 2A connection to allow access to other operator or network services.

Type 2B Connections
Type 2B connections are high usage trunk groups that are used between EOs within the same system. The type 2B connection can be used in conjunction with the Type 2A. When a type 2B is used, the first choice of routing is through a Type 2B with overflow through the type 2A. Because the type 2b connection is used for high usage connections, it can access only valid NXX codes of the EO providing that it is connected to. Type 2B connections are almost always 4-wire, two-way connections that use E&M supervision and multifrequency (MF) address pulsing.

Type 2C Connections
Type 2C connections were developed to allow direct connection to public safety centers (E911) via a tandem or local tandem switch. This interconnection type must provide additional information such as the return phone number (complicated on mobile telephone systems) and the location of the caller. This information is passed on to a public safety answering point (PSAP).

Type 2D Connections
Type 2D interconnection lines allow direct connection from an operator services system (OSS) switch. The OSS switch is a special tandem that contains additional call processing capabilities that enables operator services special directory assistance services. The type 2D connection also forwards the automatic number identification information to allow proper billing records to be created. Type 2D connection will normally use trunks employing E&M signaling with wink start, and multifrequency (MF) address pulsing.

Type S Connections
Type S connections transfer signaling messages that are associated with other interconnection types (out-of-band signaling). The type S is a data link (e.g., 56 kbps) that is used to connect the signaling interfaces between switches. Type S connections permit additional features to be supported by the network such as finding and using call forwarding telephone numbers. Because type S connections cost money, some smaller public telephone networks do not offer or use type S connections. Figure 9.3 illustrates some of the different types of private to public telephone system interconnection. This diagram shows some groups of phone lines (e.g., dial line, Type 1) that provide limited signaling information (line-side) that primarily interconnect the PSTN with private telephone systems. Another group of lines (Type 2 series) are used to interconnect switching systems or to connect to advanced services (such as operator services or public safety services). The interconnection lines (trunk-side) provide more signaling information. Also shown is the type S connection that is used exclusively for sending control signaling messages between switching system and the signaling system 7 (SS7) telephone control network.

Figure 9.3: Telephone System Interconnection Hybrid systems have parts that utilize different technologies or services. Examples of hybrid systems include fiber to the curb (FTTC) where fiber provides a connection to the curb and coax or wireless provides the connection between the curb and the end-user.

Centrex is a service offered by a local telephone service provider that allows the customer to have features that are typically associated with a private branch exchange (PBX). These features include 3 or 4 digit dialing, intercom features, distinctive line ringing for inside and outside lines, voice mail waiting indication and others. Centrex services are provided by the central office switching facilities in the telephone network. Figure 9.4 shows a typical Centrex system. This diagram shows that the EO switch is equipped with Centrex software. Individual ports from the switch are connected to individual telephones at a company. The public telephone company programs the Centrex features for specific companies into the switch software. The Centrex software monitors the ports so advanced features such as abbreviated dialing can be performed. This example shows that a telephone that is used in a Centrex system can dial a 4 digit dialing number to reach a telephone connected within the companies Centrex telephone network.

Figure 9.4: Centrex System

Private Telephone Systems
Private telephone systems are independent telephone systems that are owned or leased by a company or individual. Private telephone networks include key telephone systems (KTS), private branch exchange (PBX), computer telephony (CT), wireless PBX, LAN telephony, and multimedia communication (e.g. videoconferencing). Private telephone systems are composed primarily of telephones (called “stations” or “terminals”), 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. Private telephone systems are often equipped with key assemblies and systems including voice mail, call accounting, a local maintenance terminal, and data interfaces. The voice mail system is controlled by the PBX only receiving calls when the PBX or computer telephony 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. Figure 9.5 shows some of the basic types of private telephone systems; key telephone systems (KTS), private branch exchange (PBX), and computer telephony integration (CTI) systems. The most simplistic private telephone network is a single telephone attached to a business line. Key systems allow each telephone in a business to answer and originate calls on several business lines. A PBX system allows many extensions within a business to call each other and the PSTN. Computer telephony (CT) systems are communication networks that merge computer intelligence with telecommunications devices and technologies.

Figure 9.5: Private Telephone Systems

Key Telephone System (KTS)
A key telephone system (KTS or key systems) is a multi-line private telephone network that allows each key telephone station to select one of several telephone lines. Key systems contain a key service unit (KSU) that coordinates status lights and lines to key telephones (Key Sets). Key systems have some advanced call processing features such as call hold, busy status, and station-to-station intercom. KTS are relatively simple non-switching telephone systems. The KSU only interfaces (connects) key sets to the public telephone lines allow calls to directly pass through. The KSU does sensing and provide display status lines to each key service unit. The first generation key systems allowed multi-button telephones to have an appearance (e.g., a button) for multiple end office

lines. When the incoming telephone line received a ringing signal, the key system flashed the appropriate button. To answer the call, the user picked up the handset and pressed the flashing button. This off-hook indication is sensed by the KSU which results in the key set’s line status light to become solid. This indicated to other telephone users that the line was being used. To place a call, the user would first view the lights on telephone line buttons. If a button was not lit, the user pressed the button. Again, the KSU sensed the off-hook condition and a solid light came on all key sets. To allow key sets to talk with each other without connecting through the public telephone network, most KTS systems included an intercom feature. The intercom feature allowed a key set to call one or all the key sets that are connected to the KSU. Figure 9.6 shows a typical key telephone system. This diagram shows telephones wired to a key service unit (KSU) that is connected to the PSTN. The KSU allows the telephones to have access to the outside lines to the PSTN. The KSU controls lights on the telephone sets, intercom access, and call hold.

Figure 9.6: Key Telephone System (KTS) Operation

Private Branch Exchange (PBX)
PBX systems are small private telephone systems that are used to provide telephone service within a building or group of buildings in a small geographic area. PBX systems contain small switches that use advanced call processing software to provide features such as speed dialing or call transfer. PBX systems connect local PBX telephones (stations) with each other and to the public switched telephone network (PSTN). While a PBX is similar to a miniature telephone company EO, PBX systems typically offer more features than public telephone system. The primary function of a PBX is to receive call requests (outgoing calls) from telephone stations users as well as routing incoming calls to specific extension. Figure 9.7 shows a private branch exchange (PBX) system. This diagram shows a PBX with telephone sets, voice mail system, and trunk connections to PSTN. The PBX switches calls between telephone sets and also provides them switched access to the PSTN. The voice mail depends on the PBX to switch all calls needing access to it along with the appropriate information to process the call.

Figure 9.7: Private Branch Exchange (PBX) System

Computer Telephony (CT)
Computer telephony (CT) systems are communication networks that merge computer intelligence with telecommunications devices and technologies. Figure 9.8 shows a sample CT system computer that contains a voice card. This voice card is connected to a multiple channel T1 line. The voice card connects digital PBX stations through the voice card to individual DS0 channels on the T1 line when calls are in progress. Several software programs are installed on this system that provide for call processing, voice mail, fax, and email broadcasting. The monitor shows a directory of extensions. The advanced call processing feature shows text names along with the individual extensions to allow callers to automatically search through a company’s directory without the need to use an operator.

Figure 9.8: Computer Telephony (CT) System Computer telephony integration (CTI) is the integration of computer processing systems with telephone technology. Computer telephony provides PBX functions along with advanced call processing and information access services. These services include, pre-paid telephony access control, interactive voice response (IVR), call center management, and private PBX. CTI uses a system of interfaces between telephone switching systems (typically PBX’s) and computer systems. It is through these interfaces that information is exchanged that causes actions by the receiving system in coordination with the sender. These industry standard interfaces include telephone application programming interface (TAPI), telephony services application programming interface (TSAPI), and Java TAPI (JTAPI). Telephony API (or TAPI) is a standard for communication between computer systems and telephone systems. PBX manufactures may provide TAPI via special interface cards that directly network with computer systems. JTAPI is a software communication standard based on Java programming language that allows computers to control PBX systems using Java programming

language. Through the use of these standard interfaces, IVR and ACD systems can exchange information with PBX and CTI systems. At the core of most CTI systems is a voice board installed in a Unix or Windows based computer system. The voice board is a small switch that contains line interfaces. One of the voice board line interfaces connect to a trunk line (such as a T1 or E1 line). Voice boards usually have multiple telephone extension line interfaces. These line interfaces can be for analog or digital telephones. A single CTI computer may contain multiple voice boards or expansion assemblies may be connected CTI systems can use standard or advanced digital telephones. Figure 9.9 shows a sample CTI system computer that contains a voice card. This voice card is connected to a multiple channel T1 line. The voice card connects digital PBX stations through the voice card to individual DS0 channels on the T1 line when calls are in progress. Several software programs are installed on this system that provide for call processing, IVR, ACD, voice mail, fax, and email broadcasting. The monitor shows a directory of extensions. The advanced call processing feature shows text names along with the individual extensions to allow callers to automatically search through a company’s directory without the need to use an operator.

Figure 9.9: Computer Telephony Integration (CTI)

Wireless Private Branch Exchange (WPBX)
WPBX systems integrate wireless telephones with a PBX switching system. Wireless PBX telephones (handsets) communicate through wired base stations (fixed radio transmitters) to the WPBX switching system. Most WPBX systems have automatic switching call transfer that allows wireless handsets to transfer their calls to other base stations as the move through the WPBX radio coverage areas. Base stations are strategically located around the served area (both inside and/or outside) to provide contiguous radio coverage. WPBX systems can be completely, or partially, wireless between the system and the telephone instruments. WPBX systems fill a need where all, or part, of the work force is highly mobile in a relatively small area such as a building/plant or a small commercial campus. Hospitals and manufacturing plants tend to have several types of personnel that tend to be constantly on the move: medical emergency personnel, maintenance personnel, and production-line supervisors to name a few.

Such people are frequently away from their desk or other fixed telephone station set location; however, it is often quite important that they be contacted quickly. There are several different types of WPBX systems industry standard systems and proprietary systems. Some of the standard WPBX systems include digital enhanced cordless telephone (DECT) and cordless telephony second generation (CT2). A WPBX radio system allows for voice or data communications on either an analog (typically FM) or digital radio channel. The radio channel typically allows multiple mobile telephones to communicate on the same frequency at the same time by special coding of their radio signals. The wireless office base station is the link between the radio transmissions sent to and received from the wireless telephone and the WPBX switching system. Wireless office base stations are similar to cell sites used in mobile telephone systems as they regularly communicate directly with the WPBX switching system. Because these base stations are fairly close to the switching system, they are directly connected by cable. This allows power to be supplied by the WPBX switching system and no battery backup power supply system is required. The cable that connects the base station to the switching system typically carries multiple voice and/or data channels. The power and data signals may be supplied over a single twisted pair or dedicated lines may be used for data and power. As the signals arrive at the base station, a communications controller divides the multiple channels, processes their signals, and routes them to the base station radio signal amplifier. The design objectives of a WPBX base station are similar to those of a general mobile telephone system, but there are several additional requirements. WPBX base stations must be much simpler to install, relocate and service (diagnose or debug). Operations without skilled or highly trained staff are very desirable. Many WPBX base stations are almost “self configuring,” implying that the system sets the frequencies of each base station automatically, to both optimize the overall frequency plan and to avoid interference with non-radio RF sources which may be present. The WPBX switching system coordinates the operation of all the base stations and wireless handsets in the system. The switching hardware and software for the WPBX may be incorporated into the main office telephone system (integrated), may reside in a separate switching and/or control module (external), or be completely separate from any wired system (independent). Integrated systems allow one switch to serve all the base stations and wired telephones connected to the system. An external system is used when a radio system is added to an existing system or the older system cannot be directly upgraded to support handoff switching inside the main switch. Independent systems may be used when there is no wired system installed. An independent system may only consist of WPBX handsets that can access a public cellular system for office use at a reduced billing rate. In a WPBX installation that has handoff (call transfer between base stations) capability, there is a continual process of signaling which occurs between all the handsets which are powered up but idle and the nearest base station(s). This allows the wireless handsets to handover (call transfer) between base stations as the move to other radio coverage areas. Figure 9.10 shows a sample WPBX radio system. A WPBX system typically has a switching system that is located at the company. The WPBX switch interfaces a PSTN communication line and multiple radio base stations. Radio base stations communicate with wireless office telephones that can move throughout the system. A control terminal is used to configure and update the WPBX with information about the wireless office telephones and how they can be connected to the PSTN.

Figure 9.10: Wireless Private Branch Exchange (WPBX)

LAN Telephony
Local access network (LAN) telephony (sometimes called TeLANophy) use LAN systems to transport voice communications. LAN telephone technology is an evolution of voice over IP (VoIP) and the rapid acceptance of virtual private networks (VPN’s) as an alternative to leased line private networks. The ability to share data networks with voice systems offers significant cost reduction for telephone services. Figure 9.11 shows a LAN telephony system. This diagram shows that a LAN telephone system consists of LAN telephones, a data network, a LAN call processing system, and a voice gateway to the PSTN. LAN telephones convert audio into digitized packets that are transferred on the LAN to the call processing computer telephony integration (CTI) system. Each LAN telephone has its own network data address. The call processing system communicates with LAN telephones over the same high-speed LAN data network that communicates with computers. When calls are received from the PSTN, the call processing system looks in the database to find the associated LAN telephone address (data address) and this address is used to alert the LAN telephone of an incoming call. When calls are originated from the LAN telephone, the dialed telephone number is passed to the call processing system. This system determines if the call is routed within the data network or if the voice gateway must be used to connect the call to the PSTN.

Figure 9.11: LAN Telephony

Multimedia Communication
Multimedia communication is the delivery of different types of information such as voice, data, or video. Communication systems may separately or simultaneously transfer multimedia information. Video conferencing is an application of multimedia communication technology that merges voice and video via the use of microphones, video cameras, and special multiplexers. Routinely

companies set up certain conference rooms at their various sites and equip them with video conferencing equipment. There are various video conferencing standards including the International Telecommunications Union (ITU) H.323 and standard T.120 for multipoint data conferencing. Video conferencing standards may allow for the use of whiteboards. Whiteboards are devices that can capture images or hand drawn text so they can be displayed in a window in at the connected video conferencing system. Whiteboards allow video conferencing users to place share documents, images, and/or hand written diagrams with one (or more) video conference call attendees. Figure 9.12 shows the basic operation of sending video over an Internet connection. This diagram shows a computer with video conferencing capability that calls a destination computer. Computer #1 initiates a video conference call to computer #2 using the address When computer #2 receives a data message from computer #1, a message is displayed on the monitor and an audio tone (ring alert) occurs. If the user on computer #2 wants to receive the call, they select the answer option (via the mouse or keyboard) that is generated by the software. Computer #1 then initiates a data connection with computer #2. The video conferencing software and data processing software in the computers (e.g., USB data bus and sound card) convert the analog audio signal from the microphone and digital video signal into a digital form that can be transmitted via the data link between the computers.

Figure 9.12: Video Conferencing through the Internet

Inter-Exchange System
Inter-exchange carrier (IXCs) systems are a combination telecommunications networks that connect local exchange carriers (LECs), competitive local exchange carriers (CLECs), local post, or telephone and telegraph (PTT) with each other. IXCs provide long distance bearer service communication and may provide other value-added teleservices. IXC’s are regulated by governmental commissions but are not usually government-owned. In other parts of the world the government may own and operate LECs and PTTs. In order to provide the bandwidth necessary to carry the volume of long-distance voice and data traffic at reasonable cost, most IXCs have deployed large bundles of fiber-optic cables that

interconnect their switching systems. Burying thousands of miles of fiber cable is costly. However, each pair of fibers is capable of providing many Gbps of bandwidth. The explosion of the Internet and the demand for advanced multi-media services continues to drive the demand for increased bandwidth at low cost. To increase the capacity of fiber cables, new fiber optic technology has emerged. By utilizing a technology known as dense wavelength division multiplexing, DWDM, each fiber can carry 80 or more separate light-waves. As of 2001, some DWDM technologies were capable of providing over 1 Tbps (1,000 Gbps) of bandwidth, enough to transmit in one second the contents of 150,000 encyclopedias. Advances in optical networking equipment and light-wave amplification technologies will continue to add bandwidth the fiber networks. Figure 9.13 shows a diagram of an inter-exchange carrier network. This diagram shows that the IXC interconnects LECs and CLECs with teach other through POP switching points. Access lines connect the IXC POP switching centers with LEC and CLEC tandem switching systems. These interconnection lines are typically dedicated high-speed carrier transmission lines such as DS3 or OC3 lines.

Figure 9.13: Inter-Exchange Carrier Network IXC networks use high-speed switching systems to interconnect high-capacity transmission lines. End users connect to IXC networks either through local telephone systems or through direct connection using customer provided equipment (CPE). Network interconnections are the points where IXCs connect to other networks. Transmission lines transport signals through the IXC network. High-speed switching systems provide interconnections between transmission lines and individual channels on those transmission lines. IXCs have multiple types of international interconnection issues to adapt telecommunication formats between different types of systems. The overall operation of services, switches, and transmission lines in an IXC is coordinated by network operations centers (NOCs). NOC’s continuously monitor the status and performance of all network nodes and links. If a network transmission or equipment fails, most networks will automatically reconfigure to (reroute) communication lines or automatically switch to backup systems. Practically all network components have redundant assemblies that will automatic switch into service on detection of equipment failure. Multiple routes are required between all switching facilities. These facilities are hardened with all support systems such as power, water, local emergency access, security redundant, and sabotage-proof. NOC’s management systems are usually distributed to multiple locations. These management centers contain information related to addressing, routing, and reroute scenarios. These regional centers are capable of distributing the network configuring information to remote switching nodes through communication links. Through this application of decentralized control and operations combined with an extensive data base maintenance and support activity, the utilization, efficiency, and security of network capacity can be maximized. The actual placement of circuits and switching equipment is confidential information when viewed as an operational system. This is because of the critical nature of this type information to all

countries. Major damage to a country’s telecommunications infrastructure could easily cripple an area or even a whole country. Telecommunications is considered a vital part of national security and special requirements exist to the protection and reliability of telecommunications networks.

International Interconnection
International interconnection issues include converting transmission line and control signaling formats, transcoding different types of digital voice signals, and rating billing records. IXC networks must be capable of converting transmission line formats. These include digital signaling standards (e.g., T1 to E1), different optical standards (SONET and SDH), and command signaling protocols differences such as ISDN signaling differences. Transcoding is the conversion of digital signals from one coding format to another. Transcoding is necessary because the digital signal companding process that is used for encoding/decoding signals is different throughout the world. This companding process increases the dynamic range of a binary signal by assigning different weighted values to each bit of information than is defined by the binary system. The A-law encoding system is an international standard and the uLaw standard is used in the Americas. IXC systems must be capable of creating billing record in different formats. Billing systems in different countries use different rating systems (e.g., flat rate compared to time usage). It may be necessary for IXCs to receive and pay in different currencies and currency exchange rates for different countries rapidly vary. The payment or receipt of payments for calls routed through the IXC must be settled through clearinghouse companies that have relationships with many IXC, LEC, and PTT operators. Several independent companies have installed or operate international transmission lines. These international circuits may be leased to IXCs or to independent corporations. Companies that operate these international transmission lines are often called international carriers (IC’s) or international record carriers (IRC’s). Figure 9.14 shows the basic transcoding process between mu-Law PCM coding and A-Law PCM coding. This diagram shows that a telephone that uses A-LAW PCM speech coding in North America system is communicating with a telephone in Europe that is using u-LAW PCM speech coding. This diagram shows that the transcoding system must identifies the type of PCM audio used by each system and the location of the transcoding gateway function. The PCM transcoder converts the A-LAW PCM signal to u-LAW PCM.

Figure 9.14: International Transcoding

The Internet is a public data network that interconnects private and government computers. The Internet transfers data from point-to-point by packets that use Internet protocol (IP). Each transmitted packet in the Internet finds its way through the network switching through nodes (computers). Each node in the Internet forwards received packets to another location (another node) that is closer to its destination. Each node contains routing tables that provide packetforwarding information. The Internet was designed to allow continuous data communication in the event some parts of the network were disabled. The world wide web (WWW) is an application on the Internet that allows users to graphically navigate through computers that are connected to the Internet. The Internet is a network of networks. Although these networks communicate with each other using many different languages (protocols), they all agree to transport data within their network according to a common Internet communication language called transmission control protocol/Internet protocol (TCP/IP). TCP/IP is a set of protocols developed by the U.S. Department of Defense (US DOC) that facilitate the interconnection of dissimilar computer systems across networks. The TCP protocol coordinates the overall flow of data during a data communication session between points (nodes) in the Internet. IP is an addressing structure that allows packets of data to be routed (re-directed) as they migrate through different networks to reach their ultimate destination. Each network receives packets of data in a format that is compatible with the Internet (IP address followed by control and data information) and they encapsulate (place the whole Internet data message into their own data packet format (including the IP address and control information). This allows IP data packets (called “datagrams”) to be sent through the network regardless of their actual length or format. Figure 9.15 shows that the Internet is the network of networks and it communicates using the universal protocol language TCP/IP. This diagram shows a user who is sending email through the Internet. In this diagram, the application is email. The data from the email is divided into packets and given sequence number by TCP protocol. The destination address is appended to each packet by the IP layer. The IP packets are then sent through an Ethernet LAN by encapsulating the IP datagram within the Ethernet data packet. When the data packet is extracted from the Ethernet, it is placed on the E1 transmission line. When the IP data packet reaches the ATM network, it is subdivided into very small 53 byte data packets that travel through the ATM network. When the ATM packets reach their destination in the ATM network, the original IP

datagram is recreated and transferred via the T1 communication line. The T1 communication line interfaces to another Ethernet data network. This Ethernet data network encapsulates the IP datagram and forwards it on to the NIC of the receiving computer. The NIC of the receiving computer removes the IP address and reassembles the IP data packets to form the original email message.

Figure 9.15: Internet Data Routing

Cable Television Systems
Cable television systems provide video and data services through a system of high bandwidth coaxial cables and fibers. The cable network includes a head-end amplifier that combines the broadcast and data signals for transmission to the subscribers. High-speed Internet access is obtained by including a cable modem termination system (CMTS) function within the head-end that connects to a 10/100Mbps Ethernet router. The head-end is connected to fiber or coax trunks that carry the signals into the neighborhoods where they are tapped off to provide service to the residence. Earlier cable TV systems provided only one-way broadcast type services such as standard and premium channel television. Upgrading these earlier systems to support the two-way communications necessary to offer Internet access, pay-per-view, voice and video-on-demand services requires large capital investments. Many cable TV carriers have merged with large telecommunications companies in order to take advantage of the enormous market potential that exists. Cable TV systems can deliver high-speed Internet access at costs that are far below that of digital subscriber line (DSL). Figure 9.16 shows a typical cable television network. This diagram shows that cable television systems can be simple one-way video distribution systems to advanced two-way high-speed digital networks. The head-end is the initial distribution center for a cable television (CATV) system. The head end is where incoming video and television signal sources (e.g., video tape, satellites, local studios) are received, amplified, and modulated onto TV carrier channels for transmission on the CATV cabling system. The cable distribution system is a cable (fiber or coax) that is used to transfer signals from the head end to the end-users. The cable is attached to the

television through a set-top box. The set-top box is an electronic device that adapts a communications medium to a format that is accessible by the end-user.

Figure 9.16: Cable Television Network Also in the mid 1990’s, a major shift occurred in the broadcast industry. The conversion from analog systems to digital systems provided broadcasters with the tools they needed to bundle multiple types of services onto a television channel signals. This included cable modems, digital television, and even telephone service. The ability to integrate several services into one transmission signal allows the cable television operator to offer many new services without significant investment in new cable systems. Analog CATV systems typically provide 50-100 video channels while digital CATV systems to provide hundreds of video channels, high-speed Internet access, and telephone service. As cable systems evolved to include fiber (optical) cable and two-way amplifiers, cable networks evolved to allow data transmission in both directions. On the coaxial (RF) cable, the return path was assigned to frequencies in the range below 50 MHz. This frequency range was unassigned for television operation. Fiber optic cables use separate strands for each direction as each fiber cable often has many (30+) fiber strands. The two-way cable system requires cable modems at the user end and a coordinating modem at the head-end of the system. The cable modem is a communication device that modulates and demodulates (MoDem) data signals to and from a cable television system. A modem at the headend coordinates the customer’s modem and interfaces data to other networks (such as the Internet). Figure 9.17 shows a two-way cable television system. This diagram shows that the two-way cable television system adds a cable modem termination system (CMTS) at the head-end and a cable modem (CM) at the customer’s location. The CMTS also provides an interface to other networks such as the Internet.

Figure 9.17: Two-Way Cable Television System

A cable modem is a device that MOdulates/DEModulates data signals on a coaxial cable and divides the high data rate signals into digital signals designated for a specific user. Cable modems are often asymetrical modems as the data transfer rate in the downstream direction is typically much higher than the data transfer in the upstream direction. The typical gross (system) downstream data rates range between 30-40 Mbps and gross upstream data rates typically range up to 2 Mbps. Usually 500 to 2000 users share the gross data transfer rate on a cable system. Cable modems also have the requirement to divide the high-speed digital signals into low-speed connections for each user. In 2001, the average data rates for a cable modem users was approximately 720 kbps. Figure 9.18 shows a block diagram of a cable modem. This diagram shows that a cable modem has a tuner to convert an incoming 6 MHz RF channel to a low frequency baseband signal. This signal is demodulated to a digital format, demultiplexed (separated) from other digital channels, and is decompressed to a single data signal. This data signal is connected to a computer typically in Ethernet (e.g. 10 Base T Ethernet) or universal serial bus (USB) data format. Data that is sent to the modem is converted to either audio signals for transfer via a telephone line (hybrid system) or converted to an RF signal for transmission back through the cable network.

Figure 9.18: Cable Modem

Data Over Cable Service Interface Specifications (DOCSIS)
The data over cable service interface specifications (DOCSIS) is a standard used by cable systems for providing Internet data services to users. The DOCSIS standard was primarily developed by equipment manufacturers and CATV operators. It details most aspects of data over cable networks including physical layer (modulation types and data rates), medium access control (MAC), services, and security. The DOCSIS cable modem specifications are available from CableLabs® at The downstream information flows to all users that are tuned to a specific RF channel on the cable system. There may be several RF channels used to serve many cable modem users in a system. Each individual cable modem decodes their portion of the data on a specific RF channel. For transmitting on the upstream side, each user is assigned time of a few milliseconds each where the user can transmit short bursts of data. Dividing the channel into small slices of data is well suited for short delays to keyboard commands. To convert the Internet data into a format suitable for delivery on a cable channel, a CATV upconverter is used at the head-end of the cable system. The CATV upconverter handles both digital and analog television signals. Usually 10-20 upconverters are installed into a single equipment chassis. To allow cable modems to connect to data networks (such as the Internet), a cable modem termination system (CMTS) is used. The CMTS an interface device (gateway) that is located at the head-end of a cable television system to send and adapt data between cable modems and other networks. A single 6 MHz wide television channel is capable of 30-40 Mbps data transmission capacity. This is because coaxial cable offers a communication medium that is relatively noise free (compared to radio or unshielded twist pair cable) that allows the use of complex modulation

technologies (combination of amplitude and phase modulation). These modulation technologies can transfer several bits of data for each Hertz of bandwidth (bits per Hertz). In 2001, cable modems could transmit data using 64 QAM modulation technology. To increase the data rate, even more complex modulation technologies such as 256 QAM or even to 1024 QAM have been demonstrated [1]. The DOCSIS system is focused around packet service such as Internet Protocol (IP) and asynchronous transfer mode (ATM) to provide a variety of services (e.g., variable bit-rate, constant bit-rate) with the ability to offer varied levels of quality of service (QoS). This allows the DOCSIS system to offer multiple channels to a home or business that can provide for various services such as voice (constant bit-rate), data (high reliability), and video (high-speed data).

Cable Telephony
Cable telephony is the providing of telephone services that use CATV systems to initiate, process, and receive voice communications. Cable telephony systems can either integrate telephony systems with cable modem networks (a teleservice) or the cable modem system can simply act as a transfer method for Internet telephony (bearer service). Because of government regulations (restrictions or high operational level requirements) in many countries, some cable operators are delaying the integration of telephone services with cable network. In either case, cable telephony systems are data telephony systems that include a voice gateway, gatekeeper, and a media interface. Voice gateway is a network device that converts communication signals between data networks and telephone networks. A gatekeeper is a server that translates dialed digits into routing points within the cable network or to identify a forwarding number for the public telephone network. A multimedia transfer adapter converts multiple types of input signals into a common communications format. Figure 9.19 shows a CATV system that offers cable telephony services. This diagram shows that a two-way digital CATV system can be enhanced to offer cable telephony services by adding voice gateways to the cable network’s head-end CMTS system and media terminal adapters (MTAs) at the residence or business. The voice gateway connects and converts signals from the public telephone network into data signals that can be transported on the cable modem system. The CMTS system uses a portion of the cable modem signal (data channel) to communicate with the MTA. The MTA converts the telephony data signal to its analog audio component for connection to standard telephones. MTAs are sometimes called integrated access devices (IADs).

Figure 9.19: Cable Telephony Because of the high data transmission capability of cable television systems, cable telephony system can provide video telephony service. Video telephony is a telecommunications service

that provides customers with both audio and video signals between their communications devices.

Residential Cordless
Cordless systems are short-range wireless telephone systems that are primarily used in residential applications. Cordless telephones regularly use radio transmitters that have a maximum power level below 10 milliWatts (0.01 Watts). This limits their usable range to 100 meters or less. Cordless telephone systems consist of two types of transceivers, one a base station that connects to the public switched telephone network and the other a mobile handset unit that communicates directly with the base station. Transmissions from the mobile unit are received by the base station and then placed on the public switched telephone network. Information received from the switched telephone network is transmitted by the base station to the mobile unit. The earliest generation of home cordless telephones used a single radio channel that used amplitude modulation. These first generation cordless phones were susceptible to electrical noise (static) from various types of electronic equipment such as florescent lights. The noise encountered when using these phones sometimes created a consumer impression that cordless telephone quality was below standard wired telephone quality. Improved versions of cordless phones that used FM modulation to overcome the electrical noise resulted. As cordless phones became more popular, interference from nearby phones became a problem. In apartment buildings where there were many users of cordless phones in close proximity, the ability to initiate and receive calls could be difficult as radio channels became busy with many users. This led to the development of cordless phones that used multiple radio channels. As voice privacy became more of an issue, cordless phones began to use scrambled voice. Some of these voice privacy systems were analog while a majority of cordless phones that offer voice privacy use digital transmission. Figure 9.20 shows the evolution of cordless telephones. Until the mid 1990’s, most cordless telephones were limited to use in a small radio coverage area of their base station that was usually located in the home. That home base station was normally connected to the telephone line of the owner (either residential or a single office telephone line) and they were not intended to serve the general public. To add more value to the use of cordless phones, cordless telephones evolved to allow access to base stations in public locations. Cordless telephones could then be used in the home and in areas that were served by public base stations. The next evolution for cordless telephones was the combination of other types of wireless products and services into the cordless phone. This included the combination of wireless office and cellular telephones into a cordless phone.

Figure 9.20: Evolution of Cordless Telephone Systems Most home cordless telephones used frequencies in unlicensed radio frequency bands. Because so many homes operate cordless phones, each manufacturer must build-in circuitry to minimize the interference caused by other cordless devices. The original cordless phones use a very crowded frequency band (around 27 and 49 MHz) utilizing analog radio wave modulation.

Recently, cordless telephones have been developed that operate in the 902-928 MHz unlicensed industrial, scientific, and medical (ISM) frequency band. Residential cordless telephones must automatically coordinate their radio channel access as they operate independently of any type of network control. To coordinate radio channel access and avoid interference to other cordless handsets installed in the vicinity, cordless phones perform radio channel scanning and interference detecting prior to transmitting a signal. Because cordless telephone systems do not as a rule have a dedicated control channel to provide information, the cordless handset and base station continuously scan all of the available channels (typically 10 to 25 channels). Figure 9.21 shows the basic cordless telephone coordination process. This diagram shows that when the cordless phone or base station desires to transmit, the unit will choose an unused radio channel and begin to transmit a pilot tone or digital code with a unique identification code to indicate a request for service. The other cordless device (base station or cordless phone) will detect this request for service when it is scanning and its receiver will stop scanning and transmit an acknowledgement to the request for service. After both devices have communicated, conversation can begin. When another nearby base station detects the request for service, it will determine that the message is not intended for it and will not process the call and scanning will continue.

Figure 9.21: Cordless Telephone Operation

Mobile Telephone
Mobile telephones connect people to the public switched telephone system (PSTN) or to other mobile telephones. Mobile telephone service includes cellular, PCS, specialized and enhanced mobile radio, air-to-ground, marine, and railroad telephone services. Wireless systems are primarily designed to transfer voice and or data from one point to one or more other points, (multipoint). Many networks make use of some wireless technologies as a transport medium even though we do not consider them to be wireless networks. Examples of wireless networks include cellular, personal communication service, (PCS), paging, wireless data, satellite, and broadcast radio and television. Wireless systems are composed of radios, radio towers or base stations, interconnection systems, and network management and information systems.

Radios may be fixed in location (such as a television) or may be mobile (such as a cellular telephone). Some radios may only communicate in one direction (typically a receiver) or may have two-way capability. When a single radio has both a transmitter and receiver contained in the same unit, it is called a transceiver. Radio towers are poles, guided towers, or free standing constructed grids that raise one or more antennas to a height that increases the range of a transmitted signal. Radio towers can vary in height from about 20 feet to more than 300 feet. A single radio tower may host several antenna systems that include paging, microwave, or cellular systems. Radio towers are located strategically around the city to provide radio signal coverage to specific areas. At the base of the towers are electronic control rooms that contain the components to operate the radio portion of the communications system. Radio towers and their associated radio equipment (e.g., base station) may include one or more antennas, transmitters, receivers (for two-way systems), system controllers, communication links, and power supplies. Transmitters provide the high level RF power that is supplied to the antenna. For broadcast systems, the amount of transmitter power can exceed 50,000 Watts. Receivers boost and demodulate incoming RF signals from mobile radios. If a base station contains receivers, it is typical to use one or more different antennas for the receivers. Controllers coordinate the overall operation of the base station and coordinate the alarm monitoring of electronic assemblies. Communication links allow a command location (such as a television studio or a telephone switching center) to control and exchange information with the base station. Base station radio equipment requires power supplies. Most base stations contain primary and backup power supplies. A battery typically maintains operation when primary power is interrupted. A generator may also be included to allow operation during extended power outages. Switching facilities are typically used in two-way mobile communication systems to allow the connection of mobile radios to other radios in the system or to the public telephone network. When used in a cellular system, the switching system is typically called a mobile switching center (MSC). The MSC, just like a local telephone company, processes requests for service from mobile radios (subscribers) and routes the calls to other destinations. Wireless systems may be connected to other networks. Broadcast wireless systems are connected to media sources (such as audio or video programs) via satellite links while cellular networks may be interconnected to the public telephone network. Interconnection involves the physical and software connection of network equipment or communications systems to the facilities of another network such as the public telephone network. Government agencies such as the Federal Communications Commission (FCC) or Department of Communications (DOC) regulate interconnection of wireless systems to the public telephone networks to ensure reliable operation.

Land Mobile Radio (LMR)
Land mobile radio (LMR) systems are traditionally private systems that allow communication between a base and several mobile radios. LMR systems can share a single frequency or use dual frequencies. LMR in the United States is regulated by the FCC in part 90, Private Land Mobile Radio Services, includes various types of private radio services including police, taxi, fire and other types of two-way and dispatch services. LMR systems are traditionally private systems that allow communication between a base and several mobile radios. LMR systems can share a single frequency or use dual frequencies. Where LMR systems use a single frequency when mobile radios must wait to talk, this is called a simplex system. To simplify the mobile radio design and increase system efficiency, some LMR systems use two frequencies; one for transmitting and another for receiving. If the radio cannot transmit and receive at the same time, the system is called half duplex. When LMR systems use

two frequencies and can transmit and receive at the same time, this is called full duplex. When a company operates an LMR system to provide service to multiple users on a subscription basis (typically to companies), it is called a public land mobile radio system (PLMR). Figure 9.22 shows a traditional two-way radio system. In this example, a high power base station (called a “base”) is used to communicate with portable two-way radios. The two-way portable radios can communicate with the base or they can communicate directly with each other.

Figure 9.22: Land Mobile Radio (LMR) System LMR systems are used by: taxicab companies, conventioneers, police and fire departments, and places where general dispatching for service is a normal course of business communications. SMR radios are regularly designed to be rugged to survive the harsh environment. SMR radios can usually be programmed with a unique code. This code may be an individual code or group code (e.g., pre-designated group of users such as a fire department). This allows all the radios belonging to a group, or a sub-group, to be “paged” by any party in the group. A push-to-talk method is used during the dispatch call (page) or reply. This push-to-talk radio-to-radio communication efficiently utilizes the airwaves because of the bursty (very short transmission time) nature of the information. Automated land mobile radio systems are divided into two categories; SMR or Enhanced SMR (ESMR). Enhanced land mobile radio systems operate and have similar features to mobile telephone systems.

Cellular and PCS
Cellular and PCS mobile telephone systems allow mobile telephones to communicate with each other or to the public telephone system through an interconnected network of radio towers. When linked together to cover an entire metro area, the radio coverage areas (called cells) form a cellular structure resembling that of a honeycomb. The cellular systems are designed to have overlap at each cell boarder to enable a “hand-off” (also called a “handover”) from one cell to the next. As a customer (called a subscriber) moves through a cellular or PCS system, the mobile switching center (MSC) coordinates and transfers calls from one cell to another and maintains call continuity. Figure 9.23 shows a mobile telephone system. The wireless network connects mobile radios to each other or the public switched telephone network (PSTN) by using radio towers (base stations) that are connected to a mobile switching center (MSC). The mobile switching center can transfer calls to the PSTN.

Figure 9.23: Mobile Telephone System When a cellular system is first established, it can effectively serve only a limited number of callers. When that limit is exceeded, callers experience too many system busy signals (known as blocking) and their calls cannot be completed. More callers can be served by adding more cells with smaller coverage areas - that is, by cell splitting. The increased number of smaller cells provides more available radio channels in a given area because it allows radio channels to be reused at closer geographical distances. There are two basic types of systems: analog and digital. Analog systems typically use FM modulation to transfer voice information and digital systems use some form of phase modulation to transfer digital voice and data information. Although analog systems are capable of providing many of the services that digital systems offer, digital systems offer added flexibility as many of the features can be created by software changes. The trend at the end of the 1990’s was for analog systems to convert to digital systems. To allow the conversion from analog systems to digital systems, some cellular technologies allow for the use of dual-mode or multi-mode mobile telephones. These telephones are capable of operating on an analog or digital radio channel, depending on availability. Most dual-mode phones prefer to use digital radio channels in the event both are available. This allows them to take advantage of the new features such as short messaging and digital voice quality. Cellular systems have several key differences that include the radio channel bandwidth, access technology type (FDMA, TDMA, CDMA), data signaling rates of their control channel(s), and power levels. Analog cellular systems have very narrow radio channels that vary from 10 kHz to 30 kHz. Digital systems channel bandwidth ranges from 30 kHz to 1.25 MHz. Access technologies determine how mobile telephones obtain service and how they share each radio channel. The data signaling rates determine how fast messages can be sent on control channels. The RF power level of mobile telephones and how the power level is controlled typically determines how far away the mobile telephone can operate from the base station (radio tower).

Third Generation Wireless (3G)
Third generation wireless (3G) is a term commonly used to describe the third generation of technology used in a specific application or industry. In cellular telecommunications, third generation systems used wideband digital radio technology as compared to 2nd generation narrowband digital radio. For third generation cordless telephones, products used multiple digital radio channels and new registration processes allowed some 3rd generation cordless phones to roam into other public places. The 3G system is actually the universal mobile telecommunications System (UMTS). The UMTS system offers personal telecommunications services that use the combination of wireless and fixed systems to provide seamless telecommunications services to its users. The UMTS allows

bandwidth on-demand transmission capacities of up to 2 Mb/s in some of its radiolocations. It should be compatible with GSM and broadband ISDN systems. Figure 9.24 shows a 3rd generation broadband wireless system. This system uses two 5 MHz wide radio channels to provide for simultaneous (duplex) transmission between the end-user and other telecommunication networks. There are different channels used for end- user to the system (called the “uplink”) and from the system to the end-user (called the “downlink”). This diagram shows that 3G networks interconnect with the public switched telephone network and the Internet. While the radio channel is divided into separate codes, different protocols are used on the radio channels to give high priority for voice information and high-integrity to the transmission of data information.

Figure 9.24: 3rd Generation Wireless

Wireless Local Loop (WLL)
Wireless local loop (WLL) service refers to the distribution of telephone service from the nearest telephone central office to individual customers via a wireless link. In some cases, it is referred to as “the last mile” in a telephone network. This term is a bit misleading, though, because the coverage area of a WLL system may extend many miles from the central office. Competitive local exchange carriers (CLEC) are competitors to the incumbent local exchange carriers (ILECS) and are likely to use WLL systems to rapidly deploy competing systems. If CLECs do not use wireless systems, they must either pay the existing phone company for access to the local loop (resale) or dig and install their own wire to the local customers. Many countries, that do not have large wired networks such as the United States, are using wireless local loop as their primary phone system. Figure 9.27 shows a wireless local loop system. In this diagram, a central office switch is connected via a fiberoptic cable to radio transmitters located in a residential neighborhoods. Each house that desires to have dial tone service from the WLL service provider has a radio receiver mounted outside with a dial tone converter box. The dial tone converter box changes the radio signal into the dial tone that can be used in standard telephone devices such as answering machines and fax machines. It is also possible for the customer to have one or more wireless (cordless) telephones to use in the house and to use around the residential area where the WLL transmitters are located.

Figure 9.27: Wireless Local Loop The most basic service offered by wireless local loop (WLL) system is to provide standard dial tone service known as plain old telephone service (POTS). In addition to the basic services, WLL systems typically offer advanced features such as high-speed data, residential area cordless service, and in some cases, video services. To add value to WLL systems, WLL service providers will likely integrate and bundle standard phone service with other services such as cellular, paging, high speed Internet, or cable service. WLL systems can provide for single or multiple-line units that connect to one or more standard telephones. The telephone interface devices may include battery back up for use during power outages. Most wireless local loop (WLL) systems provide for both voice and data services. The available data rates for WLL systems vary from 9.6 kbps to over several hundred kbps. WLL systems can be provided on cellular and PCS, private mobile radio, unlicensed cordless, and proprietary wideband systems that operate the 3.4 GHz range.

Chapter 10: Voice over Data Networks
Sending voice over data networks is a process of sending digitized telephone signals over a network that was designed primarily for data communications. If the telephone signal is in analog form (voice or fax, the signal is first converted to a digital form. Packet routing information is then added to the digital voice signal so it can be routed through the data network. 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 10.1 shows how a standard telephone can call a standard telephone through the Internet. In this example, the calling phone dials a local telephone number of a voice gateway. The telephone call is routed through the local exchange company to gateway. When the gateway automatically answers the call, the gatekeeper will receive the dialed digits and it’s database will convert the telephone number to the Internet protocol (IP) address of a destination gateway that is close to the receiving telephone. The gatekeeper sends a service request message to the receiving gatekeeper asking if it is willing and able to complete the telephone call. If the receiving gatekeeper is willing and able to complete the call, the call setup information is exchanged between the gateways (e.g. call and feature preferences). This setup information allows a virtual path to be created from the calling gateway to the receiving gateway. This virtual path takes the audio, converts it to digital form, compresses and packetizes the information, adds the destination gateway address to each packet, routes the packets through the Internet to the destination gateway, and converts the digital audio back to its original analog form. When the call setup information is completed, the receiving gatekeeper dials the telephone number of the receiving telephone. If someone answers the receiving telephone, the audio path is routed via the virtual path through the Internet.

Figure 10.1: Packetized Voice over Data Networks

Voice or Media Gateway (MG)
A voice gateway or media gateway is a communications device or assembly that transforms audio that is received from a telephone device or telecommunications system (e.g. PBX) into a format that can be used by a different network. A voice gateway usually has more intelligence (processing function) than a bridge as it can select the voice compression coder and adjust the protocols and timing between two dissimilar computer systems or voice over data networks. Figure 10.2 shows the functional structure of a voice gateway device. This diagram shows that this voice gateway interfaces between a public telephone network to a packet data network. Input signals from the public telephone network pass through a line card to adapt the information for use within the voice gateway. This line card separates (extracts) and combines (inserts) control signals from the input line from the audio signal. If the audio signal is in analog form, the voice

gateway converts the audio signal to digital form using an analog to digital converter. The digital audio signal is then passed through a data compression (speech coding) device so the data rate is reduced for more efficient communication. This diagram shows that there are several speech coder options to select from. The selection of the speech coder is negotiated on call setup based on preferences and communication capability of both voice gateways. After the speech signal is compressed, the digital signal is formatted for the protocol that is used for data communication (e.g. IP packet or Ethernet packet). This call processing section of the voice gateway may insert control commands (in-band signaling) to allow this gateway to directly communicate with the remote gateway. These digital signals are sent through a data access device (e.g. router shown here) so it can travel through the data communication network. The overall operation of the voice gateway is controlled by the call processing section. The call processing section receives and inserts signaling control messages from the input (telephone line) and output (data port). The call processing section may use separate communication channels (out-of-band) to coordinate call setup and disconnection.

Figure 10.2: Voice Gateway Operation

Gatekeeper or Media Gateway Controller (MGC)
A gatekeeper or Media Gateway Controller (MGC) is a server that coordinates access to other servers. The gatekeeper receives requests from clients, determines the destination server that it needs to communicate with, and coordinates access with that server. For packet voice systems, the gatekeeper translates user names or telephone numbers into physical address for H.323 conferencing. Figure 10.3 shows the basic functions of the gatekeeper that controls a voice gateway or media gateway. This diagram shows that the gatekeeper performs access control, address translation, services coordination, control signaling coordination, and billing recording. The gatekeeper receives requests for services from the voice gateway. The gatekeeper will check it’s databases to ensure the caller is an authorized customer and what services the customer is allowed to use. If it is a request for an outgoing call, the voice gateway will then provide the gatekeeper with the destination IP address or dialed telephone number. The gatekeeper will search for the IP address of a remote gatekeeper in its database that is located near the called number. If there is no registered (pre-authorized) gatekeeper, the gatekeeper may send a broadcast request message to other gatekeepers in an attempt to find a gatekeeper that is willing to service the call. The Gatekeeper will then send a request message to the remote gatekeeper to request access to the destination telephone or communication device. If the destination gatekeeper authorizes access, the gatekeeper will then negotiate for the parameters for communication between the voice gateways (e.g. speech coder). The gateway will initiate a billing record of the call and its associated usage records.

Figure 10.3: Voice Gatekeeper Operation

Integrated Access Device (IAD)
An integrated access device (IAD) converts multiple types of input signals into a common communications format. IADs are commonly used in PBX systems to integrate different types of telephone devices (e.g. analog phone, digital phone and fax) onto a common digital medium (e.g. T1 or E1 line). Figure 10.4 shows an integrated access device (IAD) combines multiple types of media (voice, data, and video) onto one common data communications system. This diagram shows that three types of communication devices (telephone, television, and computer) can share one data line (e.g. DSL or Cable Modem) through an IAD. The IAD coordinates the logical channel assignment for device and provides the necessary conversion (interface) between the data signal and the device. In this example, the telephone interface provides a dialtone signal and converts the dialed digits into messages that can be sent on the data channel. The video interface buffers and converts digital video into the necessary video format for the television or set top box. The data interface converts the line data signal into Ethernet (or other format) that can be used to communicate with the computer. This diagram also shows that the IAD must coordinate the bandwidth allocation so real time signals (such as voice) are transmitted in a precise scheduled format (isochronous). The digital television signal uses a varying amount of bandwidth as rapidly changing images require additional bandwidth. The IAD also allocates data transmission to the computer as the data transmission bandwidth becomes available (what left after the voice and video applications use their bandwidth).

Figure 10.4: Integrated Access Device (IAD) Operation

Internet Protocol (IP) Telephony Systems
Internet protocol telephony systems provide voice services through the use of networks that use Internet protocol. These IP telephony systems initiate, process, and receive voice (and sometimes multimedia) communications using the embedded protocol of the Internet. These systems may use the public Internet, private data networks (e.g. LAN based), private IP-based networks or hybrids of public and private systems. The common signaling protocols for IP telephony systems include H.323, session initiated protocol (SIP), media gateway control protocol (MGCP), and MEGACO. There is a difference between the Internet and IP based networks. IP based networks uses Internet protocol to route information within the network. The Internet is a public data network that interconnects private and government computers together. An IP based network does not have to be part of the Internet and it is possible for an Internet network operator to partition their data network to allow for different quality of service (QoS) levels. As a result, it is possible to reliably send SS7 control messages over IP based networks that may be part of the public Internet.

H.323 Packet-Based Media Communications Systems
H.323 is an ITU industry standard for multimedia communications that combines and coordinates multiple data compression and communication standards to allow audio, picture, and video transmission between users on packet switched networks. The H.323 system has four key components: terminals, gateways, gatekeepers, and multipoint control units (MCUs). The original name for H.323 was Visual Telephone Systems and Equipment for Local Area Networks. Figure 10.5 shows the basic structure of an H.323 system. This diagram shows that the H.323 system can interconnect standard telephones and data communication devices (multimedia computers) through the use of gateways and gatekeepers. Gateways convert the audio and multimedia information into formats that can be transmitted through the packet data network. Gatekeepers coordinate, authorize, and bill (if billing is required) access through the gateways. This diagram shows that when calls are initiated through the H.323 network, the gateway requests access from the gatekeeper. The gatekeeper reviews its database to determine if the request is authorized and may perform translation of dialed digits to a data (IP) address. The destination gatekeeper is then contacted and if it authorizes service, its associated gateway will be setup to translate the call to the communication device (e.g. telephone) that is receiving the call.

Figure 10.5: H.323 System H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that may not provide a guaranteed Quality of Service (QoS). H.323 specifies techniques for compressing

and transmitting real-time voice, video, and data between a pair of videoconferencing workstations. It also describes signaling protocols for managing audio and video streams, as well as procedures for breaking data into packets and synchronizing transmissions across communications channels. The H.225 protocol is used for call signaling for the set up of connections between H.323 endpoints (terminals and gateways), over which the real-time data can be transported. Call signaling involves the exchange of H.225 protocol messages over a reliable call-signaling channel. For example, H.225 protocol messages are carried over TCP in an IP based H.323 network. After a communication session is established, H.245 protocol can be used to send control signaling between communicating H.323 endpoints. The H.245 control messages are carried over H.245 control channels. The H.245 control channel is the logical channel 0 and is permanently open, unlike the media channels. The messages carried include messages to exchange capabilities of terminals and to open and close logical channels. Supplementary Services for H.323, namely Call Transfer and Call Diversion, have been defined by the H.450 series of specifications. H.450.1 defines the signaling protocol between H.323 endpoints for the control of supplementary services. H.450.2 defines Call Transfer and H.450.3 Call Diversion. Call Transfer allows a call established between endpoint A and endpoint B to be transformed into a new call between endpoint B and a third endpoint, endpoint C. Call Diversion provides the supplementary services Call Forwarding Unconditional, Call Forwarding Busy, Call Forwarding No Reply and Call Deflection. The “hooks” for Supplementary Services are specified in H.323 Version 2. The H.323 system can interconnect standard telephones and data communication devices (multimedia computers) through the use of gateways and gatekeepers. Gateways convert the audio and multimedia information into formats that can be transmitted through the packet data network. Gatekeepers coordinate, authorize, and bill (if billing is required) access through the gateways. When calls are initiated through the H.323 network, the gateway requests access from the gatekeeper. The gatekeeper reviews its database to determine if the request is authorized and may perform translation of dialed digits to a data (IP) address. The destination gatekeeper is then contacted and if it authorizes service, its associated gateway will be setup to translate the call to the communication device (e.g. telephone). Figure 10.6 shows a sample of how call signaling may progress with a PSTN telephone user initiates a call to a terminal in an H.323 system. In this example, the caller dials the H.323 telephone number. This produces an initial address message (IAM) to the H.323 gateway. This creates an H.323 access request message (ARQ) to the gatekeeper to lookup the destination IP address. The gatekeeper responds with an Admission Confirm (ACF) message that provides the gateway with the destination address of the H.323 terminal. (IP Telephone). In this example, the H.323 gateway sends a setup messages (fast setup that includes the initial call parameters) and the H.323 terminal responds with a call proceeding message. This allows the H.323 gateway to create an SS7 CPG message to the originating PSTN switch. The H.323 sends an access request (ARQ) message to the gatekeeper. The gatekeeper responds with an access confirmation (ACF) message that allows the call to proceed. This results in the H.323 terminal sending an alerting message that indicates that the terminal is ringing. This permits the H.323 gateway to send an SS7 ACM message to the PSTN switch. The media path from the H.323 terminal is then connected to the audio channel of the PSTN switch (packet voice is converted to PCM circuit voice in the H.323 gateway). When the user at the H.323 terminal answers the call, the terminal sends a connect message (fast setup that includes parameters) to the H.323 gateway. This alerts the PSTN switch with an SS7 answer message (ANM). This allows a second media path to be connected between the PSTN switch and the H.323 terminal.

Figure 10.6: SS7 Signaling into an H.323 System

Session Initiated Protocol (SIP)
SIP is an application layer protocol that uses text format messages to setup, manage, and terminate multimedia communication sessions. SIP is a simplified system as compared to the ITU H.323 packet multimedia system. SIP is defined in RFC 2543. The SIP system is primarily composed of clients (equivalent of a media gateway) and servers (equivalent of media gateway controller or gatekeepers). SIP protocol can use session description protocol (SDP) to control media streams through a network. SDP is a text-based protocol that is used throughout to provide high-level definitions of connections and media streams. The SDP protocol is used with session initiated protocol (SIP). The SDP protocol is used in the PacketCable system. SDP is defined in RFC 2327. Figure 10.7 shows the SS7 that is used to initiate a call into an SIP system. This diagram shows that a caller that is connected to a SIP network (SIP client) initiates a call using an invite command that contains a destination SIP Uniform Resource Locator (URL). This identifier is sent to the proxy server that determines and maps the URL to the actual destination number that will be sent in the initial address message (IAM). The proxy server informs the SIP client that the call routing is in progress (it is trying to connect). The Invite command is then forwarded to the network gateway (NGW). The NGW creates an IAM that contains the destination phone number. The PSTN switch sends back an ACM to the NGW. The NGW informs the proxy server that the call is progressing and the proxy server forwards this session progress message to the SIP client. This allows an audio path to be connected between the PSTN and the SIP client. When the destination telephone user answers, the PSTN sends and answer message to the NGW. The NGW translates this command and sends a message updating the session to indicate the call has been answered. This is forwarded to the SIP client. When the SIP client acknowledges the message, the NGW can connect a second media path from the SIP client to the PSTN switch.

Figure 10.7: SS7 Signaling into an SIP System

Media Gateway Control Protocol (MGCP)
MGCP is a control protocol that uses a controller to setup, manage, and terminate multimedia communication sessions in a centralized communications system through media gateways. In essence, a system that uses MGCP has centralized control. This centralized control structure differs from other multimedia control protocol systems (such as H.323) that allow distributed network control as end points in the network may directly setup and control a communication session. MEGACO is a control protocol that is similar to MGCP and some industry experts believe that MEGACO will eventually replace MGCP. MCGP can use either text format or binary format messages to setup, manage, and terminate multimedia communication sessions. MEGACO offers the potential for large distances between the media gateway controller (MGC) and media gateway (MG).

SS7 and Internet Protocol (IP) Signaling Systems
SS7 messages can be directly transported over IP networks or the functional equivalent of SS7 control message can be sent as control messages (e.g. text based messages) directly between elements connected to a data network (e.g. the Internet). The use of IP based networks for voice, data, multimedia, and signaling offers new potential levels of network efficiency (utilization). The Internet transfers data from point to point by packets that use Internet protocol (IP). Each transmitted packet in the Internet finds its way through the network switching through nodes (computers). Each node in the Internet forwards received packets to another location (another node) that is closer to its destination. Each node contains routing tables that provide packet forwarding information. These routing tables may be dynamically changed as a result of new connections or paths that may become available through the network. This is different than the SS7 system that allows the operator to have more precise control over the routing tables. To help increase the reliability of sending SS7 control messages through IP based networks, the Signaling Transport (SIGTRAN) as been developed. The SIGTRAN protocol stack utilizes the

Stream Control Transmission protocol (SCTP). SCTP is an IP protocol that combines near-real time data transfer with reliable packet delivery and validation. To allow the use of protocols with SS7 systems, several protocol adaptation layers have been created. These protocols adapt the message structures and flow of messages to emulate the message transfer parts (MTP) of the SS7 protocol stack. Interfacing to the SS7 network to IP networks is performed by a signaling gateway (SG). The SG is used to interface a signaling control system (e.g. such as SS7) and a network device (e.g. a transfer point, database, or other type of signaling system). The signaling gateway may convert message formats, translate addresses, and allow different signaling protocols to interact. Figure 10.8 shows that SS7 signaling systems can be interconnected with voice over data networks and that SS7 messages can be transported over the Internet protocol. This diagram shows that analog and digital telephones are connected to the PSTN. To interconnect these telephones to voice over data network telephones, the media portion of each communication session is routed through a media gateway where it is converted from the PSTN circuit switched form to a IP packet data media format (packetized voice.) This diagram shows that the packet media can be routed through a data network (e.g. Internet) to an endpoint communication terminal such as a multimedia computer or an IP telephone. This diagram also shows that the SS7 network can control the PSTN through SS7 signaling messages and it can communicate to the media gateway through IP signaling messages.

Figure 10.8: Hybrid SS7 and Internet Protocol Network

Chapter 11: Services
Telephone applications are the processes or programs that provide specific features and benefits for the customer that involve the transfer of information through communication systems. Telecommunications services are the underlying communications processes that provide information for telecommunications applications. It is common to use the word services in place of applications, especially when the service is very similar to the application. Examples of communication applications include voice mail, email, and web browsing. Telephone services include voice, data, and video transmission. Voice services can be categorized into quality of service and voice privacy. Data services use either circuit-switched (continuous connection) data or packet-switched (dynamically routed) data. Video transmission is the transport of video (multiple images) that may be accompanied by other signals (such as audio or closed-caption text). Telecommunication services may be provided (distributed) to one or more users of information. Distribution of services can be categorized into broadcast, multicast, and point-to-point delivery. Broadcast service delivers the same information to all users in a network. Multicast distribution service distributes information to specific users within a network. Point-to-point service transfers information between two specific users or devices within a network. The transfer of information between users can be unmodified or modified. Telecommunication services that only involve the transport of information are called bearer services. Services that require information processing (such as store and forward) in the network are called teleservices. Services that combine bearer services and teleservices into a new unique services are called supplemental services. System features and services are typically provided by call-processing software in the telephone network that interacts with end-user equipment. As a result, some telephone equipment may be only able to operate some system features (e.g., call forwarding) while some service features require device capabilities and software (e.g., calling number identification presentation). To help ensure the correct operation of services and device interaction, industry standards are created.

Voice Communications
Voice communication is the transmission and reception of audio and other signals that can be represented by the frequency band used for voice signal transmission. Telephone systems transfer voice signals in a variety of forms through by wire, radio, light, and other electronic or electromagnetic systems. These forms include analog and digital voice signals. Options for voice communications include different voice quality of service levels and voice privacy options.

Voice Quality
Voice quality is a measurement of the level of audio quality, often expressed in mean opinion score (MOS). The MOS is number that is determined by a panel of listeners who subjectively rate the quality of audio on various samples. The rating level varies from 1 (bad) to 5 (excellent). Good quality telephone service (called toll quality) has a MOS level of 4.0. The first telephone systems used analog signals to represent the voice. To overcome the cumulative noise limitations of analog signal transmission, digital transmission systems were created. These digital transmission signals represented voice signals by discrete levels that can

be recreated eliminating the noise. As a result, in the 1960’s, many modern telephone systems began to offer digital voice communications. The first digital voice services converted (digitized) the analog voice signal to a 64 kbps digital signal. This 64 kbps digital channel called a DS0) provided “toll quality” voice with a MOS score of 4.0 or above. Generally, there is a tradeoff between system efficiency (bandwidth used) and the level of voice quality. To gain system efficiency (to add more customers per interconnection line), some telephone systems compress the voice using speech-coding (data compression) technology. The first compressed voice service uses adaptive pulse coded modulation (ADPCM) that further compresses the 64 kbps DS0 to 32 kbps ADPCM. Other voice compressed voice service have been developed that can use low bit-rate standard or proprietary speech compression algorithms. These can further compresses the 64 kbps DS0 to below 16 kbps or even 8 kbps.

Voice Privacy
Voice privacy is a process that is used to prevent the unauthorized listening of communications by other people. Voice privacy involves coding or encrypting of the voice signal with a key so only authorized users with the correct key and decryption program can listen to the communication information. Digital systems are inherently more secure than analog systems because they can easily use an encrypted mode of operation. This encrypted mode of operation “scrambles” voice data before it is sent to other users in the network. The encryption uses a key (mask value) that is calculated from some form of secret data. When the voice data is received, it must be decrypted using the same mask value that was used to encrypt it. Although an interceptor may be capable of receiving the data signals, they cannot learn the true data value unless the secret number that was added to it is also known. While the telephone system can offer an encryption mode that encrypts the signaling between the end-user’s phone line and the telephone network, it is more common for the end-user to maintain their own voice encryption system. This does help to prevent unauthorized access to the telephone system. This also allows the end-user to have many different voice encryption algorithms. The voice encryption algorithms are typically stored on the end-user’s telephone devices.

Data Communications
There are two basic types of data communications: circuit-switched data and packet-switched data. Circuit-switched data provides for continuous data signals while packet-switched data allows for rapid delivery of very short data messages.

Circuit-switched Data
Circuit-switched data is a data communication method that maintains a dedicated communications path between two communication devices regardless of the amount of data that is sent between the devices. This gives to communications equipment the exclusive use of the circuit that connects them, even when the circuit is momentarily idle. To establish a circuit-switched data connection, the address is sent first and a connection (possibly a virtual non-physical connection) path is established. After this path is setup, data is

continually transferred using this path until the path is disconnected by request from the sender or receiver of data. Figure 11.1 shows a circuit switched data system. In this figure, a computer is sending a data file through a circuit switched data communications network to a home office computer. To start the data file transfer, the computer sends the destination address (address to the data network. The destination address is used to program the switches between the points on which ports they will receive the data and which ports they will send the data. As soon as all the switching connections are made, the computer can start sending data to the office. Throughout the connection, this path will be maintained through the initial path (the same switch ports) without any changes.

Figure 11.1: Circuit-switched Data

Packet-switched Data
Packet data service provides data transfer in the form of short packets of information. The public telephone network was designed primarily to offer voice services. Shortly after the telephone network was introduced, circuit-switched (continuous) data services were offered. The operation requirements for circuit-switched and packet-switched data services are very different. Circuitswitched data has substantial time and is inefficient for serving sensing control and applications that require small amounts of information. Initially the standard telephone system had to be enhanced (functionally divided) to offer packet data service. However, with the digitization of communications systems, telephone systems operate more like packet data systems. Typical applications for packet data service include Internet browsing, wireless email, train control system, route guidance, credit card processing and many other applications that benefit from the transmission of data in bursts when communicating. Packet data systems provide effective use of the resources. Packet data systems only use network equipment resources when there is information to transfer. This provides the advantage of charging only for the amount of information used and increased system efficiency. A packet is a group of digital bits that is transported and switched through a network of packet switches (often called routers) to their destination. The structure of these packets (digital bit sequence) is arranged in a specific format to allow the determination of the destination address for each packet in addition to the data that is being transported. Optionally, the packet structure may include other information such as the packet originator and error protection bits. Transmitting data through a packet network involves dividing data files into small packets (typically under 100 bytes of information). A packet data system divides large quantities of data into small packets for transmission through a switching network that uses the addresses of the packets to dynamically route these packets through a switching network to their ultimate destination. When a data block is divided, the packets are given sequence numbers so that a

packet assembler/disassembler (PAD) device can recombine the packets to the original data block after they have been transmitted through the network. Figure 11.2 shows the basic operation that uses packet-switched data. In this example, a laptop computer is sending a file to a company’s remote computer that is connected to a packet data network. The laptop computer data communication software requests the destination address for the packets for the user to connect to the remote computer ( In this example, the source computer divides the data file into three parts and adds the packet address to each of the 3 data packets. The packets are sent through routers in the packet network that independently determine the best path at the time that will help the packet reach its destination (smart switches). This diagram shows the three packets take 3 different routes to reach their destination. When the 3 packets reach their destination, the remote computer reassembles the data packets into the original data file.

Figure 11.2: Packet-switched Data

Video Communications
Video communication is the transmission and reception of video images using electrical or optical transmission signals. Telecommunications systems can transfer video signals in analog or digital form.

Analog Video
Analog video is the representation of a series of multiple images (video) through the use of rapidly changing signals (analog). This analog signal indicates the position, luminance, and color information within the video signal. Sending a video picture involves the creation and transfer of a sequence of individual still pictures called frames. Each frame is divided into horizontal and vertical lines. To create a single frame picture on a television set, the frame is drawn line by line. The process of drawing these lines on the screen is called scanning. The frames are drawn to the screen in two separate scans. The first scan draws half of the picture and the second scan draws between the lines of the first scan. This scanning method is called interlacing. Each line is divided into pixels that are the smallest possible parts of the picture. The number of pixels that can be displayed determines the resolution (quality) of the video signal. The video signal television picture into three parts: the picture brightness (luminance), the color (chrominance), and the audio.

Digital Video
Digital video is a sequence of picture signals (frames) that are represented by binary data (bits) that describe a finite set of color and luminance levels. Sending a digital video picture involves the conversion of a scanned image to digital information that is transferred to a digital video receiver.

The digital information contains characteristics of the video signal and the position of the image (bit location) that will be displayed.

Distribution Services
Distribution is the transfer of information throughout a geographic area or through a network. Distribution services include broadcast, multicast, and point-to-point communication.

Broadcast transmission is the distribution of an information signal to a specified geographic area or network system. Broadcasting allows the same information to be received by all customers in that geographic area that can successfully receive (demodulate) and decode the information. Figure 11.3 shows broadcast communication service. This diagram shows two broadcast examples: radio broadcast and network broadcast. Part (a) shows a radio broadcast tower that is sending an audio broadcast to all radios that are within its radio signal coverage area. Part (b) shows a network broadcast system that sends a data message that is coded to indicate the message is a broadcast message. This message contains an address that indicates it is a broadcast message. When routers or other data distribution devices receive this message, each distribution device forwards the data broadcast message to the other network parts for which it is connected to. All communication devices that are connected to the network can receive the broadcast message.

Figure 11.3: Broadcast Communication

Multicast transmission is a communications service where a single message or information transmission contains an address (code) that is designated for several devices (nodes) in a network. Devices must contain the matching code to successfully forward or decode the message. Figure 11.4 shows examples of how multicast services can be implemented. The first method uses encoded video broadcast transmission and encoded messaging to allow only a select group to view the received information. While all the television broadcast receivers all receive the same radio signal, only the receivers with the correct code will be able to descramble the television signal. The second method uses multicast routing in the Internet to store and forward data to an authorized group of recipients that are connected to its router. When a router in the Internet that is capable of multicast service receives a multicast message, it will store the message for forwarding. It then uses the multicast address to lookup a list of authorized recipients in its routing table. The stored message is then forwarded to the authorized receiving device or next router that is part of the multicast service.

Figure 11.4: Multicast Communication

Point-to-point communication is the transmission of signals from one specific point to another. Point-to-point communication uses addressing to deliver information to a specific receiver of the information. It is possible to implement point-to-point communication through a broadcast network by using device addressing or through a network using network routing. Figure 11.5 shows examples of how point-to-point services can be implemented. The first method is a paging system that uses device addressing to uniquely identify a specific receiver of the information. While all the pager devices receive the same radio signal, only the receiver that has the correct paging code will be able to descramble the paging message. The second method uses network routing in the Internet to store and forward data to a specific recipient in the network. When a router in the Internet receives a point-to-point message, it will use the address to lookup the best forwarding path to transfer the information towards its destination. Using pointto-point communications by network routing, only the designated recipient will receive the data.

Figure 11.5: Point-to-Point Communications

Teleservices are telecommunication services that provide added processing or functionality to the transfer of information between users. Teleservices are categorized by their high level (application) characteristics, the low level attributes of the bearer service(s) that are used as part of the teleservice, and other general attributes. High level attributes include: application type (for example voice or messaging) and operation of the application. The low-level description includes a list of the bearer services required to allow the teleservice to operate with their data transfer rate(s) and types. Other general attributes might specify a minimum quality level for the teleservice or other special condition. The categories of teleservices available include voice (speech), short messaging, facsimile, and group voice. Figure 11.7 shows a typical teleservice in a mobile communication system. In this diagram, a telephone user wishes to send a fax to a recipient who is traveling. The designated recipient has setup a fax forwarding service where the delivery of incoming faxes can be instructed. The

sender is given the recipient’s fax number. When the sender dials the number, the call is routed through the telephone network to the fax forwarding service provider (step 1). When the incoming call is detected, the fax forwarding service receives the fax into a fax mailbox (step 2). Later that day, the recipient of the fax forwarding service calls in and enters a fax forwarding number (step 3). The fax forwarding service then checks the fax mailbox and automatically sends all the waiting faxes to the new number (possibly a hotel fax number) that has been updated by the recipient (step 4). Because this service involves both the transport and processing of information, it is categorized as a teleservice.

Figure 11.7: Teleservice

Chapter 12: Call Processing
Call processing is the steps that occur during the duration of a call. These steps are typically associated with the routing and control of the call. Call processing may involve several processes that are performed by a combination of the public switched telephone network, private telephone network, or intelligent (switching) network. This chapter provides a description of some of the more popular call processing procedures.

Distinctive Ringing
Distinctive ringing is a service feature that alerts a customer via a special ring (usually short, long or rapid ring) that an incoming call is received that has a different purpose or priority from others that are received on that same telephone line. Distinctive ringing is used for sharing multiple phone numbers on a single line or for priority ringing. Figure 12.1 shows the operation of a telephone system that has distinctive ringing feature. This diagram shows a single telephone that is assigned two different telephone numbers even though the telephone operates on one telephone number (one switch port). In this example, when an incoming call is received for the registered number 555-6234, it is re-directed (forwarded) towards the actual destination number 555-1234 along with information that allows the system to uniquely identify the call with a dual ring (2 rings in the 2 second ring period). When calls are received to 555-1234, the ring is a single 2 second/4 second cadence. This allows the receiver of the call to determine which telephone number was dialed by the distinctive ring sound.

Figure 12.1: Distinctive Ringing Operation

Call Hold
Call hold is a feature that allows a user to temporarily hold and incoming call, typically to use other features such as transfer or to originate a 3rd part call. During the call hold period, the caller may hear silence or music depending on the network or telephone feature. Figure 12.2 shows how a call can be temporarily placed on hold so the call can stay connected without the user having to continue conversation with the caller. During hold, the audio from the user is muted. For an analog line, the call hold feature involves placing a load (connection) across the line so that current may continue to flow through the circuit. For digital systems, the call hold feature may send a call hold message back to the system (such as a signaling message on the signaling channel) so the system can know that the status of the telephone station has changed to “hold.”

Figure 12.2: Call Hold Operation

Call Forwarding
Call forwarding is a call processing feature allows a user to have telephones calls automatically redirected to another telephone number or device (such as a voice mail system). There can be conditional or unconditional reasons for call forwarding. If the user selects that all calls are forwarded to another telephone device (such as a telephone number or voice mailbox), this is unconditional. Conditional reasons for call forwarding include if the user is busy, does not answer or is not reachable (such as when a mobile phone is out of service area). Figure 12.3 shows how call forwarding can be used to automatically redirect telephone calls based on specific conditions. This example shows that a call may be redirected by the switching system to one extension if the user is busy and to a different extension if the user has programmed the extension as “Do Not Disturb” or if it is busy. When the call is received by the system destined for extension 1001, the call processing system uses the indication of busy along with a redirection table to determine the call must be automatically transferred to extension 1003.

Figure 12.3: Call Forwarding Operation

Selective Call Forwarding
Selective call forwarding is a service feature that forwards calls to one (or multiple) telephone numbers dependent on the incoming call forwarding criteria. Selective call forwarding can be used to redirect calls of a specific type (such as fax calls) to a pre-designated number (such as an office fax machine.)

Figure 12.4 shows how selective call forwarding can be used to deliver calls to alternate number based on a specific criteria type. This diagram shows a selective call forwarding service that routes fax calls to different telephone number or extension after it detects the call is a fax call. After the system detects that the incoming call is a fax (by the fax tones), the switch call processing software transfers the call to the destination number that is connected to a fax machine.

Figure 12.4: Selective Call Forwarding Operation

Selective Call Acceptance
Selective call acceptance is a service feature that only delivers calls to their dialed destination if they are on a previously specified selective call acceptance telephone number list. Calls that are received by other numbers are provided with a pre-recorded announcement that states the number is not accepting their call or the call may be routed to an alternate directory number. Figure 12.5 shows how selective call acceptance can be used to only deliver calls from a specific list of callers. This diagram shows that regional managers from a service center can call from numbers that are pre-defined (their office numbers) and that when their calls are received, the will be connected to the specified telephone or extension. Call that are received from numbers that are not on the selective call acceptance list are transferred to an automated message unit that plays a pre-recorded announcement that states the number is not accepting their call.

Figure 12.5: Selective Call Acceptance Operation

Conference Call
Conference call systems provide the ability to connect three or more telephones to a telephone conversation. Figure 12.6 shows how a conference call can uses a conference bridge to allow several users to effectively communicate in a conference call (3 or more users). This example shows that this

conference bridge uses audio level detectors to determine the level of the microphone audio level for each conference call participant that is talking. As a person begins to talk, the conference bridge increases the gain on the microphone and decreases the gain on the speaker line. This process effectively dynamically reduces the background noise from non-participating members while providing good sound quality to participants that are talking.

Figure 12.6: Conference Call Operation

Call Pickup
A telephone call processing (switch control) feature that enables a telephone user to answer a telephone from another telephone station. When a call is received, a key sequence is entered from specific groups or any telephone (dependent on how the system is setup) and the call is redirected to the extension or line that has picked up (entered the code) the line. Figure 12.7 shows the operation of a telephone system that has a call pickup group feature. In this example, an incoming call is received and is directed toward the main extension 1001. When the call is received for extension 1001, the system uses the call pickup group list to determine that extensions 1001, 1002, 1003, and 1004 are programmed for call pickup. This allows any of these extensions to answer the call by pressing a key sequence (key “3” in this example).

Figure 12.7: Call Pickup Operation

Call Transfer
Call transfer is a call connection routing feature that transfers a call from one telephone or extension to another. Call can occur at various stages of the call conversation through the use of a system call request feature. The system special service request feature is often called a “Flash” feature. The flash feature is created to indicate a desire to recall a service function or to activate a custom calling feature (such as a call transfer request). A flash feature service request can be created when the user initiates a short on-hook interval or through the sending of a special service request message. The short on-hook interval is created by a momentary operation of the telephone switch hook, during a prolonged off-hook period. The special service request message can be sent by a button on a telephone (such as a PBX telephone) or by pressing the SEND key on a mobile telephone. Figure 12.8 shows a typical call transfer process used in a private telephone system. In this example, a caller is connected from the public telephone network to extension 1001 via incoming line 0001 on a telephone system (PBX). The user on extension 1001 desires to transfer the call to extension 1011. The first step involves extension 1001 sending a hold signal to the telephone system that allows it to place incoming line 0001 on hold. This is followed by the user at extension 1001 sending the digits indicating the destination of the call transfer (1011). The system then connects extension 1001 with 1011 and the user at 1001 may inform the user at 1011 that a call is being transferred to them. When the user at 1001 hangs up, the connection between 1001 and 1011 is disconnected and a connection is made between incoming line 1001 and 1011.

Figure 12.8: Call Transfer Operation

Call Waiting (CW)
Call waiting is a telephone call processing feature that notifies a telephone user that a another incoming call is waiting to be answered. This is typically provided by a brief tone that is not heard by the other callers. Some advanced telephones (such as digital mobile telephones) are capable of displaying the incoming phone number of the waiting call. After the service provides the subscriber with the notification of an incoming call while the subscriber’s call, controlling subscriber can either answer or ignore the incoming call. If the controlling subscriber answers the second call, it may alternate between the two calls. Figure 12.9 shows how call waiting service may be provided on an analog telephone line. In this example, a call is in process with caller 1. During the call, a second caller (caller 2) dials the telephone number of the user that has call waiting. The system discovers that the line or extension is busy on another call. The system also determines that this user has the call waiting service processing feature available so it sends a call waiting message tone to the user (only heard by the user). If the user desires to answer the call, the user sends a flash message (a momentary open on the line) that indicates to the telephone system to place the current call in progress on hold and switch to the other incoming call (caller 2). Each time a flash message is sent, the line alternates between each incoming caller.

Figure 12.9: Call Waiting Operation

Calling Line Identification (CLI)
Calling line identification (CLI) is a service which displays the calling number prior to answering the call that allows telephone customer to determine if they want to answer the call. The calling number may be used by the telephone device to look-up a name in memory (e.g. mom) and display the name along with the phone number. Figure 12.10 shows the calling number identification operation. Calling number identification operation starts with the reception of a call. When the call is received, the initial address message (IAM) contains the calling party number of the incoming call. The IAM may contain additional information such as the text name of the calling party. This example shows that the local switching system extracts this information and combines this information with the ring signal (using different frequencies and amplitudes) and sends it to the customer during the alerting (ringing) process. If the customer has the appropriate display equipment, the calling number information is display as the telephone rings.

Figure 12.10: Calling Line Identification (CLID) Operation

Hunting (also called “rollover”) is a telephone call-handling feature that causes a transferred call to “hunt” through a predetermined group of telephones numbers until finds an available (“non busy”) line. Figure 12.11 shows the process of hunting for an available telephone. This diagram shows that an incoming call enters into the telephone switching system and is attempted to be delivered to the main telephone line extension (or dialed telephone number port). The switching system is programmed with a hunt list that allows the switching system to determine where to redirect a call if it is unable to deliver a call. This example shows that the system first tries 1001 that is off-hook

(unavailable). The hunt group table shows that the call should be routed to 1002. Because 1002 is also unavailable, the hunt group list shows instructs the switch to try extension 1003. This telephone (or port) is available and the call can be delivered (telephone rings).

Figure 12.11: Hunting Operation

Voice Dialing
Voice dialing is a process that uses the callers voice to dial a call. Voice dialing involves the activation of the voice dialing feature (either by pressing a key or by saying a key word), saying words in the vocabulary of the voice dialing processor, and providing feedback to the user (usually by audio messages) of the status of the voice dialing process. Voice dialing can be a system (network provided) or device (stored in the telephone device) feature. There are two basic forms of voice dialing; speaker independent and speaker dependent. Speaker independent voice dialing allows any user to initiate voice commands from a predefined menu of commands. Speaker dependent voice dialing requires the user to store voice commands so these voice commands can be activated by the user and others are unlikely to match the speaker dependent voice commands. Speaker dependent voice recognition allows a user to program specific names into the telephone or network voice recognition system. Figure 12.12 shows different types of dialing using voice commands. In this example, both the telephone set and telephone network have voice dialing control capability. When the telephone is used for voice control, the voice from the user is converted to digital form by and analog to digital converter. After the audio is converted to digital form, it is analyzed for patterns and matched to previously stored voice control digital sound patterns. This example shows that the telephone set has some speaker independent patterns (such as start and digits) that have been previously stored. It also shows that this telephone also has a speaker dependent memory storage area that allows the user to store specific names. When these specific names are spoken, the telephone set will retrieve the pre-stored telephone numbers or extensions.

Figure 12.12: Voice Dialing Operation This diagram shows similar voice dialing capabilities that are located in a telephone network. This network voice control system has more accurate voice processing capability than the telephone set and each voice control module can service many line cards as users only use voice control for brief periods.

Night Service
Night service is a programmatic state of a telephone switch during closed hours where a special messaging capability gives users the option to leave messages as well as receive information in specific (usually night time) hours of operation. Call routing is adjusted on a telephone switch differently then being directed to a telephone console operator. Figure 12.13 shows how a telephone system can change its basic operation for daytime and nighttime telephone service. In this example, during the day, all the incoming calls are routed to (received by) a receptionist at extension 1001. At night (between 5 pm and 8 am), the calls are automatically redirected to an automated telephony call processing system that is connected to extension 1014. When the automated attendant detects a ring signal, answers the phone (offhook signal) and plays a pre-recorded messaging informing the caller of options they may choose to direct the call to a specific extension. In this example, the automated call attendant software decodes DTMF tones or limited list of voice commands to determine the routing of the call. The automated call attendant software then determines if the destination choice is within the option list and if the extension is available. If the extension is available, the automated attendant will send a command to the computer telephony board (voice card) that can switch the call to the selected extension. If the extension is not valid or not available, the automated attendant will provide a new voice prompt with updated information and additional options.

Figure 12.13: Night Service Operation

Automatic Callback
Automatic callback is a custom local access signaling services (CLASS) feature that allows a caller to complete a call to a busy station by dialing an activation code (usually a single digit) and hanging up. The system automatically rings both parties when the lines are available. Figure 12.14 shows the basic operation of automatic callback. To activate automatic callback service, after a call has dialed a number that is busy, the customer dials an automatic callback feature code and hangs up the telephone. The local switch (caller’s phone carrier) informs the remote (distant) switch of automatic callback request. This reserves (blocks) the called number from receiving additional calls until the automatic callback service is completed. When the called number becomes available, the remote switch sends a message to the local switch and this rings the original caller’s number (possibly with distinctive ring feature.)

Figure 12.14: Automatic Callback

Automated Attendant System
Automated attendant is a processor control system that performs telephone console attendant functions such as answering a call, transferring callers to specific user stations, directing callers to voice mail, or performing other related call-routing functions without the assistance of a live attendant. The caller’s activation’s of these features occurs through pressing keys that activate DTMF signaling. Figure 12.15 shows how computer telephony system can be used to create virtual (simulated) call attendants. In this diagram, a call is received to the main telephone number of the company to the computer telephony board. The automated telephony call processing software detects a ring signal, answers the phone (creates an off-hook signal) and plays a pre-recorded messaging informing the caller of options they may choose to direct the call to a specific extension. In this example, the automated call attendant software decodes DTMF tones or limited list of voice commands to determine the routing of the call. The automated call attendant software then determines if the destination choice is within the option list and if the extension is available. If the extension is available, the automated attendant will send a command to the computer telephony board switching the call to the selected extension. If the extension is not valid or not available, the automated attendant will provide a new voice prompt with updated information and additional options.

Figure 12.15: Automated Attendent Operation

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. Figure 12.16 shows a sample automatic call distribution (ACD) system that uses an interactive voice response (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 customer service representative

(CSR). This diagram also shows that the ACD system may also transfer customer or related product information to the CSR.

Figure 12.16: Automatic Call Distribution (ACD) Operation

Fax On Demand (FOD)
Fax on demand (FOD) is a telecommunications transmission service that sends previously stored faxes to a user that has requested a specific fax message. For FOD service, the caller listens to the options available and requests that additional information is delivered by fax. FOD service is often used to deliver previously stored instructions or marketing materials. Figure 12.17 shows how a fax on demand (FOD) system can store and automatically delivery faxes. The manager of this system has previously stored operating manuals in fax mailboxes at the company. Each fax document that is stored is assigned to a fax mailbox. The verbal (audio) name that is associated with this fax mailbox is programmed into the interactive voice response (IVR) system. This allows the user to hear their options and to make a fax mailbox selection. This example shows how a caller dials a telephone number that this company uses for automatic fax back service (555-2345 in this example). When the phone system receives a call on this line, it automatically routes the call to the IVR system at extension 1001. The IVR system prompts the caller to select the fax mailbox and to enter the destination fax number (555-8111). The fax mailbox (usually a storage area on the computer hard disk) retrieves the data and sends it to a computer fax generator. The destination digits (destination fax number) are also sent to the fax generator. This allows the computer to create the fax from the fax mailbox data and send the fax to the destination fax machine.

Figure 12.17: Fax on Demand (FOD) Operation

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 12.18 shows a sample IVR system that is used to route an incoming call. When this call is received by the PBX system, 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 12.18: Interactive Voice Response 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. The different types of systems used in private telephone networks include key telephone systems (KTS), private branch exchange (PBX), Centrex, and computer telephony integration (CTI). Key telephone and PBX systems often use proprietary specifications. There are several industry standards that are used for computer telephony and LAN telephony system.

Voice Mail (VM)
Voice mail is a service that provides a telephone customer with an electronic storage mailbox that can answer and store incoming voice messages. Voice mail systems use interactive voice response (IVR) technology to prompt callers and customers through the options available from voice mailbox systems. Voice mail systems offer advanced features not available from standard answering machines including message forwarding to other mailboxes, time of day recording and routing, special announcements and other features. Figure 12.19 shows how a voice mail system provides electronic storage mailboxes to users within the telephone system. In this example, the voice mailbox system connects to a switching system through 2 extensions (ports) on the switching system (other voice mail systems may have

many more ports). To access the voice mail system, users may select the voice mailbox system extension (usually programmed into a button on a telephone set that says voice mail). In this example, when a user dials into the telephone system to reach extension 1001, the line is busy. The system has been setup to forward calls to extension 1015 (the voice mail system) when extension 1001 is busy. To help ensure the voice mail system is accessible, if extension 1015 is busy, the call will be forwarded to extension 1016. When the call has entered the voice mail system, the interactive voice response (IVR) system will prompt the caller or user to enter information using touchtone or voice commands. This will allow callers or users to either store or retrieve messages from the digital message storage area (e.g. a computer hard disk drive).

Figure 12.19: Voice Mail Operation

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