STUDENT TEXT (ST-III) E3AQR3D132 01AC

DESIGNED FOR AETC COURSE USE NOT INTENDED FOR USE ON THE JOB

TECHNICAL TRAINING 

 

Cyber Transport Systems 

 

VOICE COMMUNICAITONS 

 

 

October 2015 

 

 

81 TRAINING GROUP

338 TRAINING SQUADRON Keesler AFB MS 39532

(U/FOUO) “For use only by elements of the U.S. Government, its contractors, or prospective contractors, in the conduct of official business.” Any other disclosure or release is strictly prohibited. This lesson contains information which may not be disclosed to international students without proper authorization. Refer to the foreign disclosure memorandum in the Security Annex of the Course Training Plan for further information

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Table of Contents      

UNIT 1 ................................................................................................................................................................................... 5 

PRINCIPLES OF TELEPHONY .................................................................................................................................................. 5 

OBJECTIVES ....................................................................................................................................................................... 5 

INTRODUCTION ................................................................................................................................................................ 5 

FUNDAMENTALS OF TELEPHONY ..................................................................................................................................... 5 

FUNDAMENTALS OF SOUND ............................................................................................................................................ 5 

Frequencies & Amplitude ............................................................................................................................................. 6 

LAND LINE CONCEPTS (6.3.15) ......................................................................................................................................... 6 

Fig 1‐1.  Example of a Landline ..................................................................................................................................... 7 

Dedicated Lines ............................................................................................................................................................ 7 

Tip and Ring .................................................................................................................................................................. 7 

TYPES OF TELEPHONES ................................................................................................................................................. 8 

Figure 1‐2.  2500 Telephone ......................................................................................................................................... 8 

Digital Telephones ........................................................................................................................................................ 9 

ISDN (Integrated Services Digital Network) .................................................................................................................. 9 

COPPER DISTRIBUTION FUNDAMENTALS ...................................................................................................................... 12 

DISTRIBUTION / TERMINATION BLOCKS .................................................................................................................... 13 

Figure 1‐6.  Protector Side of Vertical Block with Carbon Module Protectors ........................................................... 14 

SUMMARY .................................................................................................................................................................. 15 

TELEPHONY SWITCHING CONCEPTS ............................................................................................................................... 16 

CIRCUIT SWITCHING ................................................................................................................................................... 16 

Manual Telephone Exchange and Telephone Switchboards ..................................................................................... 17 

Automatic Exchange ................................................................................................................................................... 17 

Private Branch Exchange ............................................................................................................................................ 18 

Defense Switched Network (DSN) .............................................................................................................................. 19 

TRUNKING CONCEPTS ................................................................................................................................................ 20 

Access Interfaces. ....................................................................................................................................................... 21 

CALL ROUTING FUNDAMENTALS .................................................................................................................................... 22 

Video Teleconference (VTC) ........................................................................................................................................... 25 

Digital Compression. ................................................................................................................................................... 25 

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UNINTERRUPTABLE POWER SUPPLIES ........................................................................................................................... 28 

FACILITY BATTERY BACKUP ......................................................................................................................................... 29 

SUMMARY .................................................................................................................................................................. 30 

UNIT 2 ................................................................................................................................................................................. 31 

TELEPHONY MAINTENANCE PRACTICES ............................................................................................................................. 31 

OBJECTIVES ..................................................................................................................................................................... 31 

INTRODUCTION .............................................................................................................................................................. 31 

TELEPHONE TEST SET FUNDAMENTALS ..................................................................................................................... 34 

Inductive Amplifier & Tone Generator Fundamentals ............................................................................................... 34 

Figure 2‐10.  Tone Test Set ......................................................................................................................................... 35 

ISDN TELEPHONE TEST SET ......................................................................................................................................... 35 

ASYMMETRIC DIGITAL SUBSCRIBER LINE ................................................................................................................... 36 

LAN TEST SET .............................................................................................................................................................. 36 

OBJECTIVE ................................................................................................................................................................... 37 

FRAME CONNECTIONS ............................................................................................................................................... 38 

Punch‐down. ............................................................................................................................................................... 38 

WIRE WRAPPING ........................................................................................................................................................ 39 

Instructions ................................................................................................................................................................. 40 

TROUBLESHOOTING METHODS .................................................................................................................................. 41 

Half‐Split Method ....................................................................................................................................................... 41 

Substitution Method .................................................................................................................................................. 41 

“Circle the Wagons” Method ..................................................................................................................................... 42 

Simple to Complex Method ........................................................................................................................................ 42 

Instructions ................................................................................................................................................................. 42 

UNIT 3 ................................................................................................................................................................................. 43 

VOICE OVER IP (VOIP) ......................................................................................................................................................... 43 

INTRODUCTION .............................................................................................................................................................. 43 

VOICE OVER INTERNET PROTOCOL (VOIP) ................................................................................................................. 43 

Advantages of VoIP over the PSTN: ............................................................................................................................ 44 

Major Disadvantages of VoIP: .................................................................................................................................... 44 

VoIP Network Components ........................................................................................................................................ 45 

Call Processing Server ................................................................................................................................................. 45 

USER ENDPOINT DEVICES ........................................................................................................................................... 46 

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IP NETWORK COMPONENTS....................................................................................................................................... 49 

VoIP/Voice Gateway ................................................................................................................................................... 49 

NETWORK ROUTERS ................................................................................................................................................... 50 

REAL‐TIME TRANSPORT PROTOCOL (RTP) ...................................................................................................................... 50 

QUALITY OF SERVICE (QOS) ............................................................................................................................................ 50 

NETWORK SWITCHES ................................................................................................................................................. 51 

VOIP CALL ROUTING ................................................................................................................................................... 51 

VOICE OVER SECURE IP (VOSIP) .................................................................................................................................. 51 

SECURE VOICE OVER IP (SVOIP) .................................................................................................................................. 52 

SUMMARY .................................................................................................................................................................. 53 

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UNIT 1 PRINCIPLES OF TELEPHONY

OBJECTIVES a. Identify basic facts pertaining to Telephony Systems. b. Identify basic facts pertaining to Copper Distribution Systems. c. Identify basic facts pertaining to Telephony Switching Concepts. d. Identify basic facts pertaining to Critical Communications Facilities Power Systems.

INTRODUCTION The telephone has undergone an evolutionary process since Alexander Graham Bell discovered that sound could be converted into electrical impulses and transmitted over wire. The telephone continues to improve with technology and is no longer mass-produced and enclosed in a wooden box. Instead, it is becoming miniaturized with the proliferation of wireless and cellular telephones. We will focus on the analog single-line telephone set to provide you with the basic principles of how a telephone system works.

OBJECTIVE

a. Identify basic facts pertaining to Telephony Systems.

FUNDAMENTALS OF TELEPHONY FUNDAMENTALS OF SOUND Anything that you can hear is a sound. To produce sound, there must be a medium to carry sound waves from the source to the receiver. Sound transmits through three mediums, which are solid, liquid, and gas. Did you ever tie two cans to a string and talk to another person through them? That is an example of a solid (the string) carrying sound waves from one person to another. While swimming in a pool, you have most likely heard sounds while under the water which is an example of a liquid medium. Finally, the most common medium is obviously gas, or air, which is present for the direct transmission of sound from one person to another.

There is a limitation to the transmission of sound, and that limitation is distance. No matter how loud you yell, a person in Biloxi cannot converse with a person in New Orleans. The telephone solves that problem of distance limitation on transmission from point to point. The services provided by telephones reach almost everywhere whether supporting airmen or a command

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general, whether at a desk or in the field. They can simply pick up a telephone and communicate with an associate in the next room, or even talk to friends and loved ones back home when they are deployed.

The sound of the speaker’s voice is not what is actually transmitted over a long distance, but a sound like the speaker’s voice is generated to the distant point by means of electrical power. The voice power of the speaker is transformed into electrical impulses, which are transmitted to the distant end, where they are changed back into sounds that resemble the voice of the speaker.

Frequencies & Amplitude The sound of the normal speaking voice contains fundamental frequencies between 300 and 3400 Hz. The total range of hearing in a healthy young adult ranges from 16 to 20,000 Hz. For transmission of ordinary conversation, it has been found that a sufficiently high degree of intelligibility can be achieved if the frequencies are limited to 200 to 2,700 Hz. In order to transmit satisfactorily, telephone equipment is normally designed to operate within this range.

From the standpoint of voice communication, two very important qualities affect the design and construction of telephones, telephone lines, and equipment: frequency and amplitude. The frequency determines the vocal range - the highest and lowest pitches that a person can produce. Amplitude is the fullness or quality of the sound. For a telephone system to do its job right, it must electrically transmit and reproduce sound as nearly perfect as possible.

Every effort must be made to keep distortion, either electrical or mechanical, to the lowest point possible. Poor connections reduce amplitude. Faulty components of a telephone can reduce frequencies and amplitudes to such an extent that speech is not understandable. You, as the cyber transport technician, are responsible for many of these components.

LAND LINE CONCEPTS A landline, main line or fixed-line is a telephone line which travels through a solid medium, either metal wire or optical fiber (Figure 1-1). The physical characteristics of the communications link (material type, diameter, layout of conductors) determine its transmission bandwidth. This is distinguished from a mobile cellular line, where the medium used are the airwaves (wireless). Landlines usually cost less than cellular lines and provide better voice quality, and are used when there is no need for mobility or where cellular service is unavailable.

Each land line is supplied two different voltages from the telephone switch equipment. The first is 90V AC which is used to signal the called party that there is an incoming call. The second is 48V DC which is used to supply power to the line while the circuit is connected carrying the voice signal down the line. 90V AC is known as “ring voltage” due to the fact that it typically “rings” a bell or triggers a sound at the distant end while -48V DC is known as “talk voltage” due to the fact that it’s used to carry the voice signal down the line.

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A landline is also used to increase the security of communications, as it cannot be intercepted by a receiver without physical access to the line. (This does not, however, mean that a telephone company will not send the call over the air on some point in the journey.)

Figure 1-1. Example of a Landline

In 2003, the CIA reported approximately 1.3 billion main telephone lines worldwide. China had more than any other country, at 350 million, and the United States was second with 268 million – this compared with 219.4 million cellular telephones, a number which is expected to exceed the landline number within a few years.

Dedicated Lines Remember Commissioner Gordon calling Batman on the Bat phone? This was the commissioner’s dedicated line directly to Batman when there was trouble in Gotham City. All he had to do is pick up the receiver and the phone automatically rang at Wayne Manor. Dedicated lines work like the Bat phone. They are landlines that are used for only a specific purpose. The Air Force utilizes dedicated lines for all kinds of communications from direct lines to fax lines to modems. They are connections that are always available between two locations 24 hours a day by a designated user or company. This connection is provided on a dedicated line, rather than the public, or dialed up line shared among multiple users. So, in essence, a dedicated line is known as private, leased line that guarantees constant bandwidth availability and latency. These lines are used only for its intended use.

These land lines are transmitted using a Bell System Standard established in the 1960s, and today it is still a popular method of transmitting digitized information onto mediums such as: optical fibers and twisted pairs. Typical twisted copper pair size is 24 gauge and the most common connection used because it’s the cheapest solution. Optical fiber is the cleanest way to transmit audio and digital information and fiber links offer over 1,000 times as much bandwidth over distances 100 times further. The major disadvantage is that it cost 20% more than copper cabling

Tip and Ring Each subscriber’s telephone is connected to a central office through a DC loop of two wires called a twisted pair. One of the wires is called “tip” and the other is called “ring.” Telephone switches in the Central Office (CO) respond to the dial pulses or tones from the subscriber’s telephone to connect the calling party to the called telephone. When the connection is established, the two telephones communicate using the battery source supplied by the CO

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batteries. The terms Tip and Ring are holdovers from the original telephone systems. Today they are used to identify polarity of the wire pair. This polarity must be observed or the communication process will not occur. Any two telephones may be interconnected for two-way conversation.

A telephone system includes individual telephone stations, outside plant cable that connects telephone stations to a telephone exchange or interconnects another telephone exchange, and all the equipment required for making connections between telephones. The interconnection of subscriber lines is done best at some point in the center of an area over which lines are distributed.

The equipment installed at this point is the telephone switch. Whatever the size and extent of the system, any telephone station can be connected with any other station. Each telephone switch is interconnected with other telephone switches through special links called trunks. Trunks allow a subscriber to make a telephone call to another station halfway around the world. You will learn more about trunks later.

TYPES OF TELEPHONES Plain old telephone system (POTS) is a term meaning analog telephone. There are many different types of telephones on the market and used at each installation, but we will focus only on two types: the single-line desk phone and the single-line wall phone. Additionally, we will briefly go over the M5009 Business set digital telephone that is replacing most analog telephones.

1. Single-Line Desk Phone. A single-line Dual-Tone Multi-Frequency (DTMF) desk telephone is also called by its model number: a 2500 set. It has the capacity to handle one central office line, and is connected to the wall jack by a line cord. A typical 2500 set can be seen in Figure 1-2

Figure 1-2. 2500 Telephone

2. Single-Line Wall Phone. A single-line DTMF wall telephone is also called by its model number: a 2554 set. The 2554 set is used mainly as entry phones and in

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warehouses on most military installations typically enclosed in a box or weather proof enclosure. A typical 2554 set can be seen in Figure 1-3.

Figure 1-3. 2554 Telephone

Digital Telephones The M5009 Business Set (Figure 1-4) is the basic digital telephone installed by Cyber Transport technicians. It offers a choice of selected key and system features. The M5009 has three fixed feature keys, a dial pad consisting of 12 fixed keys, eight programmable features and/or directory number keys with LCD indicators. In addition, one programmable feature key without an LCD indicator (key # 9, in topmost position) and it is equipped with a speaker for alerting tones, hands free dialing, and intercom. The M5009 Business Set works with a maximum loop length of 4,572 m (15,000 ft.) of 26 AWG standard twisted pair telephone wires.

Figure 1-4. M5009 Business Set

ISDN (Integrated Services Digital Network) Read the following sections in your student text.

Mike Meyers’ Certification Passport

CompTIA Network+

ISDN pages 253-255

ISDN is designed for a digital telecommunications network. This network promises to carry voice, data, and video on a single network via a single interface (in terms of both hardware and

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communication protocols) for connecting devices such as; telephones, faxes, computers, or video teleconferencing (VTC) equipment. ISDN provides the speed, accuracy, and flexibility of digital services over existing communications infrastructure; making it possible to combine both voice and data signals over standard phone lines originally designed for analog services. The slow evolution of ISDN has made it unpopular in the civilian sector and is widely considered outdated technology. It is however, the DOD standard protocol for connecting secure telephone units (STE)’s, telephone switch trunks, and VTC.

Secure Terminal Equipment or STEs (Figure 1-5) play a vital role in military operations. The STE Data Terminal provides a reliable, secure, high speed digital data modem for applications where only data transfer (FAX, PC files, Video Teleconferencing, etc.) is required. In military operations, the STE terminal provides secure voice over the PSTN using a removable cryptographic engine called an Enhanced Crypto Card (ECC)/KSV-21, which is provided separately. The Secure Terminal Equipment is available in three configurations:

1. Office: STE Terminal provides access to ISDN (Integrated Services Digital Network).

2. PSTN: This provides basic call capabilities over the PSTN.

3. Tactical: STE Terminal provides access to ISDN, PSTN, and serial EIA-530A/EIA-232 BDI (Black Digital Interface) telecommunications systems.

The STE is divided into four main components:

1. STE (Base Assembly) 2. Handset 3. Wedge 4. Crypto card.

Figure 1-5. STE Main Components

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Enhanced Crypto Card (ECC)/KSV-21. The KSV-21 Enhanced Crypto Card is a tamper resistant reprogrammable PC card approved by the National Security Agency to provide encryption functions and key storage for STE secure telephones and other devices. The U.S. version of the KSV-21 is certified to protect classified data up through the Top Secret/SCI level as well as unclassified sensitive information.

SUMMARY

Now that we just started on the basic of a plain old telephone system, how sound can be transferred through a solid medium and we can even mess with the quality with frequency and amplitude, let’s gear up for more of how the transmission is sent though the solid mediums.

Review Exercise 1‐1 

1. What is the solid medium for transmission of a telephone line? 

 

2. What is an advantage of fiber over copper mediums? 

 

3. What is the limitation of sound? 

 

4. What does frequency determine? 

 

5. What does amplitude determine? 

 

 

6. Why do we use the term Tip and Ring and what does that identify to today’s technicians?  

 

 

7. What is the model number for a single line wall telephone? 

 

 

8. What is the model number for a basic digital telephone?  

 

9.  What are the four main components of the secure terminal equipment? 

 

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INTRODUCTION

Ever walk or drive over a manhole cover in the street. Manholes offer access to many types of services like sewer access, electrical grid access, and telephone cable access. Typically the manhole covers will be labeled with the service that can be accessed in the manhole. In telephony, the manhole system offers a way to route customer cabling underground.

OBJECTIVE

b. Identify basic facts pertaining to Copper Distribution Systems.

COPPER DISTRIBUTION FUNDAMENTALS The telephone distribution system is the physical path telecommunication circuits take from the central office (CO) to the customer location. Telephone lines, for example, originate from a telephone switch. Each line is physically connected to the horizontal side of a main distribution frame (MDF) (discussed later in this block). The line is then connected to the vertical side of the MDF via jumper wire. The vertical side connects the large scale communications cable plant (copper and fiber) to the distant buildings within the service area. This is known as outside plant since it travels outside of the CO.

Most of these cables contain many hundreds to thousands of pairs, or paths, which exit the CO underground and spread out to key secondary distribution points. These secondary points are usually additional Main Distribution Frames (MDF) or manholes where the large scale cables are scaled down to smaller cables that spread out to buildings or tertiary (or third) distribution points. Once inside a building, the cable is connected (in a communications closet or room) to a distribution block where the lines can be further distributed throughout the building to individual customers.

The wiring within the building is known as inside or premise wiring. The same process is then repeated inside of the building as outside; where larger inside cables distribute groups of circuits to separated areas of the building and scaled down to service individual customers. The larger cables are known as house cables and usually terminate to a 66 block on a satellite board. These boards can be located in a closet, room, or enclosed on a wall. Additional 66 blocks (known as station blocks) are located on the board that connects the small scale wiring, such as Category III or V, which services individual customers. A jumper wire is used to connect the desired line from the outside plant cable block to the block corresponding to the customer’s location.

At the customer location, CAT3 or CAT5 cable is terminated to the back side of a RJ-11 or RJ45 modular jack which is attached to the wall. This wall jack is where the telephone device plugs into the telephone distribution system.

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DISTRIBUTION / TERMINATION BLOCKS Horizontal/Vertical termination blocks are in essence terminal boards. Terminal boards are insulated bases or slabs provided specifically to mount a group of wiring terminals. Terminal boards provide a convenient means of connecting individual electrical wires without a splice or physically joining the ends and are one of the most flexible types of electrical connector available

The Main Distribution Frame (MDF) has two types of terminal boards: the horizontal side and the vertical side. The horizontal or switch side of the MDF contains horizontal blocks which are used to terminate cables coming from the switch equipment. The vertical or customer side contains the distribution blocks which are used to terminate cables coming into the CO from customer locations, i.e. offices and houses. Technicians use jumper wire in pairs to extend connectivity between the two sides of the MDF.

Horizontal Blocks. Horizontal blocks are connection points that extend circuit connectivity (Dial tone) between the switch equipment and the vertical side of the MDF. The horizontal block has two sides. The underside provides Amphenol cable connector receptacles for the cables coming from the switch equipment. The topside contains wire-wrapping pins where jumper wires are connected to extend the dial tone towards the vertical side of the MDF. A switch that can provide 10,000 line numbers of service would require many horizontal blocks.

Vertical Blocks. Vertical blocks come in a variety of shapes and models. Regardless of the model, the block has at least three distinct sides:

1. Equipment side 2. Protector side 3. Test Field side.

The equipment side refers to the area of the vertical block where jumper wire is attached and subsequently connects back to the switch equipment. The TIP and RING pairs are numbered from left to right starting at the top left of the block. Each cable entering the CO from a customer location is represented by a pair of connections. One tip wire and one ring wire.

Opposite the equipment side of the vertical block is the protector side. The protector side holds the carbon module protectors for each tip and ring circuit. The carbon modules help protect switch equipment from surge voltage or current damage that may come from lightning or downed power lines. Carbon modules (Figure 1-6) contain five pins: two incoming pins, two outgoing pins, and one grounding pin. These pins mate with their matching receptacle. The equipment side pins are hard wired to the incoming pins of the protector module and the

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outgoing pins are hard wired to the test field jacks. The carbon modules are numbered from left to right starting at the top, left hand corner of the block.

Figure 1-6. Protector Side of Vertical Block with Carbon Module Protectors

The front of the vertical block contains the test field jacks. The test field jacks allow a technician easy access to cable pairs during cable or circuit testing procedures. The rear of the test jack contacts are internally connected to the cable coming from the customer's building. Therefore, when a carbon module is removed, a cable pair can be tested without interference from the switch equipment. Conversely, the equipment side of the block can also be tested without interference from the cable or customer equipment. The test field positions are numbered from right to left starting at the top, right hand corner of the block.

The normal path of current flow within the MDF is as follows: Dial tone leaves the switch equipment, goes through the horizontal block, and then continues on a jumper wire connected to the equipment side pins of the vertical block. These pins are connected to the incoming pins of the carbon module receptacle. With a carbon module plugged in place the dial tone will pass through and exit the outgoing pins of the carbon module receptacle. The receptacle connects to both the rear contacts of the test field jack pins inside the vertical block and the cables leaving the CO. This extends the dial tone towards the customer as well as to the test field jack. If dial tone is not present at the test field jack, replace the module with a spare (most work centers will have a box of spares readily available) or check your switch equipment. Do NOT use an existing module (one currently in service). Doing so could possibly disconnect an active circuit.

66 Block. The terminal arrangement of a 66-type connecting block varies according to the type of block. The quick connecting terminal blocks have terminals that are designed to eliminate the need for soldering, wire wrapping, and the use of screw-type terminals. Connecting blocks also

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eliminate the need to strip each individual wire. In figure 1-7, you can view a row of connecting block terminals from the side. The wire (with insulation) is placed into the notch in the terminal. An impact tool (punch-down is used to push the wire to the back of the terminal and cut off any excess wire, all in one motion. Typically these blocks are found in the customer’s building for plain old telephones connections.

Figure 1-7. 66 Block

A 110 block is another type of quick connection terminal punch block used to terminate runs of on-premises wiring in a structured cabling system. The designation 110 is also used to describe a type of insulation displacement contact (IDC) connector used to terminate twisted pair cables, which uses a punch-down tool similar to the older 66 block. Mainly, they are used for data connections, such as computers to switches. Figure 1-8 shows a standard 110 block.

Figure 1-8. 110 Block

SUMMARY

It takes a whole underground, telephone distribution system, to move dial tone out from the CO to the customer. With the aid of the horizontal, vertical, 66, and 110 blocks technicians can extend service to the customer without having to solder each jumper connection. Additionally, it provides a quick and convenient way to move connections from one location to another without making major changes to the distribution system.

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Review Exercise 1-2

1.  What are the two sides of a MDF? 

2.  How many sides does the horizontal block have? Describe the purpose of each. 

 

3.  What is used to connect the two sides of the MDF? 

 

4.  What is a carbon module and what is its purpose?    

 

INTRODUCTION

Understanding how the telephone switch operates is step one to troubleshooting and maintaining telephony systems. In this objective, we will discuss the evolution of the telephone switch or exchange from start to finish and explain how the telephone switch connects a call from one calling party to another.

OBJECTIVE

c. Identify basic facts pertaining to telephony switching concepts.

TELEPHONY SWITCHING CONCEPTS  

Read the following sections in your student text.

Mike Meyers’ Certification Passport

CompTIA Network+

PSTN page 248-253

CIRCUIT SWITCHING Circuit switching is a communications method where data messages are transmitted over one dedicated path for the duration of the call. To make this possible, a dedicated line is selected for transmission between two parties. For decades, telephone services have run over circuit switching networks such as PSTN. This method uses a large amount of electricity and takes many people to maintain the switching facilities. Circuit switching uses a series of selectors in order to route the call from the switch to the customer over dedicated lines. Each selector

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represents a different end location for the call to terminate. In order to complete a telephone call, the telephone switch was developed. Simply put, a telephone switch is a system of electronic components that connect telephone calls. Since Alexander Graham Bell first invented the telephone in 1867 there have been two main types of telephone switches developed; the Manual Telephone exchange and the Automatic Telephone exchange.

Manual Telephone Exchange and Telephone Switchboards The telephone switchboard was a communications system used in the PSTN for interconnecting circuits in order to establish connection between telephone users. The telephone switchboard was a critical component of the manual telephone exchange and was operated by one or more telephone switchboard operators. Users dialed directly to the switchboard where they told the operator who they wanted to call or what number they wanted to reach. The telephone switchboard operator would then complete the circuit by placing a cable into the switchboard. Although the manual switch and switchboards were replaced by automatic switches, manual switches continued to be used through the 1970’s. (Figure 1-9)

Figure 1-9. Manual Switch Operator NORAD

Automatic Exchange The first automatic switch was invented by Almon Strowger (an undertaker) in 1888 and patented in 1891. Its purpose was to eliminate the need for human switchboard operators to complete the call. Starting in 1919 the Bell Telephone Company adopted the automatic switch that brought the beginning of the end for the manual switch and switchboard operators. The automatic switch used a series of selector based electrical switches in order to complete the circuit. The user was provided a dial pad in order to input digits directly to the telephone switch. Based on the dialed digits or selectors, a series of electrical switches would open or close completing the circuit from end to end.

Each user or phone is designated by a series of numbers known as a Directory Number (DN) or a telephone number. A US based DN consists of three sets of numbers:

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1. Three-digit area code 2. Three-digit Central Office code 3. Four digits for the line number

In total, a modern telephone exchange is limited to one Central Office code or 10000 users (includes 0000-9999) Figure 1-10 shows selectors used in basic circuit switching.

Figure 1-10. Basic Circuit Switching

Private Branch Exchange A PBX (private branch exchange) is a telephone system within an enterprise that switches calls between enterprise users on local lines while allowing all users to share a certain number of external phone lines. The main purpose of a PBX is to save the cost of requiring a line for each user to the telephone company's central office.

The PBX is owned and operated by the enterprise rather than the telephone company (which may be a supplier or service provider). Private branch exchanges used analog technology originally. Today, PBXs used in digital technology but the digital signals are converted to analog for outside calls on the local loop using plain old telephone service (POTS).

A PBX includes:

• Telephone trunk (multiple phone) lines that terminate at the PBX • A computer with memory that manages the switching of the calls within the PBX and in

and out of it • The network of lines within the PBX • A console or switchboard for a human operator (optional)

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Customer groups

Customer groups or Device pools in Voice over IP (VoIP) allow you to assign the same configuration to a group of devices; for example, you can assign the device pool to phones, gateways, or trunks. In general, device pools allow you to configure common parameters that need to be applied to a device. For phones, you may need to configure the device pool, the common phone profile, and the common device configuration, which work similarly to device pools (that is, they allow you to apply the same configuration to a group of phones). The device characteristics you can specify for a device pool are:

• Region • Date/time group • Cisco CallManager group • Class of service

Caller ID

Caller ID is a telephone service available on most telephone system, including VoIP that transmits the callers Directory Number, callers’ name, callers address, and other information programmed for that line by the telephone administrator.

911/E911

Most people are familiar with the emergency response number but do you know how it works? Technicians working on commercial or military telephone system must be familiar with both the 911 emergency response system and the newer Enhanced 911 (E911) systems.

- 911: the emergency telephone number for North America. From any hard line

telephone, dialing 911 will connect the caller to the closest emergency dispatch center called a Public Safety Answering Point (PSAP). The PSAP then dispatches the appropriate agency to deal with the emergency.

- Enhanced 911: approximately 96% of the US uses Enhanced 911 call systems. These systems use the telephone company’s database of phone numbers and addresses to present the PSAP with the location of the emergency. Mobile phones associate the location of the placed call using Global Positioning (GPS) and Radiolocation. Radiolocation provides the tower location and direction the call came from and the GPS provides an estimated location.

Defense Switched Network (DSN) The Defense Switched Network is the primary provider of long distance communications services for the Defense Communications System (DCS). DSN provides worldwide secure voice, non-secure voice, data, and Video Teleconference (VTC) services for the Department of Defense and its subordinate agencies. The DSN assures connection availability during times of crisis with the use of Multi-Level Precedence and Preemption (MLPP). The DSN switching subsystem consists of multifunction, standalone tandem, end office, and remote switching units.

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Using the transmission, timing, and control elements of the DISN, they interconnect all military locations worldwide and provide end-to-end long-distance common user and dedicated voice, secure voice, data, and video services worldwide. The use of the DSN is restricted to the following:

1. The use of precedence level consistent with the subject matter of the call. 2. Essential official calls requiring a timeliness that cannot be obtained by other means. 3. The minimum time required to accomplishing official business.

TRUNKING CONCEPTS Trunking in general is a concept which communication systems provide access to multiple devices by sharing a group of lines or connections. In Telephony, trunking refers to the lines that interconnect one central office with another. In most cases, fiber optic cable is used to provide the most bandwidth between the COs. The trunks are formatted in various ways like DS1 or T-1, in order to separate a single line into multiple channels, each channel carrying a separate phone call. A CO will have multiple trunks connected to different COs or carriers depending on the location of the CO. For example, the Keesler AFB CO will have a trunk to the local CO in Biloxi and have a trunk to the DSN network in order to switch calls.

Digital Signal

Digital signal refers to a basic digital signaling rate of a voice channel. For example, a DS0 is a 64kb (0.064 Mb) voice channel sampled at 8 kHz (8000 times per second) using 8bit pulse code modulation for each sample. The DS0s’ fundamental role is to carry a single telephone call and therefore forms the basis for digital multiplexing signals. If 24 DS0s are multiplexed together they will form a DS1. 96 multiplexed DS0s form a DS2 and so on. (Table 1-1)

Digital Signal Number of

Channels Designation of Carrier

Bandwidth (Mb/s)

DS0 1 NONE 0.064

DS1 24 T1 1.544

DS2 96 T2 6.312

Table 1-1. Common Digital Signal

T-Carrier

For information on T-1 and T-3 T-Carrier services read the following chapter in your student text.

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Mike Meyers’ Certification Passport

CompTIA Network+

Copper Carriers: T1 and T3, pages 235-239

Access Interfaces. There are two main access interfaces: Basic Rate Interface (BRI), and Primary Rate Interface (PRI). BRI is used for line services such as STE and Video Teleconferencing (VTC). PRI is used for trunk services between voice switches.

Basic Rate Interface (BRI)

BRI consists of two 64 Kbps B-channels and one 16 Kbps D- channel (Figure 1-11). Thus, one BRI circuit can dedicate 128 Kbps toward voice and or data transmission on a single copper cable pair. Copper lines can use a maximum rate of about 160Kbps. BRI is the minimum required Integrated Services Digital Network (ISDN) access interface used for VTC.

Figure 1-11. Basic Rate Interface (BRI)

Primary Rate Interface (PRI)

PRI consists of 23 64Kbps B-channels and one 64 Kbps D-channel totaling 24 channels. Thus, a PRI user can have up to 1, 544 Kbps. PRIs are primarily used for trunks. (Figure 1-12)

Figure 1-12. Primary Rate Interface (PRI)

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CALL ROUTING FUNDAMENTALS Call routing refers to the process a switch goes through to decide how a call gets from one customer to another. Often this involves calculating the best way or clearest trunk between COs. There are many methods a switch can use to make the best decision some of which are discussed below.

DTR (Dynamic Trunk Reservation)

The purpose of dynamic trunk reservation in public switched telephone network is to calculate the cost of a call as it’s routed through the switch. The switch will calculate the “cost” of the call over a selected route and determine if there is a better “lower cost” route to take. For example, a user places a call to Dallas Texas over trunk connection A. The “cost” of the call is 10. The switch then calculates all other available routes to determine if 10 is the lowest cost available over the programmed trunks. This feature comes in handy when the path for a particular network is heavily loaded.

Direct Inward Dialing (DID)

Direct Inward Dialing refers to the ability for a telephone number to be reached directly without interaction from the telephone switch operator or other system. This technology was developed in order to eliminate the need for a switch attendant to manually connect a call. As technology developed, DID became the standard and is now included in all telephone switch systems as a default service.

MLPP (Multi-level Precedence and Preemption)

Multi-level Precedence and Preemption (MLPP) service allows validated users to place priority calls, and if necessary, to preempt lower-priority calls. Precedence indicates the priority level of a call. Preemption is the process of terminating a lower-precedence call so a call of higher precedence can proceed. This capability assures high-ranking personnel can communicate with critical organizations and personnel during network stress situations, such as a national emergency or degraded network situation

- Precedence. Precedence indicates the priority level associated with an MLPP call. Phone users can apply a precedence level when making a call. Phone users request a precedence call by dialing the access code (NP), where N specifies the preconfigured access digit and P specifies the requested precedence level, followed by the phone number

- Preemption. Preemption is the process of terminating an active call of lower precedence so a call of higher precedence can proceed. Preemption includes the notification and acknowledgement of preempted users and the reservation of shared resources immediately after preemption and before call termination. Preemption can take one of the following two forms:

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1. User Access Preemption-this type of preemption applies to phones and other end-user devices. If a called party is busy with a lower precedence call, both the called party and the party to which it is connected receive preemption notification and the existing call is cleared immediately.

2. Common Network Facility Preemption-this type of preemption applies to trunks. If all channels of a PRI trunk are busy with calls of lower precedence, a call of lower precedence is preempted to complete the higher precedence call.

Class of Service

Each line is assigned a Network Class of Service (NCOS) that controls the types of calls that can be made from that line. By definition NCOS is a classification assigned to users of a telephony network that gives them particular rights and privileges. In an internal telephone system, it can include the ability to make international calls or dial 800 numbers. In the public telephone network, it is used for call charge rate calculations and differentiates between categories such as residential or commercial, flat rate or message units, private line or party line. (Table 1-2)

Capability COS-10 COS-5 COS-3

World Wide X X X

Local Commercial/DSN X X

DSN Only X

Table 1-2. Class of Service Example

Translation

With each assigned telephone number it is plotted in the data base table. Tables and sub-tables consist of horizontal rows called tuples and vertical columns called fields. (Figure 1-13)

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Figure 1-13

In the database this is like the header when reading a chart.

• NPA is the area code the switch holds

• MAXRTE is the trunk route

• NOAMBIGC is the switch allocating the memory for ambiguous code for the NPA

• RTEREF the routing destination

• HNPACODE is the where the routing translation is stored

• ATTRIB is routing reference

• RTEMAP is ISDN routing reference

This is where a technician tells the switch that there is new hardware (like a new cabinet) in place and ready to use. This is also where a technician tells the switch what card is in the slot of the shelf (single line phone card, digital phone card, ISDN card).

 

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Figure 1-14. VTC

Video Teleconference (VTC) Videoconferencing is the conduct of a videoconference (also known as a video teleconference) by a set of telecommunication technologies which allow two or more locations to communicate by simultaneous two-way video and audio transmissions. It has also been called 'visual collaboration'. (Figure 1-14)

With the introduction of relatively low cost, high capacity broadband telecommunication services in the late 1990s, coupled with powerful computing processors and video compression techniques, videoconferencing has made significant inroads in business, education, medicine and media.

The core technology used in a videoconferencing system is digital compression of audio and video streams in real time using a codec. The digital stream of packets are then transmitted through a digital network such as ISDN or TCP/IP. The use of audio modems in the transmission line allow for the use of POTS in some low-speed applications, such as video telephony, because they convert the digital pulses to/from analog waves in the audio spectrum range.

There are four basic requirements in order for VTC to operate. Each providing a piece of the system. The four components are:

• Digital compression (CODEC) • Video Input / Output • Audio Input / Output • Data Processor (Computer)

Digital Compression. Codecs are used in VTC systems to compress the audio and video signals before transmission over a medium and to decompress the audio and video signals after they have been received by the distant end in order to preserve bandwidth. Your codec handles the encoding and decoding, or compression and decompression, at either end. Video Conferencing systems often detect repetitive images, such as a picture on the wall in the captured image, and reuse the already-sent

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data for encoding and decoding images instead of resending all the image data for every shot. This helps to minimize the amount of bandwidth needed for a Video Conference.

There are three main standards for VTC. ITU H.320, used for VTC over PSTN or ISDN. ITU H.264, used for IP based VTC systems. And ITU V.80 used in conjunction with ITU H.324 for standard Point-to-Point VTC over POTS lines.

Video Input/Output. Video input devices can be as simple as a webcam or as complicated as an HD video system. Each system has an input device depending on the cost and requirement of the systems. Most desktop systems use a webcam while most dedicated VTC units have a wide angle lenses on an HD video camera. Output devices can be a computer monitor or a TV depending on the application

Audio Input/Output. Audio input and output devices are simply a microphone and speakers. Some units have HD audio systems. The quality of the audio is based on the cost of the system and the CODEC used to transmit the audio signal.

Data Processor. Data processing devices are typically desktop computers with interfaces for audio and video devices. The data processing unit, or computer, is responsible for interconnecting all of the audio and video components as well as applying the CODEC for data transmission.

There are two kinds of videoconferencing systems: (Figure 1-15)

1. Dedicated systems have all required components packaged into a single piece of equipment, usually a console with a high quality remote controlled video camera. These cameras can be controlled at a distance to pan left and right, tilt up and down, and zoom.

They became known as Pan Tilt Zoom (PTZ) cameras. The console contains all electrical interfaces, the control computer, and the software or hardware-based codec. Omnidirectional microphones are connected to the console, as well as a TV monitor with loudspeakers and/or a video projector.

2. Desktop systems are add-ons to normal PCs, transforming them into videoconferencing devices. A range of different cameras and microphones can be used with the board, which contains the necessary codec and transmission interfaces. Most of the desktops systems work with the H.323 standard. Videoconferences carried out via dispersed PCs are also known as e-meetings.

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Figure 1-15. Desktop VTC system (left) and a Dedicated VTC unit (right).

SUMMARY

In the DOD, VTC technology is critical for base operations. Leaders have the ability to exchange information “face-to-face” instantly from any location around the world. As a Cyber Transport technician, you will be responsible for configuring and maintaining VTC units of all types used by units on a daily basis.

REVIEW EXERCISE 1‐3 

1.  What are the two access interfaces for ISDN? 

 

2.  What does a private branch exchange include? 

 

3.  How much bandwidth is available over a DS0? 

 

4.  How many DS0s are multiplexed in order to achieve a DS1 connection?   

  

5.    A T1 operates at what speed?       

  

 

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6.   How many channels are available over a T1 line? 

 

7.   The device characteristics you can specify for a device pool are: ___________________________________ 

 

 

8. ___________indicates the priority level of a call. ___________ is the process of terminating a lower‐

precedence call so a call of higher precedence call can proceed.   

 

9.  What is used in the public telephone network, it is used for call charge rate calculations and differentiates 

between categories such as residential or commercial, flat rate or message units, private line or party line? 

  

10.  What are the four requirements for VTC? 

 

11.  What are the two main types of VTC systems? 

INTRODUCTION

Most people have experienced a power outage at some point of their lives. In the military, system availability is critical to base operations. Base power systems run off of the commercial power offered by the local community. When a power outage happens the base losses power as well. For telephone and LAN systems, reliable power is a requirement. Most critical systems like LAN and telephone have a power backup system like Uninterruptible Power Supplies (UPS) or generators.

OBJECTIVE

d. Identify basic facts pertaining to Critical Communications Facilities Power Systems.

UNINTERRUPTABLE POWER SUPPLIES An Uninterruptible Power Supply (UPS) (Figure 1-16) is an automated external power supply designed to keep a server or other device running in the event of a power failure. The UPS system takes advantage of uninterruptible power supplies that can interface with the operating system. The two main purposes of an UPS are:

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• A power source to run the server for a short time

• A safe shutdown management service

Fig 1-16. Uninterruptible Power Supply

The power source is usually a battery, but the UPS can also be a gasoline engine running an AC power supply. If the power fails, users are notified of the failure and warned by the UPS to finish their tasks. The UPS then waits a predetermined amount of time and performs an orderly system shutdown. A good UPS system will:

• Prevent any more users from accessing the server.

• Send an alert message to the network administrator through the server.

FACILITY BATTERY BACKUP Have you ever experienced a power outage? Have you tried to use the phone when your house or business has a loss of power? Switching equipment must continue to operate even if commercial power should fail for any reason. The take-over from commercial AC to an emergency source is automatic as many central offices are unattended or only staffed during normal working hours. The two types of emergency power used by the central office are:

1. Battery Backup

2. Standby Generator.

RECTIFIERS

The Central Office (CO) receives power from a commercial AC power supply from the power company near the base, or from the base power plant. This electricity runs into a number of rectifiers in the Central Office. The number of rectifiers required depends on the size of the telephone office. The rectifiers are diodes placed in an alternating current circuit. Rectifiers convert Alternating Current (AC) into Direct Current (DC) and keep the battery cells charged.

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INVERTERS

A loss of commercial power to the telephone switch would also affect associated equipment such as the maintenance positions, printers, and cooling fans. All of this equipment runs off the same 115 VAC you use in your own home. To avoid downtime of this equipment, DC to AC inverters supplies 115 VAC at 60 Hz potential to auxiliary devices. The inverter receives -48 VDC from the Power Distribution Center (PDC) and converts that DC voltage to 115 VAC. The inverter also contains a fan for cooling. The inverter features an external low voltage alarm, test jacks, circuit breakers, and an alarm lamp mounted on the faceplate. The inverter also contains four output AC receptacles located on the back of the inverter. These receptacles are the outlets that you use to plug in maintenance terminals and printers. The number of inverters installed in a switch will depend on the number of AC driven devices necessary to operate the switch during a power outage.

GENERATORS

The ampere hour rating, battery condition, charge level at the time of the AC failure, and the office load requirements determine how long the battery system will operate. Most central offices without standby generators require approximately 8 hours of reserve time when sizing office batteries. Those with a standby generator may range from three to 5 hours of reserve time.

A power outage would also affect the associated switch equipment such as the maintenance positions, printers, and cooling fans. All of this equipment runs off the same 115 VAC you use in your own home. To avoid downtime of this equipment, DC to AC inverters are used to supply 115 VAC at 60 Hz potential to the auxiliary devices. The number of inverters installed in a CO will depend on the number of AC driven devices necessary to operate the switch during a power outage.

SUMMARY

Backup power systems will be an important part of the overall system. Knowing which systems have backup power will enable you as a Cyber Technician the ability to prioritize systems in the event of loss of power. Knowing the basics on how these power systems work will make you a better technician.

REVIEW EXERCISE 1‐4 

1.  Power equipment within a central office consists of:   

2.  What is the purpose of a rectifier? 

3.  What two crucial components does an Uninterruptable Power Supply provide a network in the event of a 

commercial power failure? 

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UNIT 2 TELEPHONY MAINTENANCE PRACTICES

OBJECTIVES a. Identify basic facts pertaining to specialized tools. b. Identify basic facts pertaining to specialized test equipment. c. Using supplied publications, materials, specialized test equipment and tools, complete a circuit

in accordance with Progress Check 3.2c. d. Using supplied publications and digital equipment; configure a Telephone Switching circuit in

accordance with Progress Check 3.2d. e. Using supplied publications and digital equipment; troubleshoot connectivity problems in a

Telephone Switching circuit in accordance with Progress Check 3.2e.

INTRODUCTION While working in this career field you will find yourself working with various types of tools and test equipment. Each tool is designed to be used for a specific purpose. If you use a tool for which it is not designed to do, you may cause damage to equipment or to the tool itself.

OBJECTIVE a. Identify basic facts pertaining to specialized tools.

Impact Tool. The impact tool in figure 2-1 is more commonly known as the “punch-down tool” and is designed to terminate (connect) wire to both 66-type and 110 type quick connect blocks. Simply place the wire in position and put the tool over the split terminal. Then with slight pressure, push the wire into the terminal. The cutter will trigger automatically and cut the excess wire. Some style tips are reversible in order to work both type blocks. Other style tips will allow wires to be terminated without cutting the excess wire. You should never hit the back of the impact tool as this will damage the terminal block.

Figure 2-1. Impact Tool

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Amphenol Tool (50 Pin AMP Butterfly Crimper). A standard of the telephone industry for many years; and is the only hand-operated tool (Figure 2-2) capable of performing volume installations of 25-pair (50-pin) telephone connectors. Amphenol connectors are used primarily to terminate 25 pair cables to modular 66 blocks, MDF Horizontal blocks and other auxiliary equipment.

Figure 2-2. Amphenol Tool.

Wrap/Unwrap Tool. This tool is used to make and remove solderless wire connections on terminal pins found on both horizontal and vertical blocks. Often, the wire wrap and unwrap functions are separated into two separate tools. Figure 2-3 shows a combination wrap and unwrap tool.

Figure 2-3. Wrap/Unwrap Tool

Cable Cutting Tool (Butting Tool). The butting tool shown in Figure 2-4 is to ring-cut the sheathing of inside type plastic, fabric, or rubber-covered cable. The cable is pushed into the tool against a spring and backpressure holds it against the cutting blade for an accurate cut when the tool is rotated around the cable. Rotate the tool only once around the cable to avoid cutting the conductors inside the sheathing.

Figure 2-4. Cable Ring Cutting Tool

Wire Strippers are used to cut and remove the insulation on a wire without damaging the metal conductor. Typically, wire strippers have several different gauged slots and a blade used to cleanly cut wires and cables. Additionally, many models offer a plyer like area for gripping different gauge wire. Shown in Figure 2-5

Figure 2-5. Wire Stripper

Spudgers (Figure 2-6) taper to a flat-notched end to remove solder from terminal blocks and push wires into place. The hook side is used to separate and/or remove individual wires without damaging individual conductors, and can be used to pick up wires caught in small places.

Figure 2-6. Spudger

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Electrician's Scissors (Snips). The electrician's scissors in Figure 2-7 are designed to cut and strip telephone wire. It is made of strong, tempered steel with a scraper and file on both sides. One blade is serrated while the other is notched to strip most 19-guage and 24-gauge wire and insulation. It also features rounded safety points and an easy-opening screw hinge.

Figure 2-7. Electrician’s Scissors

Diagonal Pliers. Diagonal pliers in Figure 2-8 can be plain or notched. You use them for cutting and stripping small gauge wire.

Figure 2-8. Diagonal Pliers

REVIEW EXERCISE 2‐1 

1.  What tool is needed for a terminating wire on a 66/110 block? 

  

2.  How many pairs are needed to use the Amphenol tool correctly? 

  

3.  What termination block would utilize the wrap/unwrap tool? 

  

4.  What tool(s) can I use to strip insulation off wire? 

 

5.  What is the purpose of the notches on the electrician’s scissors? 

  

6.  What does the MON position on the telephone test set allow a technician to do? 

  

7.  How many positions are on a tone test set and what are they? 

 

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8.  What contains a speaker that amplifies the tone sent over the wires by the tone test set? 

  

OBJECTIVE

b. Identify basic facts pertaining to specialized test equipment.

TELEPHONE TEST SET FUNDAMENTALS Telephone Test Set (Butt Set). The telephone test set in Figure 2-9 is a handheld portable telephone used to verify the condition and quality of an analog circuit and performs like a single line telephone. However, the telephone test set is furnished with test clips designed to attach easily to 66-type terminal blocks instead of plugging directly into a wall jack. The telephone test set has a toggle switch with two positions: TALK and MON (monitor). Always access circuits in the MON position; this will allow you to determine if the line is free. If it is free, switch into the TALK position. If you are on an active telephone line, you should receive dial tone from the central office and you can make your call or continue troubleshooting.

Figure 2-9. Telephone Test Set (Butt Set)

Inductive Amplifier & Tone Generator Fundamentals By design, the tone generator and the inductive amplifier are used together in order to test and trace circuits from one connection to another. In the Information and Telecommunications industry, the tone generator and the inductive amplifier are known as the “fox and hound”. In this case, the tone generator is the “fox” and the inductive amplifier is the “hound” because a technician uses the inductive amplifier to find the tone over a line. The two pieces of test equipment are discussed below individually.

Tone Test Set (Tone Generator). The tone test set contains a 9V battery and is used to locate lines by generating a tone. It is equipped with a three-way switch: TONE, OFF, and CONT (continuity). When set in the tone position, the tone test set will generate a distinctive tone on the wires to help trace the cable. The CONT position can provide talk battery between two

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points on an unused pair. The CONT position can also check for shorts or opens using a Light Emitting Diode (LED), and in some models, an audible tone to indicate a short condition. A bright LED indicates a short or active line, and an off LED indicates an open. A flashing LED indicates a ringing line. Once a tone generator is plugged into the customer’s telephone jack, a technician can then go to the communications room and locate the tone (Figure 2-10).

Figure 2-10. Tone Test Set

Inductive Amplifier. The inductive amplifier is used with a tone test set to identify and trace wires within a group of wires without damaging the insulation. The inductive amplifier contains a speaker that amplifies the tone sent over the wires by the tone test set (Figure 2-11).

Figure 2-11. Inductive Amplifier

ISDN TELEPHONE TEST SET ISDN Telephone Test Set (ISDN butt set). Basic rate ISDN test set (Figure 2-12) is for testing, installation, and maintenance. Three modes enable easy analysis from the CO to all points at the customer location. NT1-TE mode provides Terminal Equipment emulation at the U interface for testing and fault isolation. TE mode tests the S/T interface between customer's network terminator and terminal equipment. LT mode provides dry-loop testing for ISDN prior to CO availability.

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Figure 2-12. ISDN Telephone Test Set

ASYMMETRIC DIGITAL SUBSCRIBER LINE ADSL Telephone Test Set. (Figure 2-13) Ability to simultaneously test Asymmetric Digital Subscriber Line services and verify POTS lines by checking attainable bit rate, channel bit rate, maximum line capacity, attenuation, output power, noise margin. As well as testing an automatic "snapshot" test for quick results, and a manual "monitor" test for up to 90 minutes of continuous testing

Figure 2-13. ADSL Telephone Test Set

LAN TEST SET LAN Test Sets (Figure 2-14) are manufactured by various companies and offer a wide range of capabilities. Many offer capabilities like ping testing, cable termination verification and network mapping. Some models offer a tone generator and network connectivity statistics in order to assist a technician with troubleshooting.

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Figure 2-14. LAN Test Set

Review EXERCISE 2‐2 

 

1. What action is performed after terminating circuit pair to the horizontal block?  

2. What termination block would utilize the wrap/unwrap tool?   

3. List the characteristics of a wrap connection.  

4. What is the maximum length of a shiner or pigtail?   

5. What are the two types of termination performed in the telephone circuit lab?  

6. In which direction do you wrap a wire around a pin and why?  

7. What is the maximum distance allowed between two wires wrapped on a single terminal?  

OBJECTIVE

2c. Using supplied publications, materials, specialized test equipment and tools, complete a circuit in accordance with Progress Check 3.2c.

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FRAME CONNECTIONS In a systems control facility, a frame is used to interface and cross connect wires and cables. In this unit, we are going to learn how to terminate or connect cables to a frame. We are going to learn about two methods of making frame connections, punch-down and wire-wrap.

Punch-down. The 66-Type and 110-Type connector blocks are designed for making fast connections of selected wires to terminals without the removal of the conductor insulation. The block will accept un-stripped No. 20 to No. 26 AWG and stripped No. 18 and No. 19 AWG conductors. The use of No. 26 AWG should be avoided because of low mechanical strength as it is easily broken.

Punch-down connections use a tool to force the wire into a slot to establish the electrical connection. This is the fastest method of making connections. The wire does not have to be cut to length or be stripped prior to making the connection. Figure 2-15 shows the use and placement of a punch down tool when terminating wires to a 66 block.

Figure 2-15

Punch-down connections remove or displace the conductor's insulation as it is seated in the connector. Punch-down connections use a Quick Connect Tool (aka Punch down or Impact) and Type 66 or 110 Connector Blocks. During termination, you press the cable between two edges of a metal clip, where the terminal penetrates the cable insulation as it is forced into place making contact with the copper conductor. The quick connect tool seats the wire tightly in the terminal slot and cuts the excess wire simultaneously. This ensures a solid connection between the copper conductor and terminating clip.

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While blades may look similar, they can be slightly different in design. If a technician terminates a jumper on a 66 block and needs to terminate the distant end to a 110 block they must first switch the impact tool blade from 66 to 110 in order to complete the install. Improper matching of termination tool handle/blade can lead to serious personal injury, as well as poor terminations.

WIRE WRAPPING Inside wiring cable is made for installation inside of a building and is not intended for wet or extremely damp locations. The plastic insulated conductors are color coded and protected by a lightweight plastic sheath. These are the cables you must be able to install and repair. It is your responsibility to learn how to identify pairs and binders so that you can properly terminate the Wrap Connections

A wrapped connection is designed to provide a good electrical connection without soldering. It consists of a series of close wraps of wire, placed under tension, around a specially designed terminal. The wraps are made using a wire wrap tool, and when properly done, provide the six basic characteristics of a permanent electrical termination:

1. Metal-to-metal contact

2. High-pressure contact 3. Gas-tight contact area 4. Large contact area 5. Mechanical stability

6. No localized stress concentrations (sharp bends).

Although mechanical, the wire wrap connection may be soldered if conditions dictate. To be considered a proper wire wrap connection, a termination is required to have a minimum of five complete wraps around the terminal and no more than seven wraps. There are numerous advantages to a wire wrap connection over soldering:

1. Less chance of personal injury 2. Savings in resources 3. Easy to disconnect as needed 4. Connections are more compact 5. Uniformity of connections 6. Stronger and more reliable

SUMMARY

As a technician, you may be required to maintain telephone circuits including making new connections and troubleshooting existing circuits. Knowing how the connections are made from end to end will help in isolating wiring faults. The lab for objective 2e will assess your ability to isolate and repair telephone wiring faults.

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INTRODUCTION

The RedCom Tactical Telephone Switch is a Theatre deployable multi access module used by communications units in a deployed environment. It was used to provide telephone and LAN connections over a Wide Area Network (WAN) connection. In this lab, you will configure two analog telephone circuits and enable trunking between the student stations.

OBJECTIVE

2d. Using supplied publications and digital equipment; configure a Telephone Switching circuit in accordance with Progress Check 3.2d.

Instructions In this lab, you will configure a telephone switch including customer lines and trunk configurations. The trunk lines and user access cables have been pre-run in order for each station to connect to their respective RedComm Telephone Switch. You will use a console program called Putty for configuring the system parameters. Follow the instructions in the lab workbook. The following steps will be performed during this lab:

1. Connect to the RedComm Telephone Switch a. Connect using Putty b. Follow the Configuration

Diagrams

2. Complete initial Configuration

3. Complete Trunk Configuration

4. Modify HMX/DNs

5. Test the Telephone Circuit 

Once complete, all student stations will have the ability to call over the trunks. Each student is responsible for programming their respective station and ensuring that each telephone has the ability to complete calls.

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INTRODUCTION

Each student will have the opportunity to troubleshoot a phone circuit. Using the telephone circuit simulators and using the provided tools and test equipment, each student will need to identify and correct various cable faults inserted by the instructors.

OBJECTIVE 

2e. Using supplied publications and digital equipment; troubleshoot connectivity problems in a Telephone Switching circuit in accordance with Progress Check 3.2e.

Read the following sections in your student text.

Mike Meyers’ Certification Passport

CompTIA Network+

The Troubleshooting Process, pages 356-362

 

TROUBLESHOOTING METHODS There are many different ways to troubleshoot electronics but we will focus on just a few in order to assist you with the lab. For now, we will only discuss some more popular methods of troubleshooting starting with the Half-Split Method.

Half-Split Method The Half-Split method is best applied when the system is well documented or if the technician has a good understanding of the system in a working state. For this method the technician will make an educated guess as to where the fault may lie and then progress through the system until the fault is isolated. This method is a logical leap frog through the system. To apply this troubleshooting method, start at the working end of a system like the telephone switch. Typically the telephone switch will represent a known good component. The next step involves “cutting” the circuit in half and testing for dial tone. Is dial tone present or not? If so, “cut” the remaining untested circuit in half and test for dial tone again. Is dial tone present or not? If not, you have isolated the area of the fault and you must then troubleshoot the components in between the last good signal and the loss of signal. This method will assist a technician in eliminating large portions of the circuit in a relatively short period of time.

Substitution Method The Substitution Method is troubleshooting by swapping a known good component with a suspected bad component in systems that with easily replaceable items. In order to apply this troubleshooting method, a technician must have a good understanding of the components in a system and make an educated guess

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as to the possible bad component. Additionally, known good components must be available. By substituting components until the problem is found, the substitution method may take more time if swapping the components doesn’t immediately fix the problem. Typically, this troubleshooting method is best applied to customer equipment instead of major components in the telephone switch.

“Circle the Wagons” Method This method involves the use of diagrams and drawings in order to logically test external components. To apply this method, get or create a drawing of the system for example, a wiring diagram. Circle the known good component on the diagram. Once circled, look at all of the components that cross the boundary of the circle and systematically test those lines or components. If the fault is not found, move to the next major component and repeat the process until the fault is isolated.

Simple to Complex Method The fourth and last method we will discuss is the Simple to Complex Method. This method is the easiest for a new technician as they don’t need an in depth knowledge of all of the components of a system. The Simple to Complex Method is often used in call centers and help desk environments. Many of the problems reported to technicians are often due to inexperienced users and can easily be isolated through questioning as mentioned in the Network+ reading assignment. To apply this method, start by isolating the easiest component to replace like a power cable or a cable connection. If the fault is not isolated by replacing the easiest components, move to the next, more complex component, and try again. Continue this process until the fault is isolated.

Summary

The mentioned troubleshooting methods are just a few of many ways to isolate and fix system faults. Many of the troubleshooting methods can be combined in order to more effectively fix problems. Often as a technician, you will use one of these methods and develop your own unique way of troubleshooting.

 

Instructions During this lab, you will troubleshoot connectivity problems in an existing telephone circuit. You are allowed tools and test equipment provided by your instructor. Once complete, test your circuit by verifying that dial tone has been restored.

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UNIT 3 VOICE OVER IP (VOIP)

OBJECTIVES a. Identify basic principles pertaining to digital voice/VoIP fundamentals. b. Using supplied publications and digital equipment, configure a Telephone VoIP circuit in

accordance with Progress Check 3.3b.

INTRODUCTION

In recent years, the communication industry has risen to a completely new level in the way we communicate. Voice over Internet Protocol (VoIP) emerged in the 1990s as a commercial product enabling users to place voice calls over the Internet, instead of using the Public Switched Telephone Network (PSTN).

In this system voice signals are broken into pieces of data – called data packets – and transported over the Internet to the receiver. Because the voice signals are disguised as data, the high costs of long-distance calls were avoided. As well as enabling voice trafficking, IP telephony provides a means of data and video convergence. Voice, data, and video can travel together across an IP network.

Since 2000, VoIP has become a mainstream alternative to the traditional telephone system. Most quality of service issues have been solved enabling VoIP traffic to be prioritized. VoIP usage is expanding quickly to include smaller companies and home users.

VOICE OVER INTERNET PROTOCOL (VOIP) Voice over Internet Protocol (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, Voice over Broadband (VoBB), broadband telephony, IP communications, and broadband phone service. The term Internet telephony specifically refers to the provisioning of communications services (voice, fax, Short Message Service (SMS), voice-messaging) over the public Internet, rather than via the Public Switched Telephone Network (PSTN). The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony, and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized and transmission occurs as Internet Protocol (IP) packets over a packet-switched network.

Packet Switching: Packet-switching is considered one of the most effective means of sharing network facilities among multiple users. Each packet contains all the necessary control and

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routing information to deliver the packets across the network. Packets can be routed independently or in series. Packets are maintained and preserved as an entity throughout the sending and receiving process. Simply put, the purpose of packet-switching is to gain more reliable data transfers and share links among many users. Packet switching relies on connectionless, non-dedicated transport technologies. Voice traffic is essentially delay-sensitive, and Quality of Service (QoS) is required to ensure voice quality and availability. Implementation of a QoS system will have to be used to prioritize traffic and allocate bandwidth to make sure everything sounds great.

Early providers of voice over IP services offered business models and technical solutions that mirrored the architecture of the legacy telephone network. Second generation providers, such as Skype, have built closed networks for private user bases, offering the benefit of free calls and convenience, while potentially charging for access to other communication networks, such as the PSTN. This has limited the freedom of users to mix-and-match third-party hardware and software.

Advantages of VoIP over the PSTN:  

• Lower costs for long-distance calls. VoIP providers compete with traditional PSTN providers by providing more affordable long-distance phone services. Long-distance VoIP calls are often very cheap or even free. • Per second charging. Generally, VoIP providers charge calls per second, enabling you to keep track of the exact cost of calls you make.

• No international service rates. VoIP can be used across borders, no matter where you are situated. The rates also remain the same, irrespective of what country or area you're in.

• Free features Extra PSTN features such as three-way calling, call waiting, callback, and caller ID are generally provided at an additional fee. For VoIP users, these features are now standard and are available for free.

VoIP continues to evolve and to gain popularity because of its many benefits. However, it also has some disadvantages.

Major Disadvantages of VoIP:

• Dependence on efficient processing. The quality of a VoIP call depends on the quality of the computer processor in use. A slow processor means that voice packets will be compressed slowly before they're sent.

• Dependence on Internet connection quality. The quality of a VoIP call depends on the Internet connection in use. A high-speed Internet connection is a minimum requirement for using VoIP.

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If your Internet connection is busy or sluggish, voice packets will be transmitted too slowly over the Internet or a VoIP call may not connect.

• Reliance on computers. Although VoIP calls can now be made from a regular telephone, they rely on users' computers to work efficiently. If your computer suffers damage and shuts down unexpectedly, your VoIP phone call will too. VoIP is affected by the same dangers that your computer faces, such as viruses and bugs.

• Reliance on additional Backup Power systems. VoIP systems rely on the available network backup power systems and do not include a built in or integrated backup power supply like PSTN. If the network backup power fails, all VoIP phones that connect through that device will no longer operate.

• Poor emergency call support. Poor support for tracing emergency calls. Emergency VoIP calls are difficult to trace a caller's location, making it difficult to deploy the most conveniently located emergency services.

VoIP Network Components The public switched telephone network (PSTN) consists of user endpoint devices, Private Branch Exchange (PBX) switches, PSTN switches, and a network of signaling devices. A PBX switch manages a network of private lines and connects it to a number of external PSTN lines. When a user attempts to make a telephone call over the PSTN, the local PBX connects the user endpoint device to a PSTN switch, which establishes the call and manages the resources required by it. The call is then switched across the necessary PSTN circuit.

Voice over Internet Protocol (VoIP) networks apply the same principles as the PSTN, but are structured quite differently. VoIP networks are based on client-server architecture and consist of four main components.

• Call-processing server: The call-processing server in a VoIP network performs the same function as a PBX in a circuit-switched network.

• User endpoint devices: The user endpoint devices or clients are the devices that users interact with, such as IP phones and softphones.

• VoIP gateway: The most notable difference between the PSTN and a VoIP network is the necessity for a VoIP gateway. VoIP networks use gateway devices to connect to the PSTN infrastructure and convert standard PSTN telephone numbers to IP addresses.

• IP network: A VoIP network requires an IP network for transporting call data from one user endpoint device to another.

Call Processing Server A call-processing server is an IP-enabled PBX (IP PBX) that directs VoIP calls (Figure 3-1). A PBX system switches analog signals across circuits, whereas an IP PBX system routes data packets across an IP network. An IP PBX system usually consists of software running on one or more dedicated servers. The call-processing server performs signal management services, such as conferencing, call-waiting, transferring VoIP calls, and finally terminating VoIP calls. In addition, the call-processing server determines the quality of service parameters required for a VoIP call.

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Figure 3-1. Cisco Unified Call Manager (CUCM) Server Suite

USER ENDPOINT DEVICES There are many different types of endpoint devices made by dozens of different manufacturers. Here, we will discuss the two major types of phones but will not discuss specific brands or models. The two types of IP phones are:

1. Physical IP phones. A physical IP phone is a telephone device that connects to an IP network via Ethernet. Modern IP phones provide a wide range of services, including hardware features such as Power over Ethernet (PoE) and secondary Ethernet ports, as well as software features for prioritizing voice traffic, for example. IP phones are usually used to provide VoIP functionality to users at a fixed location in an office.

2. Softphones. A softphone consists of a software application running on a personal computer connected to an IP network. The software application runs in the background to provide continuous VoIP access. Softphones are usually used to provide VoIP functionality to mobile users with laptops. Figure 3-2 shows an example of a physical phone (left) and a soft phone (right).

Figure 3-2. Physical phone (left) Soft phone (right)

Each VoIP user endpoint device requires an IP address within its local subnet. You can assign IP addresses manually or use the Dynamic Host Configuration Protocol (DHCP) to automatically

    

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assign IP addresses from a central location. The DHCP server leases IP addresses from a specified IP range to user endpoint devices when required. The amount of time that the IP address is assigned is determined by the type of Internet connection that the endpoint device uses. DHCP automatically assigns a new IP address when a user endpoint device connects to a new network port.

You can attach IP phones to LANs and WANs using a built-in multiport Ethernet hub, enabling them to connect seamlessly. You can also use Ethernet connections to attach IP phones to IP PBX devices to integrate IP and PBX technologies. IP phones access voice communication services from a communications application service provider (CASP) to transport real-time voice data over an IP network. This enables IP phones to support advanced conferencing features such as voice mail, name directories, display functions, and full-duplex audio. In addition, IP phones can add, drop, and transfer conference calls. Many vendors are currently developing conference IP phones for large enterprise implementations.

Since IP phones are assigned dynamic IP addresses by DHCP, you can easily add, move, or alter IP phone endpoints in a VoIP enterprise implementation. And finally, since all IP phones use the same network infrastructure and transport technologies, you can easily deploy IP phone configurations from a remote location.

Endpoint Device Protocols The features supported on IP Phones vary by the device protocol in which the phone is running. The protocols can be categorized into three groups:

1. H.323 2. Session Initiation Protocol (SIP) 3. Skinny Call Control Protocol (SCCP)

H.323 is an International Telecommunication Union-Telecommunication Standardization Sector (ITU-T) umbrella recommendation for controlling multimedia communications over IP based networks that do not provide a guaranteed quality of service. H.323 covers point-to-point communications and multipoint conferences and addresses call control, multimedia management, bandwidth management, and interfaces between LANs and other networks.

The basic components of the H.323 protocol are terminals, gateways, and gatekeepers which provide call control to H.323 endpoints. Similar to other protocols, H.323 applies to point-to point or multipoint sessions. However, compared to Media Gateway Control Protocol (MGCP), H.323 requires more configuration on the gateway since the gateway must maintain the dial plan and route patterns. The H. 323 Gatekeeper is an optional component in the H.323 network that provides a number of services to terminals, gateways, and devices. Those services include

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endpoint registration, address resolution, admission control, user authentication, and so forth. Of the various functions performed by the gatekeeper, address resolution is the most important as it enables two endpoints to contact each other without either endpoint having to know the IP address of the other.

Session Initiation Protocol (SIP) defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party or multiparty sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can function in one of the following roles:

1. User agent client (UAC) - A client application that initiates the SIP request.

2. User agent server (UAS) - A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user.

Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the User Agent that initiated the request. From an architecture standpoint, the physical components of a SIP network can be grouped into two categories: clients and servers.

• SIP clients include Softphones, Physical Phones or Gateways. Phones can act as either a UAS or UAC. Softphones and Cisco SIP IP phones can initiate SIP requests and respond to requests. Gateways provide call control. Gateways provide many services, the most common being a translation function between SIP conferencing endpoints and other terminal types. This function includes translation between transmission formats and between communications procedures. In addition, the gateway translates between audio and video codecs and performs call setup and clearing on both the LAN side and the switched-circuit network side.

• SIP servers include Proxy, Redirect, and Registrar. The proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client’s behalf. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security. The Redirect server provides the client with information about the next hop or hops that a message should take and then the client contacts the next hop server or UAS directly. The Registrar server processes requests from UACs for

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registration of their current location. Registrar servers are often co-located with a redirect or proxy server.

The Skinny Call Control Protocol (SCCP or Skinny) is a Cisco proprietary network terminal control protocol originally developed by Selsius Systems. The SCCP technology is now owned and defined by Cisco Systems. SCCP is a lightweight protocol for session signaling with Cisco Unified Communications Manager (CUCM) and is used to communicate between IP devices and the CUCM server. CUCM acts as a signaling control for call events initiated over other common protocols such as H.323, Session Initiation Protocol (SIP), ISDN and/or Media Gateway Control Protocol (MGCP).

IP NETWORK COMPONENTS In order for a VoIP phone to work, many different components must be configured and installed. For a basic VoIP implementation three main components are required.

VoIP/Voice Gateway Like all media gateways, VoIP gateways (Figure 3-3) convert circuit-switched communications into packet-switched communications. In addition, VoIP gateways implement VoIP-specific features such as voice compression, echo cancellation, and silence suppression. A VoIP gateway can also gather network performance data and establish and terminate VoIP calls. There are a number of ways of implementing a VoIP gateway, for example software package running on a personal computer or a dedicated hardware device. A gateway consists of:

• Standard line interfaces. The standard line interface (also known as the trunk interface) connects a gateway with PSTN lines.

• IP interfaces. A gateway connects to the VoIP network via its IP interfaces.

Figure 3-3. Cisco VG224 Voice Gateway Router.

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NETWORK ROUTERS A network router forwards data packets between computer networks. A router is connected to two or more data lines from different networks. When a data packet comes in one of the lines, the router reads the address information in the packet to determine its ultimate destination. A data packet is typically forwarded from one router to another through the network until it reaches its destination node. Network Routers provide two key services.

1. Real-Time Transport Protocol (RTP) 2. Quality of Service (QoS)

REAL-TIME TRANSPORT PROTOCOL (RTP) RTP defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to talk features. RTP is one of the technical foundations of Voice over IP and often used in conjunction with a signaling protocol which assists in setting up connections across the network. RTP is designed for end-to-end, real-time, transfer of streaming data. The protocol provides facilities for jitter compensation and detection of out of sequence packets, which are common during transmissions on an IP network. RTP allows data transfer to multiple destinations through IP multicast. RTP is regarded as the primary standard for audio/video transport in IP networks.

QUALITY OF SERVICE (QOS) In the field of telephony, quality of service was defined by the ITU in 1994. Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. Quality of service guarantees are important if the network capacity is insufficient, especially for real-time streaming multimedia applications such as voice over IP, online games and IP-TV, since these often require fixed bit rate and are delay sensitive, and in networks where the capacity is a limited resource, for example in cellular data communication. Figure 3-4 shows basic functionality of QoS.

Figure 3-4. Quality of service example

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NETWORK SWITCHES Network switches can provide two primary features to assist the IP telephony deployment: (Figure 3-5)

1. Power over Ethernet: In-line power capabilities allow a Cisco Catalyst switch to send power over Ethernet to a Cisco IP Phone or other in-line power-compatible devices (wireless access points) without the need for an external power supply. In-line power is commonly referred to as Power over Ethernet (PoE).

2. Voice VLAN support: One or more network devices can be connected to the back of the Cisco IP Phone. Voice VLANs place IP phone traffic into a VLAN separate from the desktop computers connected to the PC port of the phone. Voice VLAN capabilities allow Quality of Service (QOS) to be applied to the separated data when the traffic is sent through a network router.

Figure 3-5. Switches

VOIP CALL ROUTING Basic VoIP call routing consists of a series of connections made by the Call Processing server. All calls are forwarded to the call processing server first in order to determine the path between devices. The call processing server then checks its database in order to determine if the call is on the local network or off the local network. If a dialed number matches an entry in the call processing servers’ database, the call is forwarded to the appropriate phone on the local network. If the dialed number does not match an entry in the call processing servers database, the server will make the route decision by checking the designated call routing rules set up by the system administrator. The administrator sets the trunk or path an outside call will take if the call is determined to be off the local network. Off network calls typically trunk to the PSTN via a T-1 or PRI link through voice gateway router.

VOICE OVER SECURE IP (VOSIP) Voice over Secure IP (VoSIP) is the same as SVoIP in that they are technologies used to securely transmit voice communications, but with VoSIP, the security is provided by separate devices in the network (such as network encryptors) rather than the secure phones themselves. Note that the information is only secured once it passes through the encryption device — it is unsecure when

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being transmitted between each phone and its network encryptor. Figure 3-6 shows how a simple VoIP system can be made into a VoSIP system with the use of network encryptors.

Figure 3-6

SECURE VOICE OVER IP (SVOIP) Secure Voice over IP (SVoIP) is when secure phones are used to protect information sent over the VoIP network. Devices that are used in a SVoIP phones include Sectéra vIPer phone systems, an IP based replacement for the standard Secure Telephone Equipment (STE). Figure 3- 7 shows a basic SVoIP system with the use of vIPer telephone systems.

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Figure 3-7

SUMMARY

Voice over IP is the leading voice technology and is gradually replacing analog telephony systems. As a network technician, it is very important that you understand the basics of VoIP and its role in a modern network.

REVIEW  EXERCISE 3‐1 

1.  List the advantages of VoIP. 

 

2.  List the disadvantages of VoIP. 

 

3.  What four components are part of a VoIP system? 

 

4.  What are the two type’s phones? 

 

5.  Which protocols are used for endpoint devices? 

 

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6.  Which endpoint device protocol is the most widely used? 

  

7.  SIP user agents can function in which two roles? 

 

8.  Network routers provide what two services to a VoIP system? 

 

9.   What component is required for VoSIP?