telephony systems

8/12/2019 Telephony Systems

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Telephony SystemsEE-523

By: Dr. Raed Al-Zubi


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Introduction Information: audio, image, data, and video signal Regulations:

In USA Federal Communications Commission (FCC) for long

distance traffic Public utilities Commissions of individual states forlocal services

In UK

British Telecom and Mercury communicationsprovide local and trunk services In Jordan

Telecommunications Regulatory Commission (TRC)


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Introduction Standards:1. International telecommunications union (ITU)

The ITU Telecommunication sector (ITU-T)Formerly the comite consultatif telegraphique ettelephonique (CCITT)Study technical questions , operating methods andspecifications for telephony , telegraphy , and datacommunications

The ITU Radiocommunications sector (ITU-R)Formerly the comite consultatif International desRadiocommunications ( CCIR)It studies all technical and operating questions relatedto radio communications, including point - to pointcommunications, mobile services and broadcasting


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Introduction

2. International Frequency Registration Board(IFRB)Associated with ITU-R.Assignment of radio frequencies to preventinterferences

3. International Standards Organization (ISO)For many fields including Telecommunications

4. Institute of Electrical and Electronic Engineers(IEEE)


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Introduction Networks Classification

1. Body Network2. Personal Area Network (PAN): in office

3. Local Area Network (LAN): in Building (100 m),campus (1 Km)4. Metropolitan Area Network (MAN): in city (10 Km)5. Wide Area Network (WAN): in country (100 Km),

continent (1000 Km)6. The Internet: Planet


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Introduction Example of MAN: cable television network


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Notes Major backbone operators are companies like AT&T and

Sprint. They Operate large international backbonenetworks with thousands of routers connected by high-bandwidth fiber optics

Routers in Regional ISP are located in different cities

Regional Cable Network (RCN) project between Jordan,Syria, Turkey, Saudi Arabia, and UAE to establish anoptics fiber network for internet. Started in late 2010 andexpected to be launched during the third quarter of 2012.(Jordan Times in Apr 06, 2012)


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Notes Manufacturers of IP-based telephone

systems: Aastra, Alcatel-Lucent, AvayaCommunications, Cisco Systems, Inc.,Microsoft, Mitel Networks, NEC, Polycom,ShoreTel, Inc., Siemens ICN, Vertical

Amazon, Apple, Google, Microsoft, and Netflixhave all developed innovative products andsoftware that have had enormous influence onthe Internet, consumer services, and businessoperations


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Public Switched Telecommunication (Telephone) network- PSTN


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PSTN CT: centre de transit Connecting all the exchanges as a mesh

network is called junction network Connecting all the exchanges as a star

network is done via a central switchingcenter called a tandem exchange

Connecting Tandem exchanges is called

trunk network (it is used betweencountries)


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PSTN In practice, a mixture of mesh and star

network is used


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Introduction Terminology:

Trunk networkToll networkTrunk exchangeToll office

TrunkJunctorJunctionInter-office trunk

Local exchangeEnd officeExchangeCentral office

Local networkCustomer loopUKUSA


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Introduction OSI Reference Model

The Open Systems Interconnection (OSI)model was proposed by ISO to achievecompatibility between different communication

devices produced by different manufacturers The function of each device is divided into

different layers

Each layer has type of protocols that shouldperform a well-defined function


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OSI Model


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Notes MAC address: Media Access Control address.

Linked to the hardware of network adapters. Forthis reason, the MAC address is sometimescalled the hardware address, the burned-in

address (BIA), or the physical address. Assignedby the manufacturer of a network interface card(NIC) and are stored in its hardware, the card's

read-only memory IP address: is associated with software


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TCP/IP model OSI model is more general but rarely used

any more. TCP/IP model has the oppositeproperties


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TCP/IP model Internet layer: permits hosts to inject packets into any

network and have them travel independently to thedestination (potentially on different network)

It contains more complicated protocols than OSI model

OSI model was used to connect hundreds of universitiesand government installations using leased telephoneline. However, ARPANET (research network sponsoredby DoD) required a new model to connect to satellite andradio networks


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KEY UNDERLYING TECHNOLOGIES Fiber-optic cabling , multi-core processors ,

and low-cost memory are the building blocks ofmodern networks. They enable networks to carrymore information, faster

Fiber-optic: The introduction of fiber cabling by MCI (long-

distance company for fiber) was in 1983 for intercityrouting

The most significant advantage of fiber-optic cablingis its enormous capacity. Also, Non-electric signals(light) can travel 80 miles before having to beregenerated


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KEY UNDERLYING TECHNOLOGIESFiber-optic:

But data over copper cabling are subject to fading over relatively short distances(1.5 mile). Consequently, amplifiers are needed every mile and a half to boostthe electrical signals carried on copper-based networks

Also, copper cabling is heavier, and has less capacity than fiber cabling. Also,

Signals transmitted via copper react to electrical interference or noise on theline. Interference from nearby wires is called crosstalk and can be reduced bytwisting each insulated copper wire of a two-wire pair

Fiber optics and its associated electronics have evolved to the point where aconsortium of companies including Google, Japanese carrier KDDI, SingaporeTelecommunications, and Indias Reliance Globalcom are constructing and willoperate a six-pair fiber undersea cable with a capacity of 17 terabits per second(Tbps). (One terabit equals 1,000Gb.) Thats fast enough to transmit every bookin the British Library 20 times per second


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KEY UNDERLYING TECHNOLOGIESFiber-optic:

Wavelength-division multiplexing further expandedfibers capacity. These multiplexers essentially split asingle fiber into numerous channels, each able to

transmit a high-speed stream of light pulses, asshown in Figure 1-1. The current generation ofmultiplexers are capable of transmitting up to 88channels of information, each operating at 100 (Gbps)

The undersea cable will run from Singapore to Japan, withextensions to Hong Kong, Indonesia, the Philippines, Thailand,and Guam. At the time of this writing, it was scheduled to beginoperation sometime in 2012


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Note:- The development of the erbium-doped fiber amplifier (EDFA) made DWDMpossible.- C band EDFAs operate from 1530 nm to 1560 nm.- The bandwidth of a C Band system is 4 trillion Hz


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KEY UNDERLYING TECHNOLOGIES Faster, Lower-Priced Processors

They enable networks to process multiplestreams of light signals simultaneously

95 percent of mobile devices sold worlwide areequipped with ARM chips. This architecture nowincorporates 32-bit processing (the ability toprocess data in chunks of 32 bits), which means

that they process data faster. Moreover, they aresmall and inexpensive, and they use only smallamounts of power.


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Multiplexing Channel Access Methods

FDMA: Frequency division multiple access TDMA: Time division multiple access CDMA: Code division multiple access SDMA: (Space-Division Multiple Access) using directional antenna,

power control. PDMA: (Polarization division multiple access) Separate antennas are

used in this type, each with different polarization and followed byseparate receivers, allowing simultaneous regional access of satellites.Each participating earth station with an antenna that has dualpolarization

PAMA : Pulse-address multiple access. It enables the ability of acommunication satellite to receive signals from several Earth terminalssimultaneously and to amplify, translate, and relay the signals back toEarth, based on the addressing of each station by an assignment of aunique combination of time and frequency slots.

Random Access : CSMA, ALOHA (pure ALOHA and slotted ALOHA)


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Multiplexing Multiplexing combines traffic from multiple devices

or people into one stream so that they can share achannel or path through a network

Types of Multiplexing:

Space-division multiplexing Frequency-division multiplexing Time-division multiplexing Code-division multiplexing


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FDM


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4 KHz Human Voice Channel

Note: In the United States, AT&T designed its FDM systems tohandle the band of signals between 200 and 3400 Hz (Bw = 3200Hz).


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CCITT FDM Groups

12 channels per group5 groups per supergroup5 super groups per mastergroup3 master groups per supermastergroup


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900 Hz guardband between channelsSingle Sideband suppressed carrier modulation SSB


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CDMA Typically, there are 64 or 128 chips per bit

Each station is assigned a unique m-bit codecalled a chip sequence if station A is assigned the chip sequence

00011011, it sends a 1 bit by sending 00011011and a 0 bit by sending 11100100 It is known how to generate such orthogonal chip sequences using a method

known as Walsh codes. Inner product of two chip codes is zero


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CDMAExample


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TDM


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TDM In RX: sampling pulses should be synchronized with that

in TX In Telephone system: sampling rate is 8KHz (2 x 4 KHz,

Nyquist frequency). So 8k pulses/second Possible modulation methods: PAM-pulse amplitude

modulation, PWM-pulse width modulation , PPM-pulseposition modulation Problems: attenuation and delay causes dispersion of

the transmitted pulses. So pulses interfere with other

pulses of adjacent channels. Hence, inter-channelcrosstalk Solution: Pulse Code Modulation


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PCM


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PCM A/D: an analog level of voltage is

converted to a group of bits (word=32 bitsor byte= 8 bits)

In telephone system: 8 bit encoding isused (256 levels).

8 K pulse /sec ( 1 pulse = 8 bits)

Bit rate = 8 k x 8 /sec = 64 kbps (= baudrate, since bit is one symbol).


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PCM Nyquist showed that the minimum bandwidth

needed to transmit a digital signal at B bauds isB/2. So, in telephone system: minimum BW= 32 KHz.

( but for analog BW =4 KHz) PCM introduces quantization distortion which is

not found in analog transmission. If quantization is done using uniform size steps,

then high quantization error. So, non-uniformsize steps are used.


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PCM We can use uniform quantization but we need

companding (compression and expanding) Compression: reduces the dynamic range of the

analog signal such that the quantization process

results in a good SNR. Two companding laws were standardized by

CCITT: A-law in European system, mu-law used

in America and Japan system


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TDM and VoIP Once a connection is established, capacity

is saved even when the device is notsending information. But there are smallslices of silence in voice (wasting networkcapacity). This is the reason TDM is beinggradually replaced in high-traffic portionsof networks by Voice over InternetProtocol (VoIP) technologies


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Statistical Multiplexers Unlike TDMs, statistical multiplexers do not guarantee

capacity for each device connected to them. Rather, theytransmit voice, data, and images on a first-come, first-served basis, as long as there is capacity

Statistical multiplexers support more devices than TDMsbecause they dont need to save capacity when a deviceis not active. It can be used in a Wide Area Network(WAN) to connect customers to the Internet. Customerswho contract for more costly, high-priority service canobtain higher speeds than customers with lower-priorityservice during traffic spikes


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PCM Primary Multiplex Group European 30-channel system (E1-

European system 1) 24- channel system (T1- transmissionsystem 1) used in North America and

Japan.

Note: T0 and E0- are used for one channel(64 Kbps)

What about E2, E3, or T2, T3


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E1 System Uses A-law companding

Bit rate= 32*8/125=2.048 Mbps


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T1 System

Uses mu-law companding Frame =1/fs = 1/ 8K = 125 micro Seconds Number of bits= 1 (synch) + 8*24 = 194 Bit rate = 194/125=1.544 Mbps Digit 8 of each channel in each 6 th frame is used

for signaling for that channel


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Digital Hierarchy in E1 and T1 The plesiochronous digital hierarchy-PDH (old)

The synchronous digital hierarchy-SDH (new) The synchronous optical network SONET(new)

PDH: the timing and clocking information arecontained within the digital bit stream and thusthis system is self-synchronized orasynchronous.

SDH/SONET: the timing and clocking informationare obtained from a highly accurate master clock


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PDH European PDH (Ex) North American PDH (Tx or DSx)


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PDH There are some bits are added for frame

alignment and justification In North America and Japan, T3 carries

672 conversations over one line at aspeed of 45Mbps. T3 is used for largeenterprises, call centers, and Internetaccess. Small and midsize organizationscommonly use T1 for Internet access


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PDH Multiplex mountain


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SDH and SONET They are standardized protocols that

transfer multiple digital bit streams overoptical fiber Developed to replace the PDH system for

transporting large amounts of telephonecalls and data traffic over the same fiberwithout synchronization problems

(synchronization sources of variouscircuits were different)


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SDH and SONET SONET in the United States and Canada,

and SDH in the rest of the world. Althoughthe SONET standards were developedbefore SDH, it is considered a variation ofSDH because of SDH's greater worldwidemarket penetration


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SDH and SONETSTM-1: Synchronous Transport Module, level 1

STS-1: Synchronous Transport Signal, level 1


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Advantages of Digital Transmission The same digital equipments can be used to

process all types of digital sources Digital signals are highly resistant to crosstalk.

The crosstalk is most annoying when the twoparties of a call are not talking and can hear andunderstand another call. In digital, it is random.

Signaling is made simpler and cheaper Low cost


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Advantages of Digital Transmission Easy to multiplex and the ability to mix

voice, video, photographs, and e-mail onthe same transmission enables networksto transmit more data

Better performance in the presence ofnoise Higher speeds: It is faster to re-create

binary digital ones and zeros than morecomplex analog wavelengths

Disadvantages of Digital


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Disadvantages of DigitalTransmission

Higher bandwidth

The information capacity of digital systemis limited Shannon limit for information capacity C = 3.32 B log (1+ S/N) , C: bps, B:

bandwidth Hz

A/D and D/A are required Synchronization


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Selected Topics As Assignment Simulation of TCP (using MATLAB or other programs) VoIP Simulation of statistical multiplexer (using MATLAB,

Arena, or other programs) Cognitive radio

Routing protocols UWB Simulation of wireless network AirCom WRAN


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Selected Topic in Routing Dijkstra Algorithm (Step 1)


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Dijkstra Algorithm (Step 2)


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http://www.personal.kent.edu/~rmuhamma/Algorithms/MyAlgorithms/GraphAlgor/dijkstraAlgor.htm


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Network Performance Metrices Packet loss : This refers to the packets that are dropped

when there is network congestion. Voice conversations

break up when packet loss is too high Latency: This term refers to delays (in milliseconds) thatare incurred when voice packets traverse the network.Latency results in long pauses within conversations, andclipped words

Throughput: measures actual user data transmittedover a fixed period of time

Bit error rate Fairness: to determine whether users or applications

are receiving a fair share of system resources Jain's fairness index (1/n worst , 1 best)


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Network Performance Metrices Jitter : results in noisy calls that contain pops

and clicks or crackling sounds The instant at which pulses are retransmitted by a repeaterare determined by a local oscillator synchronized to the digitrate, which must be extracted from the received waveform.Variations in the extracted frequency can cause a periodicvariations of the times of the regenerated pulses, which iscalled jitter

Wander : the variation in the times of the

regenerated pulses due to changes inpropagation time


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Network Performance Metrics Power levels : usually SNR in dB is used


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Network Performance Metrics Power levels

dB PP

log10i

o=G

dBm mW1P log10 o=G

dBW W1P

log10o=

G

used to indicate power levels relative to 1 mWExample: 1 W = 30 dBm

used to indicate power levels relative to 1 W


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Network Performance Metrics


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Network Performance Metrics Relative level of a signal at any point in the

system with respect to its level at thereference point is denoted by dBr Signal level in terms of the corresponding

level at the reference point is denoted bydBmO: dBmO= dBm-dBr


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Network Performance Metrics Echo: is the annoying effect of hearing

your voice repeated. This is usuallycorrected during installation by specialecho-canceling software

In analog system: Echo is produced due toamplification in two directions.


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Network Performance Metrics


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2-wire / 4-wire circuit Using Amplifier in one direction cannot

pass signal in the second direction Using two amplifiers causes continuousoscillation called singing

So, we use hybrid transformer, but theprice is 3 dB loss in each direction Imperfect line balance causes part of the

signal transmitted in one direction to returnin the other. This is called echo.


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2-wire / 4-wire circuit


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2-wire / 4-wire circuit Total attenuation from one two-wire circuit

to the other is

Transhybrid loss (TL): is the attenuation

through the hybrid transformer from oneside of the 4-wire circuit to the other.TL = 6+B dB

42 6L G=

dB Z-NZN log20B

+=


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2-wire / 4-wire circuit B: balance return loss due to impedance

mismatch between 2-wire line and thebalance network

Z: impedance of the 2 wire line

N: impedance of the balance network The attenuation of the echo that reaches

the talkers 2-wire line

dB B2L3G-6)(BG-3L 244t +=+++=


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2-wire / 4-wire circuit The attenuation of the echo that reaches the

listeners 2-wire line

We can control echo by applying loss when 2T 4< 40 msec (increasing L2 by increasing thelength but this increases delay)

For 2T 4 > 40 msec, we use echo suppressor(electronic device)

dB B22LG-6)(BG-6)(BL 244l +=+++=


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Echo suppressor


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2-wire / 4-wire circuit Stability:


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2-wire / 4-wire circuit Singing point of a circuit is the maximum gain S

that can be obtained from 2-wire to 2-wire linewithout producing singing

Stability margin is the maximum amount ofadditional gain M that can be introduced (equallyand simultaneously) in each direction of thetransmission without causing singing

dB LBM

0M2)L2(B02ML

2

2

S

+=

=+

=

Transmission Performance In


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Telephone System A rating system was standardized by CCITT to

grade a customer satisfaction is called OverallLoudness Rating (OLR) or Overall ReferenceEquivalent (ORE) in dB

ORE= TRE + RRE + losses dB

TRE: transmit reference equivalent

RRE: receive reference equivalent -ve dB: the system is better than the reference

Transmission Performance In


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Telephone System TRE and RRE is measured using a reference

system called NOSFER in ITU lab in Geneva

How the Telephone Works !!!


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p

The first Telephone


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p

In 1876, Alexander Graham Bell made the

first Telephone, called the Bell Telephone The same equipment is used at TX andRX

Switching


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g

Types of switching:

Circuit switching (space-division switching oranalog switching)

Message switching (delay or queuing system)

Packet switching (time-division switching ordigital switching)

Circuit Switching


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g

A physical path is established in advance

between the sender and the receiver andthis path is reserved for only one call (so,for voice network)

Circuit Switching


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g

It is an example of lost-call system ( the

call cannot be stored as in messageswitching) In old systems: manual switching was

done by operator Then, automatic systems were used:

The step-by-step switching system The crossbar switching system

Step-by-step Switching System


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Uses the two-motion selector which was invented byAlmon B. Strowger

Had a lifetime of nearly 100 years It is the first automated switching system


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Crossbar Switching System


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Packet Switching


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Dividing the long messages into packets is useful to solve theproblem of queuing messages with different lengths

Used in internet

Developed in 1969 by ARPANET (Advanced Research ProjectsAgency ). So, ARPANET was the pioneer to todays Internet

The Department of Defense wanted a more reliable network withroute diversity capability. In a national emergency such as theSeptember 11, 2001 attacks in the United States on the Pentagon inWashington, DC, and the World Trade Center in New York City, theInternet still functioned when many portions of the public voice andcellular networks were either out of service or so overwhelmed with

traffic that people could not make calls

Packet Switching


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Timing of events in circuit switching, message switching,packet switching


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Deep Packet Inspection


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DPI software develops a database of patterns, alsoreferred to as signatures . Each signature or pattern is

associated with a particular application such as peer-to-peer music sharing or protocols such as VoIP or trafficfrom certain hackers

In 2009, several music companies sued a 25-year-old

graduate student for illegal music downloads and won a$670,000 judgment. In another case, music companieswon a $1.2 million judgment case against a singlemother

DPI equipment can be used to monitor e-mail messagesby detecting keywords. For example, during 2009presidential elections in Iran

Types of Services


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Connection-oriented service is modeled afterthe telephone system. The user first establishesa connection, uses the connection, and thenreleases the connection

Connectionless service is modeled after thepostal system. Each message (letter) carries thefull destination address, and each one is routedthrough the system independent of all the others

Other Types of Services


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Channel Associated Signaling


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Common Channel Signaling

Signaling


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Advantages of CCS:

Information can be exchanged more rapidly Signals relating to a call can be sent while the call is

in progress. This enables customers to alter theconnections after they have been set up. Forexample, a customer can transfer a call elsewhere, orrequest a third party to be connected into an existingconnection.

Signals can be exchanged between processors forfunctions other than call processing, for example formaintenance or network-management purposes.

Signaling


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In CAS, the successful exchange of signals over a circuitproves that the circuit is working. CCS does not

inherently provide this checking facility, so a separatemeans (e.g., automatic routine testing) must be providedto ensure the integrity of the speech circuits.

In CCS, each message must contain a label called the

circuit identity code that indicates to which speech circuitand thus to which call it belongs. So, no connection isrequired to an incoming junction before an addresssignal is received. The address signal can therefore bethe first message sent.

A signaling link operating at 64 Kbps normally providessignaling for up to 1000 to 1500 speech circuits.

Signaling Networks


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Associated Signaling

Signaling Networks


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Non-associated Signaling:

Signaling Networks


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Quasi-associated signaling: used when thereare few circuits between A and B and thus little signalingtraffic between them. It is normally provided in case theassociated-signaling link fails

Customer Line Signaling


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Pulse dialing: rapidly disconnecting andre-connecting the calling partys telephoneline. Similar to flicking a light switch on andoff . Used to determine the dialed number

Dual-tone multi-frequency signaling(DTMF): used by push-button telephone.Sends each digit as a combination of twofrequencies, why?

DTMF


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It generates a sinusoidal tone which is mixture of the row and column frequencies

FDM Signaling


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Out-band signaling is done on a channel that is dedicated for the purpose

and separate from the channels used for thetelephone call. Out-of-band signaling is used inSignaling System 7 (SS7), the latest standard for thesignaling that controls the world's phone calls.

Example: send signaling information in the guardband in the telephone channel

Disadvantage: all the routes in the network must useout-band signaling (practical problem)

Out-band Signaling


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FDM Signaling


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In-band Signaling: called voice frequency signaling (VF) Signaling is done within the band of the data Advantage: independent on the transmission

system and can work over any circuit Example: DTMF tones

Voice Frequency Signaling System


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VF Receiver


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PCM Signaling


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30-channel system:

PCM Signaling


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24-channel system:

In every sixth frame, the eighth bit for eachchannel is used for signaling instead ofspeech. This has been found to cause a

negligible increase in quantizationdistortion

Signalling System No. 7 (SS7) Before SS7:


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Before SS7: CCITT R1 (region1): national level, CAS system, Multi-frequency, USA CCITT R2 (region2): national level, CAS system, Multi-frequency, Europe SS5: both national and international, CAS system, In-band signaling SS6: both national and international, out-band signaling and it is the first CCS

system. Had a restricted 28-bit signal unit that was both limited in function andnot suitable for digital systems

SS7: Uses out-band signaling and it is CCS system Called CCITT signaling system No. 7 Called Common Channel Signaling System 7 Is a set of telephony signaling protocols which are used to set up most of the

world's PSTN telephone calls The main purpose is to start and end telephone calls but then it is used for other

service such as prepaid billing mechanism and SMS

Signalling System No. 7 (SS7)


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Signalling System No. 7 (SS7)Sign l mess ges re p ssed from the centr l processor


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Signal messages are passed from the central processorof the sending exchange to the CCS system.

The Signaling-control subsystem structures themessages in the appropriate format and queues them fortransmission. When there are no messages to send, itgenerates filler messages to keep the link active.

Messages then pass to the signaling terminationsubsystem, where complete signal unit (SU) areassembled using sequence numbers and check bitsgenerated by error-control subsystem.

At the receiving terminal, the reverse sequence is carriedout.

SS7 and OSI Model

SS7 b d l d k f


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SS7 can be modeled as a stack ofprotocols like OSI model

SS7 consists of: Level 1: the physical level

Level 2: The data-link level Level 3: The signaling-network level

Level 4: The user part

SS7 and OSI Model Level 1: performs the functions of sending bit


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Level 1: performs the functions of sending bitstreams over a physical path

Level 2: performs the functions of error control,error rate monitoring, flow control anddelineation of messages

Level 3: provides the functions required for asignaling network. Each node in the network hasa signal-point code, which is a 14-bit address.Every message contains the point code of thetransmitter and receiver.

Level 4: provides the functions required toprovide different services to the user.


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119MTP: Message Transfer Part

ISDN User Part (ISUP) : The ISDN User Part (ISUP) defines theprotocol used to set-up, manage, and release trunk circuits that carryvoice and data between terminating line exchanges (e.g., between acalling party and a called party).


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Telephone User Part (TUP): In some parts of the world (e.g., China,Brazil), the Telephone User Part (TUP) is used to support basic callsetup and tear-down. TUP handles analog circuits only. In manycountries, ISUP has replaced TUP for call management.

Signaling Connection Control Part (SCCP): SCCP providesconnectionless and connection-oriented network services. SCCP is usedas the transport layer for TCAP-based services .

Transaction Capabilities Applications Part (TCAP): TCAP supportsthe exchange of non-circuit related data between applications across theSS7 network using the SCCP connectionless service. For example,TCAP carries Mobile Application Part (MAP) messages sent betweenmobile switches and databases to support user authentication,equipment identification, and roaming.

Speech Codinghttp://www-mobile.ecs.soton.ac.uk/speech_codecs/

Narrowband speech codecs: used to give an


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Narrowband speech codecs: used to give anefficient digital representation of telephone

bandwidth speech An ideal speech codec will represent this speech

with as few bits as possible, while producingreconstructed speech which sounds identical, oralmost identical, to the uncoded speech

In practice there is always a trade-off betweenthe bit rate of the codec and the quality of itsreconstructed speech

The Basic Properties of Speech

Speech is produced when air is forced from the lungs


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Speech is produced when air is forced from the lungsthrough the vocal cords and along the vocal tract.

An important part of many speech codecs is themodelling of the vocal tract as a short term filter The spectral peaks of the sound spectrum |P(f)| are

called formants

formants are controlled by the shape of the tract andthey are the poles of the short term filter As the shape of the vocal tract varies relatively slowly,

the transfer function of its modelling filter needs to be

updated only relatively infrequently (typically every 20ms or so)


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Speech sounds can be broken into three


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Speech sounds can be broken into threeclasses depending on their mode ofexcitation :1. Voiced sounds are produced when the vocal

cords vibrate open and closed, thus

interrupting the flow of air from the lungs tothe vocal tract and producing quasi-periodicpulses of air as the excitation. The rate of theopening and closing gives the pitch of thesound.

-Voiced sounds show a high degree of periodicity at the pitch period,which is typically between 2 and 20 ms.- These figures show a segment of voiced speech sampled at 8 kHz.Here the pitch period is about 8 ms or 64 samples.


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2. Unvoiced sounds result when the excitation is a


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noise-like turbulence produced by forcing air at high

velocities through a constriction in the vocal tractwhile the glottis is held open. Such sounds showlittle long-term periodicity as can be seen from nextfigures, although short-term correlations due to the

vocal tract are still present3. Plosive sounds result when a complete closure is

made in the vocal tract, and air pressure is built upbehind this closure and released suddenly


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The shape of the vocal tract and its mode


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The shape of the vocal tract and its modeof excitation change relatively slowly, andso speech can be considered to be quasi-stationary over short periods of time (ofthe order of 20 ms)

Speech coders attempt to exploit thispredictability in order to reduce the datarate necessary for good quality voicetransmission.

Commonly Used Speech Codecs Divided into three classes: waveform

d d d h b id d


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codecs, source codecs, and hybrid codecs

Waveform Codecs

They are signal independent and work well with


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non-speech signals

They are low complexity codecs which producehigh quality speech at rates above about 16kbits/s

Time-domain waveform codecs PCM: 64 Kbps, this is the simplest waveform codecs. They have the advantages of low complexity and delay with

high quality reproduced speech, but require a relatively highbit rate.

Waveform Codecs DPCM:

It tili th l ti t i h l d t


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It utilizes the correlations present in speech samples due tothe effects of the vocal tract and the vibrations of the vocal

cords It quantizes the difference between the original and predictedsignals.

48 kbps

ADPCM: Adapts the quantization step to the difference (small step for

small difference and large step for big difference) 32 kbps (very similar to the 64 kbits/s PCM codecs) Later ADPCM codecs operating at 16,24 and 40 kbits/s were

also standardized. Used in VoIP

Waveform Codecs Frequency-domain waveform codecs:

Sub Band Coding (SBC):


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Sub-Band Coding (SBC): The input speech is split into a number of frequency bands, or sub-

bands, and each is coded independently using for example anADPCM like coder. At the receiver the sub-band signals aredecoded and recombined to give the reconstructed speech signal.

The advantages of doing this come from the fact that the noise ineach sub-band is dependent only on the coding used in that sub-band. Therefore we can allocate more bits to perceptually important

sub-bands so that the noise in these frequency regions is low, whilein other sub-bands we may be content to allow a high coding noisebecause noise at these frequencies is less perceptually important.

16-32 kbits/s Due to the filtering necessary to split the speech into sub-bands

they are more complex than simple DPCM coders, and introducemore coding delay. However the complexity and delay are stillrelatively low when compared to most hybrid codecs.

Waveform Codecs

Frequency-domain waveform codecs:


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q y Adaptive Transform Coding (ATC):

Uses a fast transformation (such as the discretecosine transformation) to split blocks of the speechsignal into a large numbers of frequency bands.

The number of bits used to code eachtransformation coefficient is adapted depending onthe spectral properties of the speech.

Toll quality reproduced speech can be achieved atbit rates as low as 16 kbits/s.


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Source Codecs

Vocoders tend to operate at around 2.4 kbits/sb l d d h hi h l h h


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or below, and produce speech which although

intelligible is far from natural sounding.Increasing the bit rate much beyond 2.4 kbits/sis not worthwhile because of the inbuilt limitationin the coder's performance due to the simplifiedmodel of speech production used. The main useof vocoders has been in military applicationswhere natural sounding speech is not as

important as a very low bit rate to allow heavyprotection and encryption

Hybrid Codecs

The most successful and commonly used


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are time domain Analysis-by-Synthesis(AbS) codecs

AbS codecs work by splitting the inputspeech to be coded into frames, typicallyabout 20 ms long. For each frameparameters are determined for a synthesisfilter, and then the excitation to this filter isdetermined


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Hybrid Codecs AbS codecs to produce good quality speech at

low bit rates


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low bit rates.

The numerical complexity involved in passingevery possible excitation signal through thesynthesis filter is huge

Types of AbS codecs: MPE: multi-pulse codecs RPE: Regular Pulse Excited (RPE) codec CELP: Code Excited Linear Prediction

The distinguishing feature of AbS codecs is howthe excitation waveform u(n) for the synthesisfilter is chosen

Hybrid Codecs MPE:

u(n) is given by a fixed number of non-zero pulses for every


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u(n) is given by a fixed number of non zero pulses for everyframe of speech. The positions of these non-zero pulses within

the frame, and their amplitudes, must be determined by theencoder and transmitted to the decoder. In theory it would be possible to find the very best values for all

the pulse positions and amplitudes, but this is not practical dueto the excessive complexity it would entail. In practice some sub-optimal method of finding the pulse positions and amplitudesmust be used.

Typically about 4 pulses per 5 ms are used, and this leads togood quality reconstructed speech at a bit-rate of around 10kbits/s.

Hybrid Codecs RPE:


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It uses a number of non-zero pulses to give the excitation signal u(n).However in RPE codecs the pulses are regularly spaced at some fixedinterval, and the encoder needs only to determine the position of thefirst pulse and the amplitude of all the pulses.

Therefore less information needs to be transmitted about pulsepositions, and so for a given bit rate the RPE codec can use many morenon-zero pulses than MPE codecs.

For example at a bit rate of about 10 kbits/s around 10 pulses per 5 mscan be used in RPE codecs, compared to 4 pulses for MPE codecs. This allows RPE codecs to give slightly better quality reconstructed

speech quality than MPE codecs. However they also tend to be more complex. The European GSM mobile telephone system uses a simplified RPE

codec, with long-term prediction, operating at 13 kbits/s to provide tollquality speech.

Hybrid Codecs CLEP:


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The excitation is given by an entry from a large vector quantizercodebook, and a gain term to control its power

Typically the codebook index is represented with about 10 bits (to give acodebook size of 1024 entries) and the gain is coded with about 5 bits

Toll quality speech at bit rates between 4.8 and 16 kbits/s The complexity of the original CELP codec was much too high for it to

be implemented in real-time

With large advances, now it is relatively easy to implement a real-timeCELP codec on a single, low cost, DSP chip Several important speech coding standards have been defined based

on the CELP principle, for example the American Department ofDefence (DoD) 4.8 kbits/s codec, and the CCITT low-delay 16 kbits/scodec .

Standard Speech Codecs

64 kbits/s PCM Codecs


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The 32 kbits/s G721 ADPCM Codec The 16 kbits/s G728 Low Delay CELP

Codec

The 13 kbits/s GSM Codec The 4.8 kbits/s DoD CELP Codec

PCM Codecs If linear quantization is used then about 12 bits per

sample are needed, giving a bit rate of about 96 kbits/s.


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p g g With non-linear quantization 8 bits per sample was

sufficient for speech quality which is almostindistinguishable from the original. This gives a bit rate of64 kbits/s, and two such non-linear PCM codecs werestandardised in the 1960s.

Because of their simplicity, excellent quality and lowdelay both these codecs are still widely used today. Forexample the .au audio files that are often used to conveysounds over the Web are in fact just PCM files.

Notes

Listen to an audio file compressed bydiff t d


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different codecs: http://hawksoft.com/hawkvoice/codecs.shtml

http://www.vocal.com/speech-coders/


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telephony systems

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