1 | 54 communication systems 9 th lecture chair of communication systems department of applied...

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Communication Systems9th lecture

Chair of Communication SystemsDepartment of Applied Sciences

University of Freiburg2008

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Communication Systemsadministrative stuff (lectures and exercises)

Please hand in the theoretical exercise sheets #4 And grab the new sheet (#5) due for the 24th of June 16th and 20th are lectures (all held here in this lecture room), next

practical course is 24th (again at the computer center, exact start time will be announced the lecture before)

You might refer to the electures.informatik.uni-freiburg.de for the lecture recordings and

slides in different formats

www.ks.uni-freiburg.de for details on upcoming lectures and material like the slides of the practical course (containing the “master” solutions to the exercise questions and some hints for the actual exercise sheets, the requirements on this lecture available from the first practical courses slide set)

http://www.ks.uni-freiburg.de/

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Communication Systemslast lectures

Lecture started with rather modern communication technologies and introduced the Internet Protocol as a global orientated packet switching network technology

IP can be run over very different physical media and intermediate protocols

and IP is used for more and more networked services

future IPv6 will solve any address scarcities

Very popular traditional service is telephony mostly 1:1 voice communication

with the more and more widely introduced “Voice-over-IP” we could observe a merge of both networks

and we will see a role change in the sphere of network providers (telephony companies vs. IP services providers, (TV) cable networks)

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Communication SystemsUpcoming lectures

To get an idea how traditional and modern wireless telephony networks work, we give an introduction to ISDN, GSM and UMTS

First traditional telephony networks its history and their concepts in general

Components of (digital) telephony networks Digitization of voice - PCM Then introduction to ISDN – a completely digitalized

communication infrastructure call setup and global routing in telephony networks

network definitions and standards

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Communication Systemsplan for this lecture

History of telephony networks and wireless information networks Line switching DTMF – dual tone multi frequency as traditional signalling

source Telephony protocol Standards in telecommunication Digital telephony networks – from analogous source to digitized

data streams - PCM ISDN – Integrated Services Digital Network

D channel protocol

DSS1 layer 3 protocol

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Communication SystemsHistory of telephony networks

Traditional analogous telephony networks 1848: State Telegraphy System in Prussia (Siemens)

1851: First trans-sea cable between Dover and Calais

1858: Transatlantic line-based telegraphy between Europe and America

1866: Durable transatlantic cable

1876: Bell patents the “phone” (Reiss in Germany – you will find even more inventors of the telephone, for each nation one :))

1880: 50.000 participants in US phone network

1881: Berlin opens the first “Fernsprechamt”

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Communication SystemsHistory of wireless information networks

Wireless signal transmission Morse codes transmitted by radio (Marconi)

1901: Radio-based telegraphy between Europe and the US

1914: Introducing the teletype/telex system

1915: Wireless telephony NY – San Francisco

1920: First public radio transmission in Königs-Wusterhausen

1923: Start of entertaining radio in Berlin

1929: First radio-based TV transmission (Funkausstellung in Berlin)

1935: First regular public TV transmissions in Berlin

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Communication Systemsdevelopment of telephony equipment

Traditional analogous telephony networks provides most of the standards (partly) in use up to now

bi-directional voice channel

bandwidth to carry voice around 300Hz - 3,4kHz – just the characteristics of the end user devices and their microphones and earpieces

you could hook up the old mid-thirties or sixties telephone set to your wall socket of your telephony provider or your private telephone installation

end devices are power supplied by the telephone exchange, so the devices independent of local (power) sources (which have some advantages in cases of power failures)

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Communication Systemsdevelopment of telephony equipment

Local loop – connection of the end uses device to the telephony exchange

Device is without power when hook on cradle

Call information is signalled with 65V alternating current

When off-hook power supplied at around 60V by a current of 20 – 40mA

Dial plate cuts the local loop for well defined periods to indicate dial information (~60ms cut, ~40ms closed in between – try to dial via cradle – system is rather robust in detection :-))

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Communication Systemsline switching

End systems has to be connected somehow to each other In the early beginnings manual switch boards (you know the pictures

of old films with young ladies called operators plugging wires to connect subscribers :-))

around 1975

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Switch boards first direct-dial switch boards

appeared around 1900 used in local area nets first and from around 1920 for long distance calls – dial plates (digits 1 – 9, 0) where added to the telephone device

using special relay boards with contacts for each dialed digit

system operated directly controlled until around 1960s

there is an impressive collection of old (mechanical) switchboards in the Nuremberg Communication Museum

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Communication Systemsline switching and signalling

Early phones used a hand generator to signal assistance by the operator at the switch board

Now: Identification of each end device through numerical ID composed of digits from decade system

dial plates (digits 1 – 9, 0) where added to the telephone device

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Communication Systemsautomated line switching

Switch boards - routers in the telephony world major drawbacks of this concept

route of the call is fixed

every dialed digit switches the next relay in the switching network

the (long distance) line was already occupied during call setup (=expensive for the telephone companies, especially if no-one answered the call)

Next step was introduction of indirectly operated switching networks middle of the fifties

before routing setup the dial information was collected and then processed

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Analogous electronic switching networks appeared with the beginning of the 1970s

allowed new type of dial indication

DTMF – dual tone multi frequency was introduced for dial information

inband signalling

pulse dial information has to be transported via copper wires and require rather high currents

puls dialing impossible over very long distances (resistor capacity of wire) and wireless transmission

major speedup for dialing (same amount of time spent for each digit transmitted)

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Communication SystemsDTMF

voice frequency band to the call switching center – frequencies selected in a way that no clash with “normal” voice

multifrequency shift keying (MFSK)

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Communication SystemsDTMF signalling

still in use on analogous lines and for signaling e.g. on voice menu systems – digital equipment uses out-of-band

special codes for signaling other data (e.g. Pay card identification) and for cost signaling between switching centers

some people were able to produce the needed frequencies to switch off payment or setup special connections (no cost, used by Telcos for maintenance)

“hacking/cracking” started not with computer networks but with automated telephony equipment – challenge of the 1970s was to setup connections/routes around the globe to call someone other in the same city (and enjoy the delay because of the huge distances)

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Communication Systemstelephony protocol

Key dials were introduced to telephones – special optimized layout (in contradiction to keyboard, counters layout used today)

So we have a well known “protocol of analogous telephony connection”

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Communication SystemsProtocol of analogous telephony connection

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Communication Systemsstandards in telecommunication

But in telephony world mostly not talked on “protocols” but interfaces

Interfaces are well-defined connection points where different parts of the infrastructure/equipment talk to each other in a certain way

International standardization body is ITU (International Telecommunication Union www.itu.int)

Process of standardization completely different to the workflows in Internet bodies

no bottom up, but top down decisions

exclusive club of the big (state monopoly) Telcos

high annual fees

much less information publically available then for IP and other open protocols

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Because of the old (nation state) monopolies there are many differences within the several networks Numbering schemes

Acoustical indication of dial states (busy, line-free, ...)

Different use, assignment of the (wireless) frequency spectrum

Not really compatible equipment (branch exchanges, ...) - every firm tries to use their own subset of “standards”

With the introduction of digital networks (ISDN and mobile) agreement on global standards started

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Inter connecting of voice streams has lots of technical problems Up to 1980s computerized switching centers but analogous voice

connections fault-prone to jamming and noise

regeneration means amplification of noise too Allow data connections over telephony networks Next step: Fully computerized switching centers

out of band signaling of call setup

digital voice streams allow better/perfect regeneration

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Communication Systemsdigital telephony networks - PCM

Voice/speech analogous signal continuous in time and value

domain

characterized by amplitude (signal strength) and frequency

bandwidth in traditional telephony networks 300Hz - 3,4kHz

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Sampling of a signal rate at least twice the max

frequency of analogous signal (Nyquest theorem)

2* fmaxb

= 2*3,4kHz = 6,8kHz

internationally the sample frequency was agreed on fSample

=8kHz=8000Hz=8000/s

we get a sample period of T=1/f=1/8000=125µs

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Analogous signal Continuous in value domain

Has to be translated into discrete values

A/D converter quantizes the signal

Splitting the value domain into equal intervals

Every measured value is approximated and assigned to one of the defined intervals

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PCM defines 128 different levels for positive and 128 negative amplitude of the signal

thus resolution is 256 bit Sample rate is 8000 per

second so we get 8000 Byte per

second and a bit stream of 64kbit/s

So we have the B channel bandwidth for ISDN ...

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Communication SystemsISDN – Integrated Services Digital Network

The development of digital switching networks led to standardization and integration of additional services into the same network

three virtual multiplex channels over the same two wire infrastructure

digital telephony (two independent lines on basic rate interface)

fax, telex

video telephony (H.323 devices may use ISDN as transport layer for their applications)

data communication of 64 or 128kbit/s

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Prerequisite for ISDN was digitalized infrastructure The ISDN standard was defined in the early 1980s by the ITU

several national standards evolved, 1TR6 in Germany, NI-1/2 in United States, DACS in UK, ...

DSS1 is the “EURO-ISDN” used in many other countries too available from 1993

EURO ISDN was defined by the new founded ETSI (European Telecommunication Standards Institute in 1988)

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ISDN is commonly used in all European countries since 2000 all switching centers use ISDN backends

so called “analogous” telephony devices (POTS – plain old telephony service) are converted to digital service at the local switching center

50% of the European BRI connections are in Germany

Germany has a 30% worldwide share

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Communication SystemsISDN – and the OSI protocol stack (mostly D channel)

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Communication SystemsISDN – Basic Rate Interface

BRI provides a total data rate of 160kbit/s standard end user connection

2 B channels (“bearer” - for data, digitized voice, ...) of 64kbit/s each

1 D channel (data channel for out-of-band signaling) of 16kbit/s

synchronization of 16kbit/s

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Physical layer specifications of the Uk0 operates over two-wire cable up to 5 km (depending on cable

diameter and quality)

switching center provides a 90V current to power the NTBA and one device (emergency function – to be independent on local power supply for at least one telephone)

other physical layer specifications for alternate U interfaces

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BRI network termination is defined by the Uk0 interface a special encoding (4B3T) is used: 4 bit digital to 3 baud ternary

4B3T is a "block code" that uses Return-to-Zero states

allows reduction of symbol rate to 120 kBaud (¾th) and thus distances up to 8km

reduction of low frequencies in the signal spectrum

better detection of code errors

three states: negative pulse, no pulse, positive pulse

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Next state (S1 - S4) to be transmitted is indicated in column labeled Go

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Alternate encoding: 2B1Q – 2 bit digital to 1 baud quaternary representation

2B1Q transmission can be simply described as an amplitude modulation scheme for DC pulses

Ordering of data blocks depends on the encoding used

Bits Voltage00 -2.5001 -0.8310 2.5011 0.83

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Communication SystemsUk0 – bit streams from switching center to NTBA

Each frame consists of 120 ternary steps 2*B+1*D takes 108 steps in 4 ternary blocks (tb) with 27 steps each

sync channel occupies 11 steps and a “maintenance” channel (mc) 1 step

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Communication SystemsUk0 – bit streams from NTBA to switching center

Connection is full-duplex over the two wires echo compensation and terminating set is needed

NTBA splits the data streams to separate up and down onto the S0 bus

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Instead of the traditional wall socket a NTBA (network terminal base adapter) is needed at end users site

NTBA provides the S0 bus to which end user devices are connected

unidirectional – on pair of wires for each direction

allows up to 12 wall sockets, 8 ISDN devices (or analogous devices via a/b converter)

provides device power up to 4,5W

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Communication SystemsISDN – S0

Provides the same B and D channels as Uk0 maintains the step and octet frequency

handles the device plugging and device activation, deactivation

has to be terminates with resistors of 110 Ohm

uses modified AMI code with currents of -0,75 and 0,75V

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Communication SystemsS0 – AMI code

Modified AMI code (avoid long sequences of symbols of the same type)

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Communication Systemsdata link layer for the D channel

No distinct layering for B channels – PCM or data directly put into frames as shown on previous slides

LAPD – Link Access Procedure on D channel derived from High-Level Data Link Control Protokoll (HDLC)

broadcasts only for network termination device

D2 frame margin – octet of binary pattern: 01111110 Keeping of frame sequence Error discovery Multiplexing of more than one logical D2 connections Flow control

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Communication Systemshigher layer protocols for the D channel

ITU Recommendation Q.921

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Communication Systemslayer 2 for the D channel

Flag character is part of the Header information, hexadecimal 7E

Address is two bytes (octets) long, and consists of three fields Service Access Point Identifier (SAPI)

Command/Response (C/R) bit

Terminal Endpoint Identifier (TEI)

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Communication Systemslayer 2 for the D channel

Control one or two octets (bytes) in length, indicates one of three frame formats

information

supervisory

unnumbered

Information carries Layer 3 Call Control (Q.931) data it may carry Unnumbered Information data (TEI assignment) or XID

(Connection Management/parameter negotiation) information

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Protocol handles the TEI (Terminal Endpoint Identifier) allocation all devices on S0 using the same bus and have to be addressable

TEI assignment is started by the connected devices after successful initialization of physical layer synchronization

non automatic assignment uses ID0 – 63, automatic 64 – 126

there is a special group TEI 127

Protocol elements information lowermost bit is set to 0

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Protocol elements Receive Ready - (01)

Set Asyncronous Balance Mode Extended - (6F/7F)

Unnumbered Information - (03)

Disconnect - (43/53)

Unnumbered Acknowledgement – (63/73)

Flow control uses sequence numbers for sending and receiving

00:E1:04:00:... Octets #4 for sending and #5 for receiving in the information

frame

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Communication Systemsdata link layer for the D channel error detection

D channel protocol uses rather sophisticated error detection protocol

Generates frame checksums Generator polynom

g(x) = (x +1)(x15+x14+x13+x12+x4+x2+x +1)

g(x) = x16+x12+x5+1 16 bit frame checksum Inverted residue of binary division

p1(x) = xk (x15+x14+...+x2+x +1)

p2(x) = x16d(x)

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Communication Systemsdata link layer for the D channel error detection

Checking for added or lost binary zeros Thus cyclic Hamming codes implemented Error detection for one, two and three bit error

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Communication Systemsnetwork layer for the D channel

DSS1 protocol handels the call setup of the calling and called site

Call destruction after finishing the session Restaring and parking if required Error handling

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Communication SystemsDSS1 layer 3 protocol

Protocol Discriminator part of the Layer 3 header information

single byte (octet) that is usually set to a value of 00001000 (hexadecimal "08") - meaning Q.931 call maintenance

Reference Value consists of either two or three bytes (octets) BRI systems have a 7-bit Call Reference value (127 references)

no particular end-to-end significance

Either end can assign an arbitrary value

used to associate messages with a particulary channel connection

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Communication SystemsDSS1 layer 3 protocol

Message Type single byte (octet) that indicates what type of message is being sent/received

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Communication SystemsDSS1 layer 3 protocol – message types

Message Type – four categories Call Establishment

Call Information

Call Clearing

Miscellaneous

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Communication SystemsDSS1 layer 3 protocol – information elements

Each type of message has Mandatory and Optional Information Elements, identified with single byte (octet)

bearer Capability (identifies transport requirements of the requested B-Channel)

cause (identifies reasons for disconnect or incomplete calls)

channel Identification (identifies type and number of B-Channel(s) requested)

progress Indicator (indicates status of outgoing call)

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Communication SystemsDSS1 layer 3 protocol – information elements

Network Specific Facilities (Useful for North American PRI calls - identifies network type, Carrier ID, Carrier Service Type [WATS/SDN/ASDS,etc.])

Calling Party Number (caller ID)

Calling Party Number sub address

Called Party Number (destination number, type of number[unknown], numbering plan)

Called Party Number sub address

When Information Elements (IE) consist of multiple octets, the following octet describes how many bytes (octets) are in the Information Element

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Communication Systemsliterature on telephony networks

E. Pehl, Digitale und analoge Datenübertragung See previous season lectures literature hints:

http://www.ks.uni-freiburg.de/php_termindetails.php?id=180

Next lectures: 16th , 20th June (here), next exercise 24th

http://www.ks.uni-freiburg.de/php_termindetails.php?id=180

1 | 54 communication systems 9 th lecture chair of communication systems department of applied...

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