principles of-mobile-communication-2011

Summer Training

Principles of Mobile Communication

Sub ‐ Sections

Principles of Mobile

Communication

1 PCM30 Basics

2 PDH Basics

3 SDH Basics

4 Introduction to data

5 GSM Introduction

6 CDMA Overview

7 GPRS Introduction

Part 1

PCM30 Basics

1 Introduction to PCM Pages (1-5)

2 Fundamentals of PCM Pages (1-14)

3 2 Mbit/s Frame and Signaling Pulse Frame

Pages (1-13)

4 Baseband Transmission of Digital Signals

Pages (1-18)

5 Block Diagram of a Primary Multiplexer

Pages (1-5)

6 Appendix Pages (1-5)

Sub ‐ Sections

PCM30 Basics

This document consists of 60 pages.

Chapter 1: Introduction to PCM

1


Aim of study This chapter introduces advantages of digital transmission.

Contents Pages

1 Introduction 2

2 Advantages of Digital Transmission 5


2

Chapter 1

Introduction to PCM

1 Introduction

When telephone communication began individual connecting paths were used,

i.e. a separate pair of wires was used for every telephone connection. This was

known as space-division multiplex (SDM) on account of the fact that a

multitude of lines were arranged physically next to each other. Since a

particularly large proportion of capital is invested in the line plant, efforts were

made at an early stage to make multiple use of at least those lines used for long-

range communications. This led to the introduction of frequency-division

multiplex (FDM). FDM is used in analog systems.

It is not the only way of making multiple use of lines however. Another

possibility is offered by time-division multiplex (TDM). Here the transmitted

telephone signals are separated in time. Fig. 1 shows a period containing 32 time

slots. This subdivision is repeated every 125 μs in consecutive periods. One time

slot in each of the consecutive periods is allocated to each telephone signal.

Fig. 1 Time-division multiplex


3

Sampling Theorem

The principle of time-division multiplex is based on the theory that a complete

waveform is not required in order to transmit signals such as those encountered

in telephony. It is sufficient to sample the waveform at regular intervals and to

only transmit these samples. When a waveform is sampled, a train of short

pulses is produced. The amplitude of each pulse represents the amplitude of the

waveform at the specific sampling instants. This conversion is known as pulse

amplitude modulation (PAM). The envelope of the PAM signal reflects the

original form of the curve.

Relatively large intervals occur between each sample. These intervals can be

used for transmitting other PAM signals, i.e. the samples of several different

telephone signals can be transmitted one after the other in repeated cycles.

Fig. 2 Periodic sampling of the analog telephone signal a


4

Fig. 3 PAM signal consisting of the samples of analog telephone signal a

Pulse Code Modulation

If the waveform samples, i.e. the pulses with differing amplitudes, are converted

to binary character signals, the term pulse code modulation (PCM) is used.

During this process the pulse-like samples are quantized and coded - 8 bits are

normally used here.

When the PCM signals of several telephone signals are interleaved they produce

a PCM time-division multiplex signal. PCM time-division multiplex signals

permit the multiple use of lines and electronic circuits. Moreover, owing to the

digital nature of the information, PCM signals are much less sensitive to

interference than are analog signals (e.g. PAM signals).


5

Progress in recent years in semiconductor technology has made pulse code

modulation economically attractive for telephone switching equipment. It has

thus become possible to replace the "analog" switching equipment used up to

now with fully electronic "digital" telephone systems.

2 Advantages of Digital Transmission

• Digital telephone systems offer the following advantages over analog

systems:

digital technology used throughout the system (high noise immunity).

• Multiple use of lines and exchange equipment by means of time-division

multiplex.

• Each speech direction has a separate channel (corresponding to the 4-wire

circuits used for analog systems).

• Low space requirements.

• Switching network with high traffic capacity, and negligible internal

blocking.

• Several services can be integrated within a single network: telephony, all

types of data transmission and high-speed telecopying e.g.

Advantages of Digital Telephones

• Separate digital channel for each speech direction right up to the

subscriber. This creates more favorable conditions for facilities such as

those required for hands-free operation.

• A signaling channel is always available in both directions between the

telephone and the public exchange. Features such as calling subscriber

number display, letter box function, mixed communication, etc., will thus

be possible in the all digital networks of the future.

Chapter 2 : Fundamentals of PCM

1

Chapter 2: Fundamentals of PCM

Aim of study This chapter introduces sampling theorem, analog-to-digital conversion & quantizing error.

Contents Pages

1 Fundamentals of PCM 2

2 Quantizing Error 6

3 Exercise 14


2

Chapter 2

Fundamentals of PCM

1 Fundamentals of PCM

1.1 Sampling Theorem

The sampling theorem is used to determine the minimum rate at which an

analog signal can be sampled without information being lost when the original

signal is recovered.

The sampling frequency (fA) must be more than twice the highest frequency

contained in the analog signal (fS):

fA > 2 fS

1.2 Analog-to-Digital Conversion

1.2.1 Sampling

A sampling frequency (fA) of 8000 Hz has been specified internationally for the

frequency band (300 Hz to 3400 Hz) used in telephone systems, i.e. the

telephone signal is sampled 8000 times per second. The interval between two

consecutive samples from the same telephone signal (sampling interval = TA) is

calculated as follows:

Fig. 1 shows how the telephone signal is fed via a low-pass filter to an electronic

switch. The low-pass filter limits the frequency band to be transmitted; it

suppresses frequencies higher than half the sampling frequency.


3

The electronic switch - driven at the sampling frequency of 8000 Hz - takes

samples from the telephone signal once every 125 μs. A pulse amplitude

modulated signal is thus obtained at the output of the electronic switch: a PAM

signal.

Fig. 1 Generation of a PAM signal

1.2.2 Quantizing

The pulse amplitude modulated signal (PAM signal) still represents the

telephone signal in analog from. The samples can, however, be transmitted and

further processed much more easily in digital form. The first stage in the

conversion to a digital signal - in this case a pulse code modulated signal (PCM

signal) – is quantizing. The whole range of possible amplitude values is divided

into quantizing intervals.


4

The quantizing principle is shown in fig. 2. In order to simplify the explanation

only 16 equal quantizing intervals are numbered + 1 to + 8 in the positive range

of the telephone signal and - 1 to - 8 in the negative range.

The appropriate quantizing interval is determined for each sample. Decision

values form the boundaries between adjacent quantizing intervals. On the

transmit side, therefore, several different analog values fall within the same

quantizing interval. On the receive side one signal value, corresponding to the

midpoint of the quantizing interval, is recovered for each quantizing interval.

This causes small discrepancies to occur between the original telephone signal

samples on the transmit side and the recovered values. The discrepancy for each

sample can be up to half a quantizing interval. The quantizing distortion which

may arise on the receive side as a result of this manifests itself as noise

superimposed on the useful signal. Quantizing distortion decreases as the

number of quantizing intervals are increased. If the quantizing intervals are

made sufficiently small the distortion will be minimal and the noise

imperceptible.


5

Fig. 2 Uniform quantizing of the samples of an analog telephone signal

If equally large quantizing intervals are used over the whole amplitude range,

relatively large discrepancies will occur in the case of small signal amplitudes

(uniform quantizing,). These discrepancies might be of the same order of

magnitude as the input signals themselves and the signal-to-quantizing noise

ratio would not be large enough. For this reason 256 unequal quantizing

intervals are therefore used in the practice (non-uniform quantizing):


6

• Small quantizing intervals for lower signal values.

• Larger quantizing intervals for higher signal values.

The ratio of the input signal to the possible discrepancy as a result of quantizing

is therefore approximately the same for all input signal values.

Non-uniform quantizing is specified with the aid of characteristics. The CCITT

recommends two such characteristics in G.711:

a) The "13 segment characteristic"

(A-law, e.g. for the PCM30 transmission system in Europe).

b) The "15 segment characteristic"

(μ-law, e.g. for the PCM24 transmission system in the USA).

2 Quantizing Error

Fig. 3


7

Quantizing Error

Fig. 4

Quantizing and Coding for Basic Speech Transmission Systems

The particularities of non-linear quantizing are determined by specific

characteristics described in the CCITT-recommendation G.711:


8

The 13-segment characteristic is made up of six linear sections in the positive

and negative area. The two segments located at the relative point zero form

together a linear segment. Thus, the characteristic comprises a total of 13

segments. In the proximity of point zero there are two Nr. 1 levels, a positive

and a negative one. The transmission therefore requires in all 2 x 128 = 256

levels. The 13-segment characteristic (also called A-law) is used, for example,

for the 30 channel system PCM mainly in Europe.

Each quantizing level is allocated a 8-bit code word. The first transmitted bit

determines the positive or negative sign of a sample. The following 3 bits (23 =

8) indicate one of the 7 or 8 segments. The remaining 4 bits (24 = 16) form the

code words for the linear levels within a segment.

Systems in accordance with G.711 have a sampling frequency of 8 kHz. Since

every 125 μs = 64000 bit/s = 64 kbit/s.

Load Capacity

Fig. 5


9

13-Segment Characteristic (A-Law)

Fig. 6


10

13-Segment Characteristic (A-Law)

Fig. 7


11

Conversion of the 12 bit Word into the 8 bit PCM Word

Fig. 8


12

Summary

Transmitting End

1. VF Band-pass-filter

2. Sampling (PA;

3. Quantizing, Encoding (PCM)

Fig. 9

Fig. 10


13

Summary

Receiving end

4. Decoding (PAM)

5. Holding circuit

6. VF-low pass filter

Fig. 11

PCM: Receiving End

Fig. 12


14

3 Exercise

1. What is the sampling frequency for a voice channel?

2. How many samples per voice channel are transmitted per second?

3. How many bits per sample are transmitted in a data channel?

4. How many bit/s are transmitted in a data channel?

5. How many bits are transmitted in a voice channel?

6. What is the disadvantage of uniform quantizing?

7. Is the uniform or non-uniform quantizing method used for the coding of

PCM30 voice channels?

8. What does quantizing distortion mean?

9. Name the technical terms of the quantizing methods.

10.Which quantizing method is applied for transforming a VF signal into a

PCM signal?

Why is this method used for?

11.What are the four essential steps for transforming a VF voice signal into a

PCM signal?

12. Is there any possibility for decoding a PCM30 coded signal into a PCM24

primary multiplexer; if not, why not?

Chapter 3: 2 Mbit/s Frame and Signaling Pulse Frame

1

Chapter 3: 2 Mbit/s Frame and Signaling Pulse

Frame

Aim of study This chapter introduces HDSL structure of the 2 Mbit/s frame, structure of the signaling pulse

frame & PCM transmission systems.

Contents Pages

1 Structure of the 2 Mbit/s Frame According to CCITT

Recommendation G.704 2

2 Structure of the Signaling Pulse Frame According to

CCITT Recommendation G.704 4

3 CRC4-Synchronization for Primary Multiplexer 6

4 Alarms 8

5 PCM Transmission Systems 10

6 Connecting Options of the Primary Multiplexer PCM30 11

7 Interfaces of PCM30H 12

8 Exercise 13


2

Chapter 3

2 Mbit/s Frame and Signaling Pulse

1 Structure of the 2 Mbit/s Frame According to CCITT

Recommendation G.704

2-Mbit/s-Pulse frame

In the direction of transmission the primary multiplexer PCM30 transforms up

to 30 signals with different features into 64-kbit/s-digital signals and then

combines them by the time division multiplexing procedure to a 2048-kbit/s (2-

Mbit/s)-signal, as shown in the pulse frame of fig. 1. The individual signals can

be either LF-speech signals converted by pulse code modulation, or digital

signals (e.g. data). In the receive direction a demultiplexer isolates the individual

signals out of the 2 Mbit/s signal. The 64-kbit/s-digital signals are then

converted again into analog signals.

The 2-Mbit/s pulse frame accord. to CCITT-recommendation G.704 consists of

32 time intervals with 8 bits each (octets). In the intervals 1 to 15 and 17 to 31

speech or digital signals are transmitted. Interval 16 contains the channel-

associated signaling information (CAS) combined in one multiframe or,

optionally, an additional device specific data channel. In the interval 0 there is

an alternate transmission of a frame alignment signal (FAS) or a service word

(SVW).

In order to isolate the individual signals out of the pulse frame the FAS is

searched for in the received 2-Mbit/s-signal. As soon as the bit pattern is

recognized, the demultiplexer part of the central multiplexer synchronizes itself

to time interval 0.


3

To additionally ensure the synchronization the CRC4-procedure, which will be

described in the following, is applied. The service word is used for the

transmission of urgent and non-urgent alarms (bit A and bit Sa4), for loop

commands (bits Sa6 and Sa7) (CCITT-Redbook: bits D, N and Y1 to Y3).

Fig. 1


4

2 Structure of the Signaling Pulse Frame According to CCITT

Recommendation G.704

Signaling pulse frame

If analog signal insets are used the PCM30 transmits up to 30 speech signals in

the time intervals 1 to 15 and 17 to 31 of the 2-Mbit/s-pulse frame. It has to be

ensured that the 64-kbit/s-signals in the time intervals 17 to 31 are counted as

channels 16 to 30. The individual channel-associated signaling information is

coded with 4 bits (a, b, c, d) separate from the speech signal. The signaling of 30

channels can therefore be combined in 15 octets, which are supplemented by a

code and service word of 8 bits, to a multiframe (signaling pulse frame). This

multiframe is transmitted in time interval 16 by 16 consecutive 2-Mbit/s-pulse

frames (R0 to R15). The code and service word contained in interval R0 is

necessary for the multiframe synchronization and for alarm messages.


5

Fig. 2


6

3 CRC4-Synchronization for Primary Multiplexer

With the data transmission of synchronous 64 kbit/s digital signals it is possible

that the bit patterns of the FAS and the SVW are transmitted (either randomly or

on purpose) in the time intervals defined for user signals. If there is a

synchronization of the receive side demultiplexer to this bit pattern, an isolation

of the individual signals is impossible. Therefore, the CRC4-procedure (Cyclic

Redundancy Check by 4 bits) described in CCITT-recommendation G.704 is

used in addition, to ensure the synchronization.

For this, 16 consecutive 2-Mbit/s frames are combined to a CRC4 multiframe

consisting of 2 data blocks and of the multiframe parts I and II. The highest

rating bits of the service words in the first twelve 2-Mbit/s frames form the

multiframe code word ('001011'). Here, the synchronization is based on two

criteria: finding the FAS of the 2-Mbit/s frame and the FAS of a CRC4

multiframe.

To continually supervise the synchronization, a data block (e.g. block I) is

modified in a data transmitter accord. to a certain algorithm, whereby a rest of 4

bits (the control bits C1 and C4) is left over. These bits are transmitted as

highest rating bits in the 2- Mbit frame alignment words of the following data

block (block II). The data receiver processes the incoming data block according

to the same algorithm as the transmitter. Again, a rest of 4 bits is left over,

which are compared individually to the control bits received in the next data

block (block II). In case of a correspondence, block I is considered to be error-

free.

If 915 or more out of 1000 checked blocks were found to be faulty, a new

synchronization is started.


7

A CRC4-error is indicated by two E-bits (CCITT-Redbook: Si-bits) at the

transmit side; these two E-bits are transmitted as highest rating bits of the

service words in the 2-Mbit/s frames 13 and 15 of the CRC4 multiframe. The

BER of the 2-Mbit/s-signal can be derived from the number of faulty blocks.

Thus, for example, a number of 512 or more faulty blocks within a measuring

interval of 1 s results in a BER > 10-3.

Fig. 3 2-Mbit/s-pulse frame and CRC4-multiframe


8

4 Alarms

4.1 AIS Alarm Indication Signal

D-Bit Service bit

AIS: Alarm indication signal

The AIS is an all-one-signal which, if an error occurs, is inserted as

"replacement signal" only in forward direction.

• If a low bit rate signal (64 kbit/s) is lacking at the input, the AIS is

inserted in the corresponding time slot of the highest bit rate signal (2

Mbit/s), i.e. all other time slots of the higher bit rate signal remain

unaffected.

• If a faulty signal is received at the higher bit rate interface (2 Mbit/s), the

AIS is inserted into all lower bit rate signals (64 kbit/s). A blocking signal

evaluated by the operator is inserted into telephone channels.

The higher bit rate signal is considered to be faulty if there is no signal

available, the synchronizing word is not recognized (synchr. with the FAS

or optionally with CRC4), or if the BER > 10-3.

In this case the D-bit is transmitted at the 2-Mbit/s output as feedback for

the distant end station (frame (SVW) TS0, bit 3).

• AIS can also be inserted if a device internal fault arises, such as an error

in the transmission clock. The error is determined device-specifically.

• If at the higher bit rate interface (2 Mbit/s) a signal with BER > 10-5/-6 is

received, the N-bit can be transmitted optionally (i.e. device-specifically)

at the 2 Mbit/s output as feedback for the distant end station (frame

(SVW) TS0, bit 4). In this case, no AIS is inserted.


9

• If the higher bit rate interface receives an AIS, this is through-connected

to the lower bit rate signals (64 kbit/s) and the D-bit transmitted in

backward direction.

• If the signaling multiplexer is out of order, it is possible to insert the AIS

in time slot 16 (multiframe AIS).

• If a multiframe AIS is received, the DK-bit is transmitted in backward

direction.

• If the multiframe signaling word (TS0) (=TS16 of the frame) is not

recognized, the speech signals are blocked and the DK-bit transmitted in

backward direction (multiframe TS0, bit 6).

• In case of a seizure acknowledgment alarm, the NK-bit is transmitted in

backward direction (multiframe TS0, bit 7). This alarm occurs if the

exchange receives no appropriate acknowledgment after a telephone

channel has been seized.

AIS Alarm Indication Signal

D-Bit Service bit D

Transmission of AIS and bit D

Fig. 4


10

5 PCM Transmission Systems

The transmission systems recommended by the CCITT and described below are

the PCM30 system, with 2048 kbit/s (CCITT Recommendation G.732), and the

PCM24 system, with 1544 kbit/s (CCITT Recommendations G.733); these

combine 30 and 24 telephone channels per transmission direction respectively to

form a time-division multiplex signal. PCM30 transmission systems are used

throughout Europe and in may non-European countries; PCM24 transmission

systems have been installed mainly in the USA, Canada and Japan. PCM30 and

PCM24 are also known as "primary transmission systems" or basic systems.

Their most important features are given in the figure.

Fig. 5 Characteristics of the PCM30 and PCM24 Transmission Systems


11

6 Connecting Options of the Primary Multiplexer PCM30

Fig. 6


12

7 Interfaces of PCM30H

Fig. 7


13

8 Exercise

1. How many time slots does a frame consist of?

2. In which time slot is the voice channel 14 transmitted?

In which time slot is the data channel 25 transmitted?

3. What does the time slot 16 serve e for?

4. What does the time slot 0 serve for?

5. What does the D-bit serve for?

6. How many bits does the synchronization word contain and in which time slots

is it transmitted?

7. What is the duration for the transmission of one multi-frame?

8. What is the duration of one frame?

9. How many multi-frames are transmitted per second?

10. In which part of the multi-frame are the signaling bits of voice channel 22

transmitted?

11. What does CRC4 mean?

12. What is the CRC4 code used for?

13. Is it possible to synchronize a primary multiplexer without CRC4 code?

Chapter 4: Baseband Transmission of Digital Signals

1


Aim of study This chapter introduces codirectional operation mode, contradirectional operation mode &

important PCM interfaces.

Contents Pages

1 Introduction 2

2 Interface Codes 3

3 Digital Signal Regeneration 7

4 Reasons for Bit Errors 10

5 Codirectional Operation Mode 14

6 Contradirectional Operation Mode 16

7 Important PCM Interfaces 17

8 Exercise 18


2

Chapter 4

Baseband Transmission of Digital Signals

1 Introduction

Digital signal devices process the signals as purely binary information, i.e. the

signal level does not change between bits with the same logical state. For this

reason, these so-called NRZ signals (no return to zero) can only be processed

together with the corresponding clock, which enables the identification of

individual bit positions.

This separate clock is not available for the transmission of data signals and thus it

has to be possible to derive (i.e. regenerate) the clock from the data signal on the

receiving side. It is obvious that for a NRZ code this is very complicated, if not

virtually impossible. A further disadvantage of the NRZ code is that it carries a

certain amount of dc-voltage which excludes the signal's galvanic isolation at the

interface (transformer etc.). Due to these disadvantages, various interface codes

have been developed, all of which comply with the following requirements:

• Good clock retrieval features.

• No dc-component.

Fig. 1 Processing of NRZ signals with the aid of separate clock


3

2 Interface Codes

A suitable interface code has a maximum of transitions between the different

signal levels, even for the transmission of lengthy sequences of identical logical

states; it has no dc-component. The survey shows the development of individual

codes.

A rather important advantage of the interface code is the possibility it offers to

detect transmission errors by supervising the coding rules. With the HDB3 code,

for example, the receiving of four zero bits would represent the violation of a

coding rule, i.e. at least one bit error must have been occurred during

transmission.


4

The standardization of interface codes only refers to device interfaces. The codes

for conductor-bound transmission paths are manufacturer-dependent and are

generally adapted to the requirements of the respective terminating unit.

Digital Interface Codes

Fig. 2

Fig. 3 shows the amplitude spectrum of various interface codes. For codes

without a dc-component the maximum energy is within the range of a frequency

which corresponds to half of the bitrate value. This is obvious when comparing

the definitions of frequency and bitrate respectively.


5

Fig. 3 Amplitude spectrum of various codes

The bit sequence represented in fig. 4 shall serve as an example. One signal

period covers 2 bits and corresponds to the basic wave of the data signal. This

wave contains the greatest amount of energy and has a frequency which equals

half of the bitrate value. This is also the frequency that is indicated by a frequency

counter connected to a source of a digital signal.

Fig. 4 Bit sequence 0101...


6

HDB3-Coding rules

(Third-Order-High-Density-Bipolar-Code)

The HDB3-code is a modified version of the AMI-code. Binary signals or AMI-

code signals may contain lengthy "0" sequences, which hinder the clock retrieval

in the regenerative repeaters along digital transmission paths. The HDB3 code

enables the elimination of "0" sequences with more than 3 zeros.

1. If there are more than 4 consecutive "0"-signal elements, the fourth "0"-

signal element shall be replaced by a V-signal element (= "1"-signal

element) (000V). Hereby, the V-signal element takes on the same polarity

as the "1"-signal element. A V-signal element causes a Violation of the

AMI-rule.

2. If between the V-signal element, inserted according to the conditions

specified above (rule 1), and the preceding V-signal element there is an

even number of "1"-signal elements, then the first of four "0"-signal

elements shall be replaced by an A-signal element (= "1"-signal element).

The polarity of the A-signal element complies with the AMI rule. The last

of four "0"-signal elements becomes again a V-signal element (A00V). In

this case the A- and V-signal elements have the same polarity.


7

Fig. 5 Transformation of two binary signals into HDB3-signals

3 Digital Signal Regeneration

The digital signal regeneration is one of the advantages of the digital transmission

technique. Theoretically, it enables the signals to be transmitted via an unlimited

distance without any quality losses.

During transmission, a digital signal is attenuated and distorted, which results in a

reduction of the signal/noise ratio. The regeneration process has the task of

canceling such distortions and regenerate the originally sent signal from the

actually received signal. That is why every interface on the receiving side is

followed by a regeneration circuit.


8

Fig. 6 Principle of digital signal regeneration

Four basic function blocks are necessary for the digital signal regeneration:

• Amplification block (balancing of attenuation losses).

• Clock retrieval block.

• Amplitude decision block.

• Time decision block.


9

Fig. 7 Block diagram of a digital signal regenerator

These four functions are represented in fig. 7.

• The receiving signal is fed into an automatic gain controlled amplifier

(AGC) which keeps the amplitude of the outgoing signal at a constant

value over a wide range of incoming amplitudes. Thus, the attenuation of

the transmission path is balanced.

• The constant output level is a precondition for the functioning of the

amplitude decision block (AD) which follows. This AD decides on the

basis of an internal threshold value whether the level of incoming signal is

above or under this threshold value. Accordingly, a signal with the levels

Log. 0 or Log. 1 is emitted at the output. The output signal thus consists of

pulses, the width of which only depends on the period during which the

output signal exceeds the decision threshold.


10

• The time decision block (TD) has the task of generating signal pulses with

constant width. For this, it requires the regenerated receive signal clock

which samples the output signal of the amplitude decision block. If, at the

time of sampling the signal has a level of Log. 1, the time decision block

emits a pulse with constant width. Thus, incoming pulses of any width are

turned into pulses corresponding exactly to the bit width of the transmitted

signal. The time decision process is the final stage of regeneration.

• The clock retrieval CR block is in charge of regenerating the transmitted

signal clock from the receive signal clock. In order to effect this function, a

phase locked loop (PLL) is employed, basically consisting of a voltage-

controlled oscillator whose frequency can be changed by a control-voltage.

By adequate evaluation of the receiving signal it is now possible to reach a

control voltage which can set the oscillator to the exact clock frequency

value of the transmitting signal.

4 Reasons for Bit Errors

The decisive quality criteria for the transmission of digital signals is the bit error

rate (BER). This BER represents the proportion of bits which have been mutilated

(i.e. incorrectly recorded) during transmission, to the total amount of bits

transmitted within a certain interval. The BER directly influences the quality of

the transmitted services (e.g. voice channels, data channels, video signals). Two

significant BER are explained exemplary in the following:


11

BER = 10-6

This BER virtually cannot be perceived in a voice channel. For the transmission

of data channels, however, this value represents the maximum acceptable limit.

The transmission system is in a state of "degraded quality", which is indicated by

a degradation alarm (low priority) on the devices involved. The transmission path

remains, nevertheless, in operation.

BER = 10-3

This BER causes a strong interference noise in a voice channel. The operating

state is judged to be of "unacceptable quality", which is signaled by the devices

involved by the emission of a failure alarm (high priority). The transmission path

goes out of operation.

How do bit errors arise?

In the previous section it was mentioned that digital signals can be regenerated as

requested, i.e. a transmission without quality reduction is possible. This statement

is, however, only partially true, i.e. whenever the impairment of the transmission

signals is within limits which still permit the regeneration at the receiving side. The

reasons for the formation of bit errors are

• Low signal/noise ratio.

• Jitter.

• Intersymbol interference.

Low signal/noise proportion

Noise amplitudes which influence the amplitude decision process are

superimposed to the originally sent signal.


12

The superimposed interference peaks lead to an incorrect signal interpretation at

the receiving end. Reasons for a low S/N-ratio are:

a) Too strong signal attenuation during transmission.

b) External interference during transmission.

For transmission in cable sections (especially optical fiber) both reasons can be

largely eliminated by careful planning.

Fig. 8 Low S/N-proportion

Jitter

Due to jitter, the transitions between signal levels log. 0 and log. 1 do not take

place at periodically recurring points in time (characteristically moments) as for

undisturbed signals, which means that the transitions oscillate around the

characteristically moments.

Jitter is characterized by jitter amplitude (unit intervals UI) and jitter frequency.

One UI means that, because of deviation from the characteristically moments, the

signal edges are within a range equal to the width of 1 bit.


13

The jitter frequency is the number of oscillations around the characteristically

moment per one second. Jitter influences the time decision process in the

regenerator and causes bit errors for high jitter amplitudes and frequency.

Fig. 9 Representation of an Unit Interval (UI)

Jitter arises in the devices used for signal transmission (i.e. in regenerators and

demultiplexers = systematically jitter), or on the transmission path due to external

influences (non-systematic jitter).

Intersymbol interference

Is caused by a discrepancy between the band width of the transmission path and

the bandwidth required for the digital signal. This leads to a bit extension, so that

there is an overlap of bits which follow each other. Thus, bit errors occur, the

reasons of which can be traced back to the impairment of amplitude decision

process. For conductor-bound transmission of digital signals this effect can be

excluded by adequate planning. For transmission on radio paths this effect is of

fundamental importance as the frequency response of the transmission path can

change due to atmospherical influence.


14

5 Codirectional Operation Mode

Codirectional

Designation of an interface between two devices A and B where the clocks T and

T' are transmitted in the same direction as the digital signals S and S' to which

they belong (opposite term: contradirectional).

Fig. 10


15

Fig. 11 Interface code G.703 (64 kbit/s)


16

6 Contradirectional Operation Mode

Contradirectional

Designation of an interface between two devices A and B, where the clocks are

supplied only by the one device B. Thus the clock T' belonging to the signal S'

(from B to A) is transmitted in the same direction as the signal.

Remark: with a contradirectional interface the device B (e.g. PCM-multiplexer)

requests a digital signal from the device A (e.g. device which bundles switching

data).

Fig. 12


17

7 Important PCM Interfaces

LF interface F2:

Speech frequency band 300 to 3400 Hz

Resistance for 2-wire operation 850 Ω sym. or

900 Ω sym.

Resistance for 4-wire operation 600 Ω sym.

Level variable

64-kbit/s-, 128-kbit/s-data signal interface D2:

Codirectional operation (G. 703/1.2.1)

Bit rate 64 kbit/s 128 kbit/s

Baud rate 256 kbaud/s 512 kbaud/s

Code AMI

Resistance 120 Ω sym.

Amplitude at the output 1 Vs0

Contradirectional operation (G. 703/1.2.3)

Bit rate 64 kbit/s

Code AMI

Resistance 120 Ω sym.

Amplitude at the output 1 Vs0

Clock signal 64 kHz

Resistance (clock signal) 120 Ω sym.

Amplitude at the output (clock signal) 1 Vs0


18

2-Mbit/s-interface (G703/6) F1:

Bit rate 2048 kbit/s 50 ppm

Code HDB3

Resistance 120 Ω sym. or

75 Ω coaxial

Amplitude at the output 3 Vs0 sym. or

2.37 Vs0 coaxial

8 Exercise

1. What demands are made on the transmission codes?

2. Which two modes of operation are used for data channels?

3. Which symbol rate (baud/s) has the data thus transmitted?

Chapter 5: Block Diagram of a Primary Multiplexer

1


Aim of study This chapter introduces transmit & receive side.

Contents Pages

1 Transmit Side 2

2 Receive Side 3

3 Exercise 5


2

Chapter 5

Block Diagram of a Primary Multiplexer

1 Transmit Side

The required functional entities of the DSMX 64 K/2F are accordingly

subdivided into a transmit section and a receive section. The transmit section

incorporates the transmit unit and transmit-side speech circuits in the telephone

channel units or the transmit-side circuits of the data channel units; the receive

section comprises the receive unit and the receive-side speech circuits in the

telephone channel units or the receive-side circuits of the data channel units.

Fig. 1 Block diagram of transmit side, showing functional blocks


3

Telephone Channel Unit

The transmit-side speech circuit takes over the telephone signals present on the

associated telephone lines (VF signals). The signals are band-limited from 300

Hz to 3400 Hz.

Depending on the required relative level setting, the signal is amplified or

attenuated before the A/D conversion. The relative level in transmit direction is

referred to the input of the A/D converter. Thus a relative level setting of -14

dBr means that the level before the amplifier is 14 dB less than at the input of

the A/D converter. With other terms the signal is amplified by 14 dB.

The resulting PAM values (pulse amplitude modulation) are converted into 8-bit

code words by the encoder. This non-linear quantizing is amplitude-dependent.

The encoding characteristic (A-law), which is symmetrical with respect to the

zero line, consists of 13 linear segments, giving an approximately logarithmic

response.

2 Receive Side

The 2 Mbit/s PCM-E signal is fed to all the channels and the signaling

multiplexer. The addressing, which is derived from the clock generator, ensures

that the individual 8-bit words are read into the associated channel units or the

signaling multiplexer with the correct timing.

The functional blocks synchronization and sections of the clock generator and

distributor are concentrated in a highly integrated I2L device mounted in the

transmit unit (PCM receive device).


4

Fig. 2 Block diagram of the receive side

Telephone Channel Unit

The PAM signal is generated from the 8-bit words in the decoder. A low-pass

filter with (sin x) / x equalization reconverts the sample-and-hold signal formed

from the sequence of PAM values to the VF signal.

Depending on the required relative level setting the signal is amplified or

attenuated after the D/A converter. The relative level in receive direction is

referred to the output of the D/A converter. Thus a relative level of 4 dBr means

that the level after the amplifier is 4 dB higher than at the output of the D/A

converter. With other terms the signal is amplified by 4 dB.


5

Data Channel Unit

The 8-bit words are read into a memory with the 2-MHz receive clock, undergo

serial-parallel conversion and are then read out again at 64 kHz. The 64 kbit/s

signal is then encoded and fed out from D2out at 256 kbaud.

3 Exercise

Which blocks of the primary multiplexer may be distinguished for the

application of voice channel transmission and data channel transmission?

Chapter 6: Appendix

1

Chapter 6: Appendix

Contents Pages

1 Levels 2

2 Formulas 2

3 Conversion from the Power Level to the Voltage Level and Vice Versa

3

Chapter 6: Appendix

2

Chapter 6

Appendix

1 Levels

The Absolute Level

The absolute level is a logarithmic value which shows the difference between the measured value and the reference value. Reference Values (relative point zero)

2

2 Formulas Absolute power level: nPabs = lg (Pm/1 mW) [B]

nPabs = 10 lg (Pm/1 mW) [dBm]

B=Bel; Pm = measured power

Absolute voltage level: nUabs = 20 lg (Um/0.775 V) [dBu]

10 lg (Pm/1 mW) = 10 lg ((Um2x600 Ω) (0.7752V2xRm))

20 lg (Um/0.775 V) + 10 lg (600 Ω/Rm)

Chapter 6: Appendix

3

Absolute current level: nlabs = 20 lg (lm/1.29 mA) [dBi]

10 lg (Pm/1 mW) = 10 lg ((lm2xRm) / (1.292 mA2x600 Ω)) =

20 lg (lm/1.29 mA) + 10 lg (Rm/600 Ω)

3 Conversion from the Power Level to the Voltage Level and Vice Versa

10 lg (Pm/1 mW) = 20 lg (Vm/0.775 V) + 10 lg (600 Ω/Rm) absolute absolute correction power level voltage level factor level

TIP With a resistance of 600 Ω, the absolute levels of voltage, current and power have the same value - if the recommended standard values are used! In the field, the absolute power level and the absolute voltage level are used for telecommunication path measurements. The absolute power level may be calculated with the voltage level and the level correction factor according to the above formula. Important level correction factors ncor:

Rm [Ω] ncor [dB] 3000 -6.989 2400 -6.0206 600 0 = 0 250 3.82 150 6.0206 = 6 75 9.0309 = 9 50 10.79 35 12.43

Chapter 6: Appendix

4

dB level relation (e.g. attenuation or gain) dBr relative level

referred to the zero relative level point,

dBm absolute power level, referred to 1 mW dBu/dBv absolute voltage level, referred to 0.775 V dBm0 absolute power level

referred to the relative level (dBr), relation: dBm0 = dBm - dBr

dBm = dBr + dBm0

dBv0 absolute voltage level referred to the relative level (dBr), relation: dBv0 = dBv - dBr dBv = dBr + dBv0

dBmp absolute (noise) power level, referred to 1 mW and CCITT weighted, weighted = pondered i.e. measured with a psophometer, A-filter

dBm0p absolute (noise) power level, referred to the relative level and CCITT weighted relation: dBm0p = dBmp - dBr

dBmp = dBr + dBm0p

dBrnc absolute noise level, "C-characteristic" weighted, reference noise c-characteristic weighting

dBa absolute noise level, "FIA-characteristic" weighted, adjusted weighting acc. to FIA characteristic

dBV absolute peak-to-peak voltage level for TV signal measurements relation: OdBV = 1 Vpp/75 Ω

= 2.2 dBm/75 Ω = -6.8 dBv/75 Ω

Chapter 6: Appendix

5

dBvs absolute voltage level in the sound channel, referred to 0.775 V

dBvps absolute (noise) voltage level in the sound channel referred to 0.775 V and CCITT weighted

dBv0ps absolute (noise) voltage level in the sound channel, referred to the relative level (dBr) and CCITT weighted.

Part 2

PDH Basics

1 Application of Plesiochronous Multiplex Systems

Pages (1-8)

2 Time‐Division Multiplexing of Digital Signals

Pages (1-19)

3 Frame Structure of the Digital Signal Hierarchies 2..4

Pages (1-8)

4 Functional Description of Multiplexer/Demultiplexer

Pages (1-6)

5 Baseband Transmission of Digital Signals

Pages (1-14)

Sub ‐ Sections

PDH Basics


Chapter 1: Application of Plesiochronous Multiplex Systems

1

Chapter 1: Application of Plesiochronous Multiplex

Systems

Aim of study This chapter introduces digital signal hierarchies.

Contents Pages

1 Introduction 2

2 Digital Signal Hierarchies 3

3 Connecting Options for the Digital Multiplex Systems 5


2

Chapter 1

Application of Plesiochronous Multiplex Systems

1 Introduction

Digital multiplexers are applied wherever a high transmission capacity with

effective use of transmission paths to be realized.

The basic idea of multiplexing is the time-interleaving of digital signals of

different sources i order to form a common signal with a bitrate which is

correspondingly higher (multiplex process). On the system's receiving side the

appropriate separate signals are reobtained from the sum signal (demultiplex

process). This means that the original digital signals of the multiplexed signal

sources are available again at the output of such a system.

Example:

The output signals of four PCM30 systems are combined to a signal of 8 Mbit/s

and transmitted via a common transmission path to the receiving side (multiplex

procedure).

On the receiving side the sum signal is then distributed to the corresponding

input of PCM30 systems (demultiplex procedure).

In this example only one direction of transmission is shown.


3

Fig. 1

2 Digital Signal Hierarchies

2.1 Multiplex Hierarchy (CEPT)

The European plesiochronous digital hierarchy (CEPT-standard) is based on a

2048 kbit/s digital signal (stage 1) which may come for example from a PCM30

system, a digital exchange or from any other device in accordance with this

interface norm (standard). Starting from this signal the next higher hierarchies

are formed, each having a transmission capacity which is four times the previous

one.


4

The multiplying factor for the bitrates is greater than four, as for each hierarchy

level additional bits for pulse frame generation and other additional information

are inserted.

Fig. 2

2.2 Multiplex Hierarchy PCM24

The plesiochronous hierarchy used in USA and Japan is based on a 1544 kbit/s

digital signal (PCM24). From the table below the structure of superordinate

hierarchy levels can be seen.

Only the CEPT-hierarchy will be dealt with in the following.

Fig. 3


5

3 Connecting Options for the Digital Multiplex Systems

Each multiplexer normally has

• 4 inputs/outputs for the lower hierarchy level (Tributaries).

• One input/output for the higher hierarchy level. As to Siemens systems,

the lower hierarchy level is termed F2-side (secondary side), the higher

one F1-side (primary side).

The inputs/outputs 1 to 4 can be connected with any type of system which is in

accordance with the corresponding CCITT interface conditions.

Some examples for the connection of the individual multiplexers are represented

on the following pages.

Connecting Options for the Multiplex System 2/8 Mbit/s

Fig. 4


6

Connecting Options for the Digital Multiplex Device DSMX 8/34 Mbit/s

Fig. 5


7

Connecting Options for the Digital Multiplex Inset DSMX 2/34 Mbit/s

Fig. 6


8

Connecting Options for the Digital Multiplex Inset DSMX 34/140 Mbit/s

Fig. 7

Chapter 2: Time‐Division Multiplexing of Digital Signals

1

Chapter 2: Time-Division Multiplexing of Digital

Signals

Aim of study This chapter introduces basic methods of multiplexing and basic pulse frame structure.

Contents Pages

1 Basic Methods of Multiplexing 2

2 Synchronization between Transmitting End and Receiving End 4

3 Definition of Plesiochronous Digital Signals 6

4 Clock Alignment of Plesiochronous Signals 9

5 Basic Pulse Frame Structure 11

6 Realization of the Positive Justification Method 13


2

Chapter 2

Time-Division Multiplexing of Digital Signals

1 Basic Methods of Multiplexing

For the generation of the sum signal out of the individual separate signals the

following two methods may be used:

Code word interleaving

With this method code words of the individual separate signals (i.e. bit

combinations having some kind of relation between each other) are arranged one

after the other in a time sequence. Such is the case for the generation of a 2-

Mbit/s-signal, where the 8 bit binary words of the coded PCM-voice channels

are transmitted sequentially in a 125 µs cycle.

This figure shows the code word interleaving of two separate signals with a

word length of four bits.

Fig. 1


3

Bit-by-bit interleaving

This method is used for all systems beyond the 2 Mbit/s hierarchy. Here a cyclic

transmission sequence is applied, where only one bit of each separate signal is

transmitted. This means that the signal of a certain multiplexer input appears

only in every fourth bit of the sum signal.

The figure shows the bit-by-bit interleaving of two separate signals.

Fig. 2

Two basic cases can be distinguished with multiplexing:

1. The original signals are synchronous, i.e. their clocks are exactly the same.

This is valid for a PCM30 system, where the clocks of the individual 64-

kbit/s-signals and the 2 Mbit/s-clock are derived from a central system

clock. In this case the multiplexing process is restricted to a simple

parallel-to-serial conversion of the 8 bit code words.


4

2. The original signals are not synchronous, i.e. their clocks come from

different sources. This is valid for the multiplexing of output signals,

originating from various PCM30 systems their clocks being generated in

each system in an autonomous way. Here it is necessary to take

appropriate measures in order to compensate the occurring clock

differences.

2 Synchronization between Transmitting End and Receiving

End

For each type of multiplexing it has to be ensured that the sum signal can be

resolved into the individual original signals (demultiplexing process). The

receiver of the sum signal thus has to know which bits are assigned to the

individual subsystems. To allow for this, a fixed bit combination, the so-called

frame alignment word (FAW) is inserted by the transmitting system in

periodically recurring intervals into the sum signal.

If the receiver detects the frame alignment word in the received signal it is

possible to perform the assignment of the following bits to the subsystems by

means of the regenerated receiving clock.

The time intervals between the beginning of a FAW and the beginning of the

following FAW are called pulse frames.


5

Fig. 3

2.1 Recovery of Frame Alignment

During recovery of frame alignment (e.g. during initial commissioning of a

system) the receiver continuously examines the incoming signal upon

occurrence of the FAW. If this FAW is detected for the first time, the receiver

expects a renewed occurrence only after the specified pulse frame period has

elapsed (counting of the receiving signal clocks). In this case the process will be

repeated; the synchronization is established. Otherwise, the system takes the

continuous searching up again. This procedure ensures that a synchronization to

a bit combination, which accidentally has the same content as the FAW, is

excluded.


6

2.2 Loss of Frame Alignment

Only if the FAW does not appear in the expected positions for several

consecutive times (e.g. four) the frame alignment is supposed to be lost. This

guarantees that in case of transmission errors the system does not perform an

immediate desynchronization.

For each faulty frame alignment word a pulse is produced, which can be used

for the estimation of the bit error rate (see also chapter 6, in-service

measurement of bit error rates).

3 Definition of Plesiochronous Digital Signals

Supposed a data source (S) transmits a digital signal with a bitrate fS to a data

drain (D). The data drain decides with the aid of an internally generated clock

frequency fR whether the incoming signal is zero or one in the moment of the

clock pulse. The two clock signals fS and fR are thus generated in different

places and although they do have the same nominal frequency, they will always

differ from each other to a certain extent.

Definition:

Data signals are termed plesiochronous if their clock rates have the same

nominal value, but may differ from each other within certain tolerance ranges.


7

Fig. 4

The effects of these clock deviations are represented in the two figures below:

Sampling clock fR > transmission clock fS

Two sampling instants are within one bit interval of the transmitting signal. The

data drain (D) interprets this situation as double transmission of bit a5.

Fig. 5


8

Sampling clock fR < transmission clock fS

One transmitted bit is between two sampling instants. Bit b5 not detected by the

data drain (D).

Fig. 6

Plesiochronism during Multiplexing Process

The multiplexing process may be represented with the aid of the following

figure.

A rotating pointer samples the feeder links (tributaries) for the separate signals

with a frequency which is four times higher than the nominal bitrate

fS (fR = 4 X fS), i.e. each digital signal is sampled with a nominal fS. As both, the

digital signal sources (S1...S4) as well as the sampling frequency (fR) are

generated by different clock sources, the result is a plesichronous state of

operation for every feeder link.


9

Example:

The signal sources (S1...S4) are PCM30 devices transmitting with their

individual transmission clock a 2 Mbit/s-signal with clock tolerances to the

inputs of a 2/8 multiplexer.

Fig. 7

4 Clock Alignment of Plesiochronous Signals

During multiplexing of plesiochronous digital signals the so-called positive

justification method is applied, which is based on the following principles:

• A bitrate for each subsystem is provided in the multiplex signal, which is

somewhat higher than the subsystem’s nominal bitrate. This means that

the transmission capacity is systematically higher than actually needed.

• The difference between the bitrate of the subsystem and the multiplex

bitrate per system is compensated for each channel by the justification

bitrate, which does not contain any information and serves only for the

compensation mentioned above.


10

• The justification bitrate is thus always adjusted to the difference between

the bitrate of the subsystem and the multiplex system and thereby

compensates for each channel the tolerance between the tributary signal

bitrates and multiplex signal bitrates.

Example:

Fig. 8

• The signal sources S1..S4 emit signals with a nominal value of 2048

kbit/s.

• The sampling pointer rotates with a frequency of fR = 2052 kHz, i.e. the

transmission capacity per channel is 4 kbit/s higher than the nominal

bitrate of the subsystem.

• Supposed the signal sources transmit the following actual bitrates:

S1 : fS1 = 2048.1 kbit/s

S2 : fS2 = 2048.05 kbit/s

S3 : fS3 = 2048.0 kbit/s

S4 : fS4 = 2047.9 kbit/s


11

This results in the following justification bitrates:

For channel 1: 2052 kbit/s - 2048.10 kbit/s = 3.90 kbit/s

channel 2 : 2052 kbit/s - 2048.05 kbit/s = 3.95 kbit/s



Thus, the resulting signals at the rotating pointer’s sampling points are

synchronous. The multiplexing procedure can be performed without the former

discussed problems of omission or double sampling of individual bits.

5 Basic Pulse Frame Structure

How is a variable justification bitrate realized?

The signals of higher hierarchy levels are transmitted within a predetermined

frame structure, the same as for the 2 Mbit/s signal of the first hierarchy level.

This frame begins with a frame alignment word of fixed length and content in

order to allow on the demultiplex side of the system an allocation of the

following bit-interleaved tributary bits to the appropriate channels. In addition,

the frames of the plesiochronous hierarchy contain one bit position per

individual signal, which is either used for the transmission of a tributary bit, or

not used at all. This bit position is called justification bit. By alternate use/non-

use of this bit position, the transmission capacity for the individual signals may

be varied to some extent.

This process is called positive pulse justification; thus, the non-use of the

justification bit position corresponds to an increase in the justification bitrate (=

decrease in the transmission capacity), whereas the use of the justification bit

position has the opposite effect.


12

Fig. 9

The receiving end of such signals requires information on how the justification

bit position has been used (non-information bit or tributary bit). To allow for

this, there are justification service bits arranged before the justification bits in

the time sequence. The content of the justification service bits indicates how the

following justification bit position has to be interpreted. If, for example, the

content of the justification service bit for channel 3 is a binary one, the receiver

ignores the following justification bit positions of channel 3. The other way

round (JS3 = 0), the position JB3 is interpreted as tributary bit.

Example:

The frame structure in a 8 Mbit/s pulse frame:

Frame duration: 100.38 µs

Overall number of bits in blocks TB1:200 bit, TB2:208 bit, TB3:208 bit,

TB4:204 bit or 208 bit.

This results is an actual bitrate/channel


13

This is the bitrate /channel if the justification bit position is always unused.

If every justification bit position is used for a tributary bit of the separate signal

the following actual bitrate/channel is calculated:

By alternate use/non-use of the justification bit position in the frames the

transmission capacity for the individual channels in this example may be varied

within a range of 9.962 kbit/s.

6 Realization of the Positive Justification Method

6.1 The Elastic Store (Multiplex-Side)

How can the justification process be realized?

An elastic store consists of a number of 1 bit memory cells (typ. 12) which can

be written in and read out independently of each other (i.e. at the same time it is

possible to write in one cell, while another is read out). The incoming separate

signal with its own clock is written in the cells 1...8, 1...8 etc. in a cyclic way.

The store is read out with a clock, generated in the multiplexer; a clock which is

systematically higher than the bitrate of the separate signal. The difference

between write address and read address is monitored by an address comparator.

It goes without saying that the write address always has to be ahead of the read

address. Due to the greater read out velocity the read address continually

approaches the write address. If the difference between the two becomes < 3

memory cells, the comparator releases a signal.


14

Then the following procedures are started:

If the justification service bit position in the frame is reached, the bit is set to

one. On reaching the justification bit position, the read address is maintained for

one clock period and the actual memory cell is read out once more. This is the

justification bit which is ignored at the receiving end. By maintaining the read

address during one clock cycle the difference between the addresses increases

and the whole procedure is repeated in the same way. Thus, the plesiochronous

clock rate of the channel is matched to the multiplex bitrate.

Between the initiation of the justification process (comparison of addresses) and

its execution there may be an interval of max. 1 frame period, within which the

read address approaches the write address more and more. That is why the

justification process is initiated already when the address spacing is smaller than

3, in order to ensure a reserve against memory overflow, e.g. an empty memory.

Each channel is assigned an elastic store. As the read out clock for all channels

come from the same clock supply (in the multiplexer), the output bitrates of the

elastic stores are synchronous. The actual multiplexing procedure is thereby

continued to a simple parallel-to-serial conversion of the output signals of the

elastic stores for the four separate signals.


15

Fig. 10 Principle of an elastic store


16

Fig. 11 Block diagram of an elastic store

Realization of the positive justification method

Positive justification method: f2 > f1

Fig. 12


17

Example (see also fig.12):

• The bitrate of the input signal shall be f1 = 2048 kbit/s.

• The pulse frame of the multiplex signal shall be 100, 38 µs and contains 1

justification bit per channel.

• The read out timing rate shall be 2052 kbit/s.

The reading pointer would overpass the writing pointer (2052 kHz-2048

kHz = 4 kHz) 4000 times per second. That is why on average one

justification bit is inserted every 250 µs (1/4 kHz = 250 µs). For a pulse

frame of 100, 38 µs, this means that one justification is effected on

average in every 2,5th frame (250 µs/ 100, 38 µs) (2 in 5 frames).

• The bitrate of the input signal shall now be T1 = 2047, 90 kbit/s.

Now the justification must be effected every 243, 90 µs, i.e. in every 2,4

pulse frame.

• The bitrate of the incoming signal shall be T1 = 2048, 10 kbit/s.

A justification is required every 256, 40 µs, i.e. in every 2, 56 frame

6.2 The Elastic Store (Demultiplex-Side)

The task of the demultiplexer is to distribute the sum signal in the right sequence

to the output of the separate signals. Therefore, the incoming multiplex signal is

divided into 4 separate signals by means of parallel-to-serial conversion. By

control of the frame alignment signal the 4 separate signals can be assigned to

the right channels. Besides, the justification service bits and justification bits can

be identified (by counting the bits transmitted since the beginning of the frame).

By means of this information the justification process is canceled, i.e. all bits

which do not come from the original signal are removed from the separate

signals.


18

Thus, a signal with timing gaps instead of the removed bit positions is

generated. In order to guarantee a continuous signal at the outputs, elastic stores

are used on the demux-side to smooth the signal.

For this, the incoming datas signal is written into the store with the gap timing

and read out of the store with a continuous timing which corresponds to the

average value of the gap timing; thus the signal is forwarded in a smoothed

condition to the outgoing subsystem interface.

Fig. 13 Principle of an elastic store (demultiplex-side)


19

A continuous timing is generated from the gap timing by means of a phase-

locked loop (PLL). For this, a voltage-controlled oscillator is synchronized to

the gap timing frequency. If the critical frequency of the control loop is selected

sufficiently low (low-pass filter) it is ensured that the voltage-controlled

oscillator adjusts itself to the average value of the gap timing frequency.

6.3 Jitter caused by Multiplexers

The gap in the write clock of the elastic store result in phase shifts on the input-

side of the PLL’s phase comparator, which are converted to voltage shifts. These

voltage shifts are smoothed by the low pass filter of the PLL, but they can never

be smoothed perfectly.

That is why the smoothed clock of the control voltage will vary accordingly also

at the output of the PLL circuit, i.e. jitter is generated. The jitter in the output

signal depends on the system. The highest jitter frequency is determined by the

limit frequency value of the PLL low-pass filter.

Chapter 3: Frame Structure of the Digital Signal Hierarchies 2..4

1

Chapter 3: Frame Structure of the Digital Signal

Hierarchies 2..4

Aim of study This chapter introduces frame structure of 8, 34, 140 Mbit/s hierarchies.

Contents Pages

1 Frame Structure of 8, 34, 140 Mbit/s Hierarchies 2

2 Timing Sequence of the Multiplex Process 6

3 Transmission of Additional Data Channels with Y-Bits 8


2

Chapter 3

Frame Structure of the Digital Signal Hierarchies 2..4

1 Frame Structure of 8, 34, 140 Mbit/s Hierarchies

Alignment word

The frame of all hierarchy levels begin with the frame alignment word (FAW),

by means of which the receiving system (demultiplexer) detects the beginning of

the frame and is thus able to interpret the following bit positions correctly.

Besides, an in-service-supervision of the incoming signal’s bit error rate can be

performed by continuous evaluation of the FAW.

Signaling bits D, N

Immediately after the FAS the signaling bits D and N are transmitted. They

provide information about the state of the opposite transmission direction.

Hereby, urgent alarms (failure) are signaled via the D-bit (remote alarm

indication RAI), and non-urgent alarms (interference) via the N-bit. If it is

possible to renounce the backward transmission of non-urgent alarms, the N-bit

can be used for the asynchronous transmission of external data (so-called Y-data

channels via V.11 interface).

Blocks TB

Here the signals (tributary bits) of channels 1...4 are transmitted bit-by-bit

interleaved.


3

Blocks JS

These blocks consist of 4 bits and contain the justification service bits of

channels 1...4. In order to provide a protection against transmission errors, the

justification bits are transmitted in a redundant way and evaluated on the

receiving end by majority decision. The 3 JS blocks (5 at 140 Mbit/s) contain

the same information in the bit error-free state. If, due to a transmission error,

one of the justification service bits (2 at 140 Mbit/s) is wrongly detected, the

majority decision nevertheless allows the correct evaluation of the following

justification bit positions. A wrong interpretation of the justification bit position

would inevitably result in a desynchronization of the affected subsystem.

Fig. 1


4

Probability of a loss of synchronization depending on the bit error rate due to

• Loss of the justification information: (2 J) at 8 Mbit/s

(3 J) at 140 Mbit/s

• Loss of the frame alignment: (2) at 8 Mbit/s

(3) at 140 Mbit/s

Block JT

This block contains the justification bit positions (justifying bit or tributary bit)

and is integrated into a TB block. By use respectively non-use of this bit

position, the transmission capacity is matched of the individual channels (as

described in the previous sections).

8-Mbit/s-Pulse Frame

Fig. 2


5

34-Mbit/s-Pulse Frame

Fig. 3

140-Mbit/s-Pulse-Frame

Fig. 4


6

In addition to the service bits this frame has two data bits, which can be used for

the asynchronous transmission of external data signals with bitrates of up to

approx. 10 kbit/s.

2 Timing Sequence of the Multiplex Process

Fig. 5 Time sequence of the address difference in the elastic store depending on the frame structure

The figure shows an example of the time sequence of the address difference

between write in respect. read out address of the elastic store along several

frames. The rising edges (reduction of the distance between addresses) occur in

the tributary information blocks, i.e. when the store is read out. Every time if no

tributary information is transmitted (with JS, FAS) the read out process is

interrupted (1 clock at JS, 3 clock at FAS) and the difference between the

addresses increases accordingly (1 address at JS, 3 addresses at FAS).


7

Despite of these interruptions of the read out process, the actual read out timing

frequency is higher than the write in frequency.

The necessity of the justification is determined in frame Nr. N (getting below the

difference of 3 bits). Then all justification service bits (JS) of the affected

channel are set to 1 in frame N + 1.

When the justification bit position (JT) in frame N + 1 is reached, the read-out

address is stopped during one clock cycle; the transmitted bit is interpreted as

justification bit and the difference between the addresses increases by one

address.

Example:

Let us look at a multiplex system 2/8 Mbit/s and at the structure of the 8 Mbit

frame. The nominal write in frequency is fE = 2048 kHz. The elastic store is read

out during the tributary information blocks with one fourth of the system clock:

This read out clock is interrupted by the JS, FAS blocks. The actual read out rate

results from the relation of tributary bits per frame to the overall number of bits:

The nominal justification bitrate fj can be calculated from the difference between

fM and fE.


8

3 Transmission of Additional Data Channels with Y-Bits

With the frame of the 8 and 34 Mbit/s hierarchy the N-alarm bits can be used for

transmission of external data channels. The 140 Mbit/s frame has two specially

designed bit positions (so-called Y-bits) for this purpose. The bitrate for one Y-

bit corresponds to the frame clock.

This bitrate cannot be used for the external signal, as for this a synchronization

to the frame clock would be necessary. Therefore, the maximum allowable

bitrate for the external data signal is restricted to approx. one fifth of the Y-

bitrate. One bit of the signal to be transmitted is then sampled several times by

the Y-bits (Oversampling).

The distortion of the transmitted signal results from the relation of the Y-bitrate

to the bitrate of the signal to be transmitted.

Chapter 4: Functional Description of Multiplexer/Demultiplexer

1

Chapter 4: Functional Description of

Multiplexer/Demultiplexer

Aim of study This chapter introduces functional units of the multiplexer & the demultiplexer.

Contents Pages

1 Functional Units of the Multiplexer 2

2 Functional Units of the Demultiplexer 3

3 Supervision and Alarms 5


2

Chapter 4

Functional Description of Multiplexer/Demultiplexer

1 Functional Units of the Multiplexer

The incoming separate signals of inputs F2 are regenerated and the individual

line-code is decoded (D1..D4). Besides, the clock of the input signal is also

regenerated (T1 .. T4). With these clocks the digital signals are cyclically written

into the elastic store (ES) where the plesiochronous bitrates are matched to the

F1 clock (multiplex clock). The ES are read out by a central gap timing GT

which has an instantaneous frequency of one fourth of the multiplex bitrate and

shows extraction gaps in the time intervals for the FAS and the justification

service bits.

The output signal of the ES already contains the justification service bits and

justification bits and has gaps only for the FAS.

In the ensuing parallel-to-serial converter the synchronous output signals of the

ES are put together to form the multiplex signal. In addition, the FAS and the D-

bits respectively N-bits are inserted here into the timing gap of the ES output

signal. These D- and N-bits come from the supervision section of the

multiplexer and thus feeded from the outside.

The signal that has been formed in this way is transformed into the line code and

matched to the F1 interface.


3

Fig. 1 Functional diagram of the multiplexer

Fig. 2 Gap timing for the reading out of the ES (simplified representation)

2 Functional Units of the Demultiplexer

The incoming signal of input F1 is regenerated and decoded. The receiving

clock is recovered and controls all processes on the demux-side.


4

Then the signal is converted from serial to parallel and thus distributed

arbitrarily to the four outgoing lines. The following FAS detector performs two

tasks. First, it determines how the FAS is distributed to the four lines, and thus

allows an allocation of the lines to the channels. Secondly, it supervises the

periodical occurrence of the FAS, and thus allows a synchronization to the

transmitting signal. In the channel switch-over the four separate signals are

controlled by the FAS detector and distributed to the right channels. The

justification service bit evaluation forms the individual gap timings with the aid

of the frame clock and the regenerated receiving signals. For this the clock is

suppressed during the FAS-period and the JS bit positions. In addition, the data

signals D1...D4 are evaluated at the instant of the JS bitpositions. If the

evaluation result indicates that a justification position follows, the clock is

suppressed at the justification bit position for the individual channels

(GT1...GT4).

These gap timings GT1...GT4 ensure that only tributary information bits are

written into the ES. The ES are read out with the smoothed gap timing (PLL).

Before each output there is an encoder, which converts the signal into the

individual line codes.


5

Fig. 3 Functional diagram of the Demultiplexer

3 Supervision and Alarms

Multiplex systems have a section which supervises the system’s state of

operation and initiates alarm reactions in case of interference.

The supervision functions are as divided into three groups:

• Supervision of the incoming signals at the F2in interfaces.

• Supervision of the incoming signals at the F1in interfaces.

• Supervision of device internal functions.


6

Likewise, the alarm reactions can be divided into:

• Optical display of alarm at the system inset.

• Emission of an AIS signal.

• Forwarded of alarms to the local alarm system.

• Release of signaling bits to the remote station.

Fig. 4 shows at which position in the signal path the various criteria are

supervised, respect. Where AIS, D-and N-bits are inserted.

Fig. 4 Supervision functions in a multiplex system (see also next page)


1


Aim of study This chapter introduces interface codes, digital signal regeneration & reasons for bit errors.

Contents Pages

1 Introduction 2

2 Interface Codes 3

3 Digital Signal Regeneration 7

4 Reasons for Bit Errors 11


2

Chapter 5

Baseband Transmission of Digital Signals

1 Introduction

Digital signal devices process the signals as purely binary information, i.e. the

signal level does not change between bits with the same logical state. For this

reason, these so-called NRZ-signals (No return to zero) can only be processed

together with the corresponding clock, which enables the identification of

individual bit positions.

Fig. 1 Processing of NRZ signals with the aid of separate clock

This clock is not separately transmitted and thus it has to be possible to derive

(i.e. regenerate) the clock from the data signal on the receiving side. It is

obvious that for a NRZ code this is very complicated, if not virtually impossible.

A further disadvantage of the NRZ code is that it carries a certain amount of dc-

voltage which excluded the signal’s galvanic isolation at the interface

(transformer etc.). Due to these disadvantages, various interface codes have been

developed; all of which comply with the following requirements:

• Good clock retrieval features.

• No dc-component.


3

2 Interface Codes

A suitable interface code has a maximum of transitions between the different

signal levels, even for the transmission of lengthy sequences of identical logical

states; it has no dc-component. The survey shows the development of individual

codes (fig. 2).

RZ Code A log. 1 is represented as half-bit with a change of signals levels

from Low High Low.

Advantage: Clock retrieval possible also for adjacent log.1 bits.

Disadvantage: No clock information for zero sequences, dc-

component.

AMI Code The state log. 1 is represented alternatively as positive or

negative signal level.

Advantage: Clock retrieval possible also for adjacent log.1

bits, no dc-component.

Disadvantage: No clock information for zero sequences.

HDB 3 Code Is derived form the AMI code? Here, four consequent zero bits

are replaced by a 1001 or 0001 combination. This is done in such

a way that the signal receiver detects the mutilation of

informational contents and cancels it.

Advantage: Maximum clock information, no dc-component.


4

Disadvantage: None this code is applied for the device

interfaces from 2 Mbit/s up to 34 Mbit/s

(baseband transmission). The exact coding rules

are enumerated in the following.

CMI Code Due to its easy generation with delay lines and simple gate

functions the CMI code is suited especially for interfaces with

high bitrates. Therefore, this code is standardized for the 140

Mbit/s device interfaces.

A further important advantage of the interface code is the possibility it offers to

detect transmission errors by supervising the coding rules. With the HDB3 code,

for example, the receiving of four zero bits would represent the violation of a

coding rule, i.e. at least one bit error must have been occurred during

transmission.

The standardization of interface codes only refers to device interfaces. The

codes for conductor-bound transmission paths are manufacturer-dependent and

are generally adapted to the requirements of the respective terming unit.


5

Digital Line Codes

Fig. 2

Fig. 3 Amplitude spectrum of various codes


6

Fig. 3 shows the amplitude spectrum of various interface codes. For codes

without a dc-component the maximum energy is within the range of a frequency

which corresponds to half of the bitrate value. This is obvious when comparing

the definitions of frequency and bitrate respectively.

Fig. 4 Bit sequences 0101....

The bit sequence represented in fig. 4 shall serve as an example. One signal

period covers 2 bits and corresponds to the basic wave of the data signal. This

wave contains the greatest amount of energy and has a frequency which equals

half of the bitrate value. This is also the frequency that is indicated by a

frequency counter connected to a source of a digital signal.

HDB3-Coding rules

(Third-Order-High-Density-Bipolar-Code)

The HDB3-code is a modified version of the AMI-code. Binary signals or AMI-

code signals may contain lengthy “0“ sequences, which hinder the clock

retrieval in the regenerative repeaters along digital transmission paths. The

HDB3 code enables the elimination of “0“ sequences with more than 3 zeros.

1. If there are more than 4 consecutive “0“-signal elements, the fourth “0“-

signal element shall be replaced by a V-signal element (=„1“-signal

element. A V-signal element causes a Violation of the AMI-rule.


7

2. If between the V-signal element, inserted according to the conditions

specified above (rule 1), and the preceding V-signal element there is an

even number of “1“-signal elements, then the first of four “0“-signal

elements shall be replaced by an A-signal element (=“1“-signal element).

The polarity of the A-signal element complies with the AMI-rule. The last

of four “0“-signal elements becomes again a V-signal element (A00V). In

this the A- and V-signal elements have the same polarity.

Fig. 5 Conversion of binary signals into HDB3-signals

3 Digital Signal Regeneration

The digital signal regeneration is one of the advantages of the digital

transmission technique. Theoretically, it enables the signals to be transmitted via

an unlimited distance without any quality loss.


8

During transmission, a digital signal is attenuated and distorted; which results in

a reduction of the signal /noise ratio. The regeneration process has the task of

canceling such distortions and regenerating the originally sent signal from the

actually received signal. That is why every interface on the receiving side is

followed by a regenerator.

Fig. 6 Principle of digital signal regeneration

Four basic function blocks are necessary for the digital signal regeneration:

• Amplification block (balancing of attenuation losses).

• Clock retrieval block.

• Amplitude decision block.

• Time decision block.


9

Fig. 7 Block diagram of a digital signal regenerator

These four functions are represented in next figure.

• The receiving signal is fed into a controlled amplifier (AGC) which keeps

the amplitude of the outgoing signal at a constant value over a wide range

of incoming amplitudes. Thus, the attenuation of the transmission path is

balanced.

• The constant output level is a precondition for the functioning of the

amplitude decision block (AD) which follows. This AD decides on the

basis of an internal threshold value whether the level of incoming signal is

above or below this threshold. Accordingly, a signal with the level Log. 1

or Log.0 is emitted at the output. The output signal thus consists of pulses,

the width of which only depends on the period during which the output

signal exceeds the decision threshold.


10

• The time decision block (TD) has the task of generating signal pulses with

constant width. For this, it requires the regenerated receive signal clock

which samples the output signal of the amplitude decision block. If, at the

time of sampling the signal has a level of Log. 1, the time decision block

emits a pulse with constant width. Thus, incoming pulses of any width are

turned into pulses corresponding exactly to the bit width of the transmitted

signal. The time decision process is the final stage of regeneration.

• The clock retrieval CR block is in charge of regenerating the transmitted

signal clock from the receive signal clock. In order to effect this function,

a phase locked loop (PLL) is employed, basically consisting of a voltage-

controlled oscillator whose frequency can be changed by a control-

voltage.

By adequate evaluation of the receiving signal it is now possible to reach a

control voltage which can set the oscillator to the exact clock frequency value of

the transmitting signal.

The following examples show a regenerator for HDB3 signals, as well as the

signal shape between individual function blocks.


11

Fig. 8 Regeneration of HDB3 signals

4 Reasons for Bit Errors

The decisive quality criterium for the transmission of digital signals is the so-

called bit error rate (BER). This BER represents the proportion of bits which

have been mutilated (i.e. incorrectly recorded) during transmission, to the total

amount of bits transmitted within a certain interval. The BER directly influences

the quality of the transmitted services (e.g. voice channels, data channels, video

signals). Two significant BER are explained exemplary in the following:


12

• BER = 10-6

This BER virtually cannot be perceived in a voice channel. For the

transmission of data channels, however, this value represents the maximum

acceptable limit. The transmission system is in a state of "degraded quality",

which is indicated by a degradation alarm (low priority) on the devices

involved. The transmission path remains, nevertheless, in operation.

• BER = 10-3

This BER causes a strong interference noise in a voice channel. The

operating state is judged to be of "unacceptable quality", which is signaled

by the devices involved by the emission of a failure alarm (high priority).

The transmission path goes out of operation.

How do bit errors arise?

In the previous section it was mentioned that digital signals can be regenerated as

requested, i.e. a transmission without quality reduction is possible. This statement

is, however, only partially true, i.e. whenever the impairment of the transmission

signals is within limits which still permit the regeneration at the receiving side. The

reasons for the formation of bit errors are

• Low signal/noise ration.

• Jitter.

• Intersymbol interference.

Low signal/noise ratio

Noise amplitudes which influence the amplitude decision process are

superimposed to the originally sent signal.


13

The superimposed interference peaks lead to an incorrect signal interpretation at

the receiving end. Reasons for a low S/N-ratio are:

1. Too strong signal attenuation during transmission.

2. External interference during transmission.

For transmission in cable sections (especially optical fiber) both reasons can be

largely eliminated by careful planning.

Fig. 9 Low S/N-ratio

Jitter

Due to jitter, the transitions between signal levels log. 0 and log. 1 do not take

place at periodically recurring points in time (characteristically moments) as for

undisturbed signals, which means that the transitions oscillate around the

characteristically moments.

Jitter is characterized by jitter amplitude (unit intervals UI) and jitter frequency.

One UI means that, because of deviation from the characteristically moments,

the signal edges are within a range equal to the width of 1 bit.


14

The jitter frequency is the number of oscillations around the characteristically

moment per one second. Jitter influences the time decision process in the

regenerator and causes bit errors for high jitter amplitudes and frequency.

Jitter arises in the devices used for signal transmission. (I.e. in regenerators and

demultiplexers = systematical jitter), or on the transmission path due to external

influences (non-systematic jitter).

Fig. 10 Representation of a Unit Interval (UI)

Intersymbol interference

Is caused by a discrepancy between the bandwidth of the transmission path and

the bandwidth required for the digital signal. This leads to a bit extension, so

that there is an overlap of bits which follow each other. Thus, bit errors occur,

the reasons of which can be traced back to the impairment of amplitude decision

process. For conductor-bound transmission of digital signals this effect can be

excluded by adequate planning. For transmission on radio paths this effect is of

fundamental importance as the frequency response of the transmission path can

change due to atmospherically influence.

Part 3

SDH Basics

1 PDH Multiplexing Pages (1-12)

2 Principles and Characteristics of the SDH

Pages (1-16)

3 Basic Elements of STM‐1 Pages (1-7)

4 Mapping Pages (1-50)

5 Pointer Pages (1-16)

6 Overhead Pages (1-26)

7 Monitoring, Maintenance and Control in the SDH

Pages (1-33)


Sub ‐ Sections

SDH Basics


Chapter 1: PDH Multiplexing

1

Chapter 1 PDH Multiplexing

Aim of study This chapter introduces principles of PDH multiplexing and multiplexing / demultiplexing

of PDH signals.

Contents Pages

1 Introduction 2

2 Principles of PDH Multiplexing 2

3 ANSI / CEPT Bit Rates 3

4 Frame Structure of a PDH Signal 7

5 Multiplexing / Demultiplexing of PDH Signals 7

6 Summary 10

7 Exercise 11

8 Solution 12


2

Chapter 1

PDH Multiplexing 1 Introduction

In the early 1970s, digital transmission systems began to appear, utilizing a

method known as Pulse Code Modulation (PCM), first proposed in 1937.

PCM allowed analog waveforms, such as the human voice, to be represented

in binary form, and using this method it was possible to represent a standard 4

kHz analog telephone signal as a 64 kbit/s digital bit stream. Engineers saw

the potential to produce more cost effective transmission systems by

combining several PCM channels and transmitting them down the same

copper twisted pair as had previously been occupied by a single analog signal.

In Europe, and subsequently in many other parts of the world, a standard

TDM scheme was adopted whereby thirty 64 kbit/s channels were combined,

together with two additional channels carrying control information, to produce

a channel with a bit rate of 2.048 Mbit/s.

2 Principles of PDH Multiplexing

PDH signals with a higher transmission rate are obtained by multiplexing

several lower rate signals. The term PDH will be defined in the next few

pages, however, let us consider the following concepts:

Multiplex Operation

Four input signals with the same nominal bit rate are combined to form one

multiplex signal and then relayed to the receive side via one common

transmission path.


3

De-multiplex Operation:

On the receive side, the sum signal is again distributed to the corresponding

outputs.

Fig. 1

3 ANSI / CEPT Bit Rates

As demand for voice telephony increased, and levels of traffic in the network

grew ever higher, it became clear that the standard 2 Mbit/s signal was not

sufficient to cope with the traffic loads occurring in the trunk network. In

order to avoid having to use excessively large numbers of 2 Mbit/s links, it

was decided to create a further level of multiplexing. The standard adopted in

Europe involved the combination of four 2 Mbit/s channels to produce a single

8 Mbit/s channel. This level of multiplexing differed slightly from the

previous in that the incoming signals were combined one bit at a time instead

of one byte at a time i.e. bit interleaving was used as opposed to byte

interleaving. As the need arose, further levels of multiplexing were added to

the standard at 34 Mbit/s, 140 Mbit/s, and 565 Mbit/s to produce a full

hierarchy of bit rates.


4

The multiplexing hierarchy described above appears simple enough in

principle but there are complications. When multiplexing a number of 2

Mbit/s channels they are likely to have been created by different pieces of

equipment, each generating a slightly different bit rate. Thus, before these 2

Mbit/s channels can be bit interleaved they must all be brought up to the same

bit rate (called "adaptation"), adding 'dummy' information bits, or 'justification

bits'. The justification bits are recognize as such when demultiplexing occurs,

and discarded, leaving the original signal. This process is known as

plesiochronous operation, from Greek, meaning "almost synchronous".

The same problems with synchronization, as described above, occur at every

level of the multiplexing hierarchy, so justification bits are added at each

stage.

The use of plesiochronous operation throughout the hierarchy has led to

adoption of the term "Plesiochronous Digital Hierarchy", or PDH.

Another Explanation to help define PDH is:

If two digital signals are Plesiochronous, their transitions occur at “almost” the

same rate, with any variation being constrained within tight limits. These

limits are set down in ITU-T recommendation G.703. For example, if two

networks need to interwork, their clocks may be derived from two different

PRCs. Although these clocks are extremely accurate, there’s a small frequency

difference between one clock and the other. This is known as a Plesiochronous

difference.

It may be useful to explain the term Asynchronous at this stage:


5

In Asynchronous signals, the transitions of the signals don’t necessarily occur

at the same nominal rate. Asynchronous, in this case, means that the

difference between two clocks is much greater than a Plesiochronous

difference.

Standardized Bit Rates in the "Plesiochronous Digital Hierarchy" (PDH)

Traditionally, digital transmission systems and hierarchies have been based on

multiplexing signals which are plesiochronous (running at almost the same

speed).Also, various parts of the world use different hierarchies which lead to

problems of international interworking; for example, between those countries

using 1.544 Mbit/s systems (U.S.A. and Japan) and those using the 2.048

Mbit/s system.

To recover a 64 kbit/s channel from a 140 Mbit/s PDH signal, it’s necessary to

demultiplex the signal all the way down to the 2 Mbit/s level before the

location of the 64 kbit/s channel can be identified. PDH requires “steps” (140-

34, 34-8, 8-2 demultiplex; 2-8, 8-34, 34- 140 multiplex) to drop out or add an

individual speech or data channel. This is due to the bit stuffing used at each

level.

Comparison of the ANSI and CEPT Hierarchies

We will consider only two hierarchies, even though Japan has its own

hierarchy it will not be studied in this course.

• PDH in accordance with ANSI (American National Standards Institute); basic bit rate employed is 1,5 Mbit/s, e.g. USA.

• PDH in accordance with CEPT (Conférence Européene des Administrations des Postes et des Telécommunications) basic bit rate employed is 2 Mbit/s, e.g. Europe.


6

Fig. 2

Fig. 3 Plesiochronous digital hierarchy


7

4 Frame Structure of a PDH Signal

Every signal within a CEPT hierarchy level has a specific frame structure

which basically consists of the following blocks:

Fig. 4 Frame structure of a PDH signal

5 Multiplexing / Demultiplexing of PDH Signals

A multiplex sum signal is generated from the partial signals 1, 2. 3 and 4 (also

termed input, incoming, or sub signals) through the method of bit interleaving

==> bit-by-bit multiplexing.

Fig 5


8

Fig. 6 Here, the insertion of the Frame Alignment Signal (FAS), the justification bits,

etc. into the multisignal is not yet taken into consideration.

The bits of the frame alignment signals (FAS) contained in the input signals I

and II respectively are also inserted bit-by-bit into the multiplexed signal.

Fig. 7


9

Caution!

After the multiplex operation, the two FAS no longer form a joint unit. Beside

performing the bit interleaving, the multiplexer has also the function to create

a new CEPT frame for the multiplexed signal. Within this frame, the tributary

information is represented by the two complete CEPT frames of input signals I

and II.

Fig. 8


10

There is no phase relationship between the FAS of the multiplexed signal and

the individual frame alignment signals of the tributary signals 1 and 2. A new

frame for the multiplexed signal is created. This new frame has its own FAS.

Fig. 9 6 Summary

Principles of PDH Multiplexing:

• Bit rates in accordance with ANSI: 1,5 Mbit/s, 6 Mbit/s and 45 Mbit/s

• Bit rates in accordance with 2 Mbit/s, 8 Mbit/s, 34 Mbit/s and 140 Mbit/s

CEPT:

• Every signal has a separate frame structure.

• Bit-by-bit multiplexing.

• No frame synchronization of the tributary signal inputs.

• The input signals of the tributaries are plesiochronous to each other, i.e. their clock rates have the same nominal value, but there is, however, a slight amount of variation between the two.


11

7 Exercise

1. What are the bit rates of the CEPT Hierarchy?

2. What are the elements of PDH frames and what is their function?

3. How many different FAS do exist in a 140 Mbit frame?


12

8 Solution

1. What are the bit rates of the CEPT Hierarchy?

2 Mbit/s

8 Mbit/s

34 Mbit/s

140 Mbit/s

2. What are the elements of PDH frames and what is their function?

FAS Frame Alignment Signal

D+N bit Service bits

TB Tributary bits

CB Control bits for justification

JB Justification opportunity bit

3. How many different FAS do exist in a 140 Mbit frame?

4

Chapter 2: Principles and Characteristics of the SDH

1

Chapter 2 Principles and Characteristics of the

SDH

Aim of study This chapter introduces introduction to the Synchronous Digital Hierarchy SDH.

Contents Pages

1 Introduction to the Synchronous Digital Hierarchy SDH 2

2 ITU-T and SDH, an Introduction 4

3 ITU-T Recommendations for SDH Bit Rates 6

4 Structure of an STM-1 Frame 7

5 Byte-by-Byte Multiplexing of SDH Signals 8

6 Synchronization of STM-1 Frames 9

7 Line Codes used in SDH 11

8 Summary 14

9 Exercise 15

10 Solution 16


2

Chapter 2

Principles and Characteristics of the SDH 1 Introduction to the Synchronous Digital Hierarchy SDH

Why use SDH instead of PDH?

The problem of flexibility in a plesiochronous network is illustrated by

considering what a network operator may need to do in order to be able to

provide a business customer with a 2 Mbit/s leased line. If a high speed

channel passes near the customer, the operation of providing him with a single

2 Mbit/s line from within that channel would seem straightforward enough. In

practice, however, it is not so simple.

The use of justification bits at each level in the PDH means that identifying

the exact location of the frames in a single 2 Mbit/s line within say a 140

Mbit/s channel is impossible. In order to access a single 2 Mbit/s line the 140

Mbit/s channel must be completely demultiplexed to its 64 constituent 2

Mbit/s lines via 34 and 8 Mbit/s. Once the required 2 Mbit/s line has been

identified and extracted, the channels must then be multiplexed back up to 140

Mbit/s.

Obviously this problem with the "drop and insert" of channels does not make

for very flexible connection patterns or rapid provisioning of services, while

the "multiplexer mountains" required are extremely expensive.


3

Another problem associated with the huge amount of multiplexing equipment

in the network is one of control. On its way through the network, a 2 Mbit/s

leased line may have traveled via a number of possible routes. The only way

to ensure it follows the correct path is to keep careful records of the

interconnection of the equipment. However, as the amount of reconnection

activity in the network increases it becomes more difficult to keep records

current and the possibility of mistakes increases. Such mistakes are likely to

affect not only the connection being established but also to disrupt existing

connections carrying live traffic.

Another limitation of the PDH is its lack of performance monitoring

capability.

Operators are coming under increasing pressure to provide business customers

with improved availability and error performance, and there is insufficient

provision for network management within the PDH frame format for them to

be able to do this.

Hence the development of the SDH technology.

What do we mean by Synchronous Digital Signals in an SDH network?

Having just described why we may want to use SDH in preference to PDH, let

us now define a PDH and SDH network in simple terms.

When data signals with the same nominal bit rate (which could have different

sources, as in plesiochronous signals), are controlled by a central clock

frequency (the master clock), the signals are termed synchronous, (i.e. as in a

synchronous network). Thus we can say:


4

In a plesiochronous network, the individual link sections are not synchronous

to each other.

In the synchronous network, on the other hand, the link sections are

synchronous to each other.

Now let us focus on a system that uses synchronous digital signals, i.e. SDH.

2 ITU-T and SDH, an Introduction

Background

Before SDH, the first generations of fiber-optic systems in the public

telephone network used proprietary architectures, equipment line codes,

multiplexing formats, and maintenance procedures. The users of this

equipment wanted standards so they could mix and match equipment from

different suppliers.

The task of creating such a standard was taken up in 1984 by the Exchange

Carriers Standards Association (ECSA) in the U.S. to establish a standard for

connecting one fiber system to another. In the late stages of the development,

the CCITT became involved so that a single international standard might be

developed for fiber interconnects between telephone networks of different

countries. The resulting international standard is known as Synchronous

Digital Hierarchy (SDH).

SDH Advantages

The primary reason for the creation of SDH was to provide a long-term

solution for an optical mid-span meet between operators; that is, to allow

equipment from different vendors to communicate with each other.


5

This ability is referred to as multivendor interworking and allows one SDH-

compatible network element to communicate with another, and to replace

several network elements, which may have previously existed solely for

interface purposes. SDH (Synchronous Digital Hierarchy) is a standard for

telecommunications transport formulated by the International

Telecommunication Union (ITU), previously called the International

Telegraph and Telephone Consultative Committee (CCITT).

SDH was first introduced into the telecommunications network in 1992 and

has been deployed at rapid rates since then. It’s deployed at all levels of the

network infrastructure, including the access network and the long-distance

trunk network. It’s based on overlaying a synchronous multiplexed signal onto

a light stream transmitted over fiber-optic cable. SDH is also defined for use

on radio relay links, satellite links, and at electrical interfaces between

equipment.

The comprehensive SDH standard is expected to provide the transport

infrastructure for worldwide telecommunications for at least the next two or

three decades. The increased configuration flexibility and bandwidth

availability of SDH provides significant advantages over the older

telecommunications system. These advantages include:

• A reduction in the amount of equipment and an increase in network reliability.

• The provision of overhead and payload bytes; the overhead bytes permitting management of the payload bytes on an individual basis and facilitating centralized fault sectionalization.

• The definition of a synchronous multiplexing format for carrying lower level digital signals (such as 2 Mbit/s, 34 Mbit/s, 140 Mbit/s) which greatly simplifies the interface to digital switches, digital cross-connects, and add-drop multiplexers.


6

• The availability of a set of generic standards, which enable multi vendor interoperability.

• The definition of a flexible architecture capable of accommodating future applications, with a variety of transmission rates.

3 ITU-T Recommendations for SDH Bit Rates

The ITU-T (International Telecommunication Union -Telecommunication

sector) specified a base signal, the STM-1 (Synchronous Transport Module-1)

with 155,520 Mbit/s.

All multiplex levels in the SDH are positive integer multiples of this base

signal "STM-1".

In this way, a world-wide uniform concept for the transmission of 155 Mbit/s

data signals was provided, which means that all previous PDH signals (CEPT

/ ANSI ) must be interleaved to the SDH base signal by a procedure called

"MAPPING".

Fig. 1


7

4 Structure of an STM-1 Frame

The two-dimensional representation of an STM-1 frame includes 9 rows with

270 bytes each.

The sequence of transmission is: top left to bottom right.

Fig. 2

The STM-1 frame consists of three blocks:

• Pointer (PTR): indicates the start address of the tributary information.

• Section OverHead (SOH): additional transmission capacity.

• Payload: tributary information.

The frames are transmitted in intervals of 125 µs.

The STM-1 frame is repeated (1s: 125 µs) = 8000 times per second.

Thus, every byte in an STM-1 frame has a transmission capacity of 64 kbit/s.


8

Fig. 3

5 Byte-by-Byte Multiplexing of SDH Signals

Multiplex Technique (Transmitter)

Contrary to the PDH, the SDH uses the method of BYTE INTERLEAVING

to generate the multiplex sum signal * out of the sub-signals I and II

byte-by-byte multiplexing.

The multiplex signal STM-4 has the same frame duration as the STM-1, i.e.

125µs.

NOTE!

The explanation given in the example has been simplified. SDH equipment

adds SOH at different stages of the multiplex process. There is no need to go

into too much detail at this stage. This is sufficient to give an understanding of

the principles.


9

For a better understanding, the generation of an STM-4 frame was explained

here with only 2 STM-1 frames, although in practice 4 STM-1 frames are

multiplexed.

Fig. 4

Fig. 5 Frame 2 x STM-1 ----> 1 x STM-2

6 Synchronization of STM-1 Frames

Even in a synchronous network, the frames STM-1 # 1 and STM-1 # 2(3, 4)

are usually delayed in time (e.g. due to different run times).


10

Fig. 6

Prior to multiplexing, the subsignals STM-1 # 1 and STM-1 # 2(3, 4) are

synchronized to each other that means frame alignment signal and pointer are

synchronized but the tributary information is neither modified nor delayed.

The modification of the pointer during synchronization is called "pointer

adjustment operation".

Fig. 7


11

7 Line Codes used in SDH

7.1 Codes and Interfaces of SDH

The optical line code for all STM-N signals is a scrambled

"NON - RETURN -TO -ZERO (NRZ)" -Code!

By scrambling the NRZ code it is ensured that when sending an STM signal

on the line, the signal includes sufficient clock edges to allow timing recovery

on the receiver side. The transmission of long "0" or "1" bit sequences must

therefore be avoided.

STM-1, STM-N Scrambler

Fig. 8 STM-N scrambling

When sending an STM signal on the line it must be ensured that the signal

includes sufficient clock edges to allow timing recovery on the receiver side.

The transmission of long „0“ or „1“ bit sequences must therefore be avoided.


12

For transmission on coaxial lines the established practice for electric signals is

to select a line code which suffices to enable clock recovery on the receiver

side.

Both STM-1 and STM-N are provided for transmission on optical fiber routes.

Scrambling of the electrical signal is sufficient for transmission of the optical

signal. Long „0“ or „1“ bit sequences are avoided and no elaborate line code is

required. In addition to the optical interface, a CMI-coded electrical interface

is defined (G.703) for the STM-1 signal.

General Scrambler Function

The scrambler is a transmit-side device, which converts an existing digital

signal to a different signal with a pseudo-random bit sequence without altering

the bit rate. A descrambler on the receive side then reconstructs the original bit

sequence. The scrambler/descrambler is technically implemented through a

shift register whose output is logically linked with the input.

Application to STM-N

The STM signal of the synchronous hierarchy is scrambled only prior to its

optical conversion for transmission on optical fiber. Accordingly, an STM-1

or STM-N signal is not scrambled if it is first being encoded to a higher-level

multiplex signal. Only a multiplex signal, which is converted, to an optical

transmission signal is subjected to the scramble procedure.

The scramble procedure is applied to all bytes in the respective STM-1 or

STM-N frame apart from the first row in the overhead section (Fig. 8). The

first row (N x 9 bytes) includes the frame alignment signal amongst others.


13

As this frame alignment signal is not scrambled, the synchronization to this

FAS is possible without previous descrambling.

7.1.1 STM-1, Electrical Interface

The following values in conformity with G.703 are apply to electrical

interface.

Bit rate: 155.52 Mbit/s Code: CMI (coded mark inversion code) Level: 1,0 VSS + 0,1 V

The CMI code is a binary transmission code. The binary values „1“ are

alternately represented by a positive and negative status and the binary values

„0“ are always represented by a negative status in the first half and a positive

status in the second half of the binary interval.

Fig. 9 CMI code for electrical interface

7.1.2 STM-1 and STM-N, Optical Interface

Bit rate: N x 155.52 Mbit/s scrambled

Code: NRZ


14

8 Summary

Principles and Characteristics of the SDH:

• Bit rates exceeding 140 Mbit/s are standardized on a worldwide basis.

• Both synchronous and plesiochronous operation is possible.

• All current PDH signals (CEPT/ANSI) can be transmitted within the SDH (except 8 Mbit/s).

• The "Section OverHead" bytes provide a high transmission capacity for monitoring, maintenance and control tasks.

• High-level multiplex signals are integer multiples (=N) of the basic bit rate (155,520 Mbit/s).

• For the first time, the optical line code is standardized worldwide.


15

9 Exercise

1. What are the bit rates in the SDH?

2. How many bytes are transmitted in a STM-1 signal?

3. What are the three main blocks of the STM-1 signal?

4. List the line codes used for SDH optical and electrical line signals?

5. What is the main function of scrambler?

6. Which part of the STM-N is not scrambled?


16

10 Solution

1. What are the bit rates in the SDH?

155.52 Mbit/s

622.08 Mbit/s

2.5 Gbit/s, 9953,228 Mbit/s, and 39813,12 Mbit/s

Nx155.52 Mbit/s

2. How many bytes are transmitted in a STM-1 signal?

2430

3. What are the three main blocks of the STM-1 signal?

SOH Pointer Payload

4. List the line codes used for SDH optical and electrical line signals?

CMI (electrical)

NRZ (optical)

5. What is the main function of scrambler?

Generation of a bit sequence with balanced number of 0 and 1 to ensure clock

recovery on the receiving end.

6. Which part of the STM-N is not scrambled?

The first 9 x n bytes of the STM-N signal.

Chapter 3: Basic Elements of STM‐1

1

Chapter 3 Basic Elements of STM-1

Aim of study This chapter introduces elements of an STM-1 signal.

Contents Pages

1 Elements of an STM-1 Signal 2


2

Chapter 3

Basic Elements of STM-1 1 Elements of an STM-1 Signal

1.1 Terminologies

Before discussing the basic elements of an STM frame, we will look at the

terminology used.

The suffixes used throughout the SDH multiplex levels derive from the older

PDH multiplex orders.

For instance

PDH Multiplex Order SDH suffix used

First Order of Multiplexing (2 Mbit/s) XX 1x (e.g. VC 12)

Second Order of Multiplexing (6 Mbit/s) XX 2 (e.g. TU 2)

Third Order of Multiplexing (34 Mbit/s) XX 3 (e.g. TUG 3)

Fourth Order of Multiplexing (140 Mbit/s) XX 4 (e.g. VC 4)

The above terms will be described in detail later in this section. Note,

however, that the First Order of multiplexing has two sub divisions, one for 2

Mbit/s PDH signals, e.g. VC 12, and one for 1.5 Mbit/s PDH signals e.g. VC

11. The other Orders of multiplexing have only one designation.


3

1.2 Container C

Prior to its transmission in the STM-1 frame, every piece of tributary

information, whether plesiochronous or synchronous, is interleaved in

containers (fig. 1).

The term container C describes a defined network-synchronized transmission

capacity. The container size is given in bytes. This byte total is provided every

125 µs as container transmission capacity. The defined container sizes are

tailored to the current plesiochronous signals.

The following containers are distinguished:

Designation Signal to be transmitted

C-11 1 544 kbit/s

C-12 2 048 kbit/s

C-2 6 312 kbit/s

C-3 44 736 kbit/s

or 34 368 kbit/s

C-4 139 264 kbit/s The tributary information must be fitted into these containers. This is done

with bit-by-bit and byte-by-byte justification for plesiochronous signals, by

means of purely positive justification as well as negative/zero/justification.

The container includes:

1. Pure tributary information (e.g. PDH signal).


4

2. Fixed justification bytes and bits (fixed stuffing) for approximate timing

alignment. These bytes (or bits) are always without information content and

are used to approximately match the bit rate of the PDH signal to the

basically higher container bit rate. The precise bit rate alignment which

follows are performed with single justification opportunity bits.

3. Justification opportunity bits for precise timing alignment. These bits can be

used as tributary bits or justification bits as required.

4. Justification control bits to notify the receiver whether the justification

opportunity bits is an information bit or a justification bit.

1.3 Virtual Container VC

A path overhead (POH) is added to each container C. Together with its

associated POH, the container C is designated a virtual container VC and

routed as a non-modified entity via a through-connected path in the network.

The POH carries supplementary information ensuring the reliable transport of

the container from signal source to destination. It is added at the start of the

path when the VC is set up and first interpreted, and at the end of the path

when the container is cleared down. The POH includes information on the

supervision and maintenance of a path switched in the network.

Depending on its size, one virtual container can either be transmitted alone in

the STM-1 frame or otherwise interleaved in a larger VC, which is directly

transported in the STM-1.

A distinction is made between higher-order virtual containers (HO) and lower-

order containers (LO).


5

All containers transmitted in one „larger“container are termed LO containers.

LO VC describes VC-11, VC-12, and VC-2. VC-3 is described as a LO VC if

transmitted in a VC-4.

Those containers transmitted directly in the STM-1 frame are termed HO

containers. VC-4 is a HO VC. The same designation applies to a directly

transmitted VC-3.

1.4 Administrative Unit AU

The higher-order virtual containers VC-4 and VC-3 are transmitted directly in

the STM-1 frame.

In this case the pointers (AU-PTR block) embedded in the STM-1 frame

record the phase relationship between the frame and the respective virtual

container. That component of the STM-1 frame within which the VC is able to

„float“ is termed administrative unit (AU). The corresponding pointer,

described as AU pointer, likewise counts as part of the AU. Three 3-byte AU

pointers are included in the first 9 bytes of the 4th row of the STM-1 frame. A

distinction is made between the AU-4 and AU-3.

It is possible to transmit the following AU in the STM-1 frame:

either 1 x AU-4 or

3 x AU-3. ( This has not been implemented by ETSI)

Transmission of the VC-3 is possible either

1. Directly (AU-3) in the STM-1 (not the ETSI recommendation).

2. Indirectly via an AU-4, where 3 x VC-3 are interleaved in one VC-4 (This is the ETSI recommendation, and it is how our equipment maps the VC signal).


6

1.5 Administrative Unit Group AUG

Several AU are byte-interleaved, i.e. multiplexed byte-by-byte, to one AU

group (AUG). The AUG is a frame-synchronized structure corresponding to

the STM-1 without SOH. If the STM-1 SOH is added to the AUG, an STM-1

is produced.

An AUG can be composed either of 1 x AU-4 or 3 x AU-3.

1.6 Tributary Unit TU

With the exception of the VC-4, all VC can be interleaved in a larger VC and

transported in the STM-1. The „smaller“VC can generally float in phase terms

inside the „larger“(higher-order) VC. For this purpose a pointer establishing

the phase relationship between the two VC must be positioned at a fixed

location in the higher order VC. Tributary unit TU is the term used to describe

the component of the higher order container inside which the embedded LO

VC can vary plus the corresponding pointer (TU pointer).

The following TU are distinguished: TU-11, TU-12, TU-2, TU-3.

1.7 Tributary Unit Group TUG

Before being interleaved in the higher-order container, the TU are combined

in one group, i.e. byte-interleaved. Such a group is termed a TUG (tributary

unit group).

The following TUG has been defined: TUG-2 and TUG-3.


7

Fig. 1 Container sizes and bit rates

Fig. 2 Synchronous digital hierarchy

Chapter 4: Mapping

1

Chapter 4 Mapping

Aim of study This chapter introduces mapping of 140 Mbit/s signal, 34 Mbit/s signal, 2 Mbit/s signal &

ATM cells.

Contents Pages

1 Mapping of a 140 Mbit/s Signal into the STM-1 2

2 Mapping of a 34 Mbit/s Signal to the Container C-3 9

3 Mapping of a 2 Mbit/s Signal to STM-1 17

4 Mapping of ATM Cells into the STM-1 30

5 Concatenation of Payloads 41

6 Summary 46

7 Exercise 49

8 Solution 50

Chapter 4: Mapping

2

Chapter 4

Mapping 1 Mapping of a 140 Mbit/s Signal into the STM-1

Fig. 1

1.1 "Mapping" of a 140 Mbit/s Signal to the Container C-4

Prior to its transmission in the STM-1 frame, the 140 Mbit/s PDH signal is

interleaved into a container C-4. The position of the signal bits in the container

is exactly defined. The term "mapping" describes this fixed bit arrangement.

The size of the container C-4 amounts to 2340 byte. For a better

understanding, a two-dimensional representation of the container is shown

below (9 x 260):

Chapter 4: Mapping

3

A C-4 is provided as network-synchronous transmission capacity every 125

µs.

A comparison of the number of possible, usable bits per container C-4

260 byte x 9 = 2340 byte x 8 = 18720 bit

Fig. 2

And the number of bits (nominal bit rate: 139,264 Mbit) actually to be

transmitted per container

139,264 Mbit/s: 8000 Hz = 17408 bit,

Reveals an over-capacity of the C-4.

Chapter 4: Mapping

4

Fig. 3

Beside the pure tributary information bits (140 Mbit/s) the following bits are

transmitted in the container C-4:

• Fixed justification bits and bytes (approximate clock alignment.

• Justification opportunity bits (positive justification for precise clock alignment).

• Justification control bits (justification information bits).

• Overhead bits (no function specified) 140 Mbit/s in C-4.

The 140-Mbit/s

Plesiochronous signal is aligned to the C-4 container bit rate through bit-by-bit

positive justification. 1 justification opportunity bit and 5 justification control

bits are provided per container row. The exact mapping of these bits in the

container is shown in fig. 5.

Chapter 4: Mapping

5

Fig. 4 Plesiochronous 139,264-kbit/s signal in VC-4 VC-4 block structure

Fig. 5 Plesiochronous 139,264 k-bit/s signal in VC-4 1 VC-4 row

Chapter 4: Mapping

6

The C-4 container has a total transmission capacity of 260 x 9 x 8 bits/125 s.

A capacity of 2080 bits is available per container row.

The 140-Mbit/s signal has a nominal bit rate of 139.264 Mbit/s, corresponding

to 17408 bit/125 µs. This, results in 1934.222 bits per signal container row.

The C-4 container provides 1934 I-bits and 1 stuffable bit per row for

transmission of this useful information. Each row further contains 5 stuff

check bits as well as overhead and fixed stuff bits and bytes respectively.

1.2 Interleaving of the C-4 into the STM-1

In order to transmit the container C-4 in the STM-1, container-specific

supplementing must be effected:

1. Addition of the Path OverHead (POH)

The VC-4 includes a "Path OverHead" (POH) with a size of 9 byte.

Fig. 6

Chapter 4: Mapping

7

Additional Information about Path:

The route which a container and its overhead take through the SDH network is

also called "path".

The path is defined by the operator. At the beginning of the path, every

container is assigned a trace, which can be checked at the end of the path.

The block resulting from the container C-4 and the POH is called

Virtual Container 4 = VC-4.

Fig. 7

2. Addition of the Pointer (PTR)

There is a floating embedding of the Virtual Container VC-4 into the STM-1

frame of the payload. Part of the Virtual Container VC-4 is transmitted in one

STM-1 frame, and another part in the next frame.

Chapter 4: Mapping

8

Fig. 8

The Pointer (PTR) indicates the start of the Virtual Container (VC-4) in the

payload.

That component of the STM-1, inside which the

VC-4 is able to "float" and which is made up of

PTR and payload, is designated

Administrated Unit 4 = AU-4

The AU-4 Pointer is abbreviated AU-4 PTR.

3. Addition of the Section OverHead (SOH)

In order to complete the STM-1 frame, the Section OverHead (SOH) is

added to the AU-4.

Chapter 4: Mapping

9

2 Mapping of a 34 Mbit/s Signal to the Container C-3

2.1 3 x 34 Mbit/s -> STM-1

Fig. 9 Prior to its transmission in the STM-1 frame, the 34 Mbit/s PDH signal is

interleaved into a container C-3 (=Mapping).

The size of the container C-3 amounts to 756 byte. For a better understanding,

a two-dimensional representation of the container is shown below (9 x 84):

A C-3 is provided as network-synchronous transmission capacity every 125

μs.


9 byte x 84 =756 byte x 8 = 6048 bit

And the number of bits (nominal bit rate: 34,368 Mbit/s) actually to be

transmitted per container

Chapter 4: Mapping

10

34,368 Mbit/s: 8000 Hz = 4296 bit

Reveals an over-capacity of the container C-3!

Fig. 10

The reason for the over-capacity is a recommendation by ITU-T specifying

that the transmission of a 44, 736 Mbit/s signal (ANSI) must also be carried

out in the container C-3.

= 44, 736 Mbit/s: 8000 Hz = 5593 bit.

Fig. 11

Chapter 4: Mapping

11

When considering the number of payload bits per STM-1 frame

9 byte x 261 x 8 = 18720 bit,

it emerges that only three C-3 (3 x 6048 bit) at maximum can be transmitted

per STM-1 frame => this means only 3 x 34 Mbit/s instead of the 4 x 34

Mbit/s which can be transmitted in a 140 Mbit/s PDH signal.

Beside the pure tributary information bits (34 Mbit/s)! the following bits are

transmitted in the container C-3:

• Fixed justification bits and bytes (approximate clock alignment).

• Justification opportunity bits (positive/negative justification for precise clock alignment).

• Justification control bits (indicate whether there is a positive, negative, or no justification).

• Overhead bits (no function specified).

34 Mbit/s in C-3

The positive/zero/negative justification method is used for transmission of the

34 Mbit/s plesiochronous signal in the C-3 container. For this purpose 2

justification opportunity bits within 3 container rows are provided (fig. 13).

Chapter 4: Mapping

12

Fig. 12 Plesiochronous 34,368 kbit/s signal in VC-3 Block structure

Fig. 13 Plesiochronous 34,368-kbit/s signal in VC-3 3 rows of the VC-3

Three C-3 container rows at a time provide 2016 bits for transmission. These

bits comprise 1431 I-bits, 2 justification opportunity bits, 2 x 5 justification

control bits as well as overhead and fixed stuff bits. The 34 Mbit/s signal has a

nominal bit rate of 34,368 Mbit/s.

Chapter 4: Mapping

13

1432 bits must thus be transmitted per 3 C-3 container rows. The 1431 I-bits

in the container are used up by the incoming signal at the nominal bit rate.

One justification opportunity bit must permanently be used as an I-bit. One

justification opportunity bit is transmitted as a justification bit (without

information).

However, if the bit rate of the incoming signal is below the nominal value, the

second s-bit (an I-bit in the nominal case) must also be stuffed if necessary

(positive justification).

If the bit rate of the incoming signal exceeds the nominal value, the first S-bit

(a justification bit in the nominal case) is used as an I-bit if required (negative

justification).

2.2 Interleaving of Three C-3 into the VC-4

The transmission of three C-3 in the STM-1 requires some container-specific

supplementing to be effected for every C-3.

Fig. 14

Chapter 4: Mapping

14

2.3 Creation of the Tributary Unit 3 (TU3)

Every C-3 receives a "Path OverHead" (POH) with a size of 9 byte. The block

resulting from the C-3 and POH is termed Virtual Container-3 = VC-3.

Every Virtual Container VC-3 (=LOWER ORDER VC) is assigned a 3-byte

Pointer PTR, which allows the VC-3 to float. The area in which the VC-3 can

float with the aid of the Pointer PTR is called Tributary Unit 3 = TU-3. The

3-byte pointer in the TU-3 is called TU-3 Pointer. The PTR contains an

address which indicates the start of the VC-3 in the TU-3.

Fig. 15 2.4 Creation of the Tributary Unit Group 3 (TUG-3)

A Tributary Unit TU-3 is always supplemented with six fixed justification

bytes which do not contain any information. The block resulting from the

TU-3 and the fixed justification bytes is called Tributary Unit Group 3 =

TUG-3.

Chapter 4: Mapping

15

Fig. 16

2.5 Interleaving of TUG-3 into VC-4

The three resulting TUG-3 (#1, #2 and #3) are byte-interleaved into a Virtual

Container VC-4 (=HIGHER ORDER VC ).

To adjust the three byte-interleaved TUG-3 to the VC-4 it is necessary to add

two columns of fixed justification bytes.

Fig. 17

Chapter 4: Mapping

16

2.6 Interleaving of the VC-4 into the STM-1

The Virtual Container VC-4 is transmitted directly in the STM-1 frame

(Payload).

In this case, the pointer (PTR) embedded in the STM-1 frame contains an

address indicating the beginning of the VC-4 in the payload.

That component of the STM-1, inside which the VC-4 can "float" and which

comprises the two blocks PTR and Payload is designated.

Administrative Unit 4 = AU-4.

In the AU-4, the pointer is abbreviated AU-4 PTR.

Fig. 18

To supplement the STM-1 frame, the Section OverHead (SOH) is added to the

AU-4.

Chapter 4: Mapping

17

Fig. 19

3 Mapping of a 2 Mbit/s Signal to STM-1

Fig. 20

Chapter 4: Mapping

18

3.1 "Mapping" of a 2 Mbit/s Signal to the Container C-12

Prior to its transmission in the STM-1 frame, the 2 Mbit/s PDH signal is

interleaved into a container C-12 (=Mapping).

The size of the container C-12 amounts to 34 byte. For a better understanding,

you can find a two-dimensional representation of the container below:

Fig. 21


= 34 byte x 8 = 272 bit

And the number of bits (nominal bit rate: 2,048 Mbit/s) actually to be

transported per container

2,048 Mbit/s: 8000 Hz = 256 bit,

Reveals an over-capacity of the container C-12.

Beside the pure tributary information bits (2 Mbit/s),

Chapter 4: Mapping

19

The following bits are transmitted in the container C-12:

• Fixed justification bits and bytes (approximate clock alignment).

• Justification opportunity bits (positive/negative justification for precise clock alignment).

• Justification control bits (indicates whether there is a positive, negative or no justification).

• Overhead bits (no function specified).

Fig. 22 3.2 Creating a VC-12 Frame

In order to transmit 63 containers C-12 (with an own 2 Mbit/s signal in each

case) in the STM-1 frame, container-specific supplementing is necessary for

every C-12.

A "Path OverHead" (POH) with the size of 1 byte is added to every C-12.

The function of these bytes will be explained in chapter 6.

Chapter 4: Mapping

20

A VC-12 is provided as network-synchronous transmission capacity every 125 µs.

Fig. 23

3.3 Creating the Tributary Unit TU-12

Fig. 24

Chapter 4: Mapping

21

3.4 Creation of the Tributary Unit Group TUG-2

Three TU-12 (= 3 x 2 Mbit/s signals) from different multiframe TU-12 are

multiplexed byte-by-byte to form a "Tributary Unit Group-2" (TUG-2).

For a better understanding, again a two-dimensional representation of a TU-12

partial fame will be shown.

Fig. 25

Chapter 4: Mapping

22

3.5 Creation of a Tributary Unit Group TUG-3

In a next step, seven TUG-2 (=21 x 2 Mbit/s signals) are combined to form a

TUG-3, i.e. byte-interleaved.

Fig. 26

Chapter 4: Mapping

23

3.6 Interleaving of TUG-3 into a Virtual Container VC-4

The three resulting TUG-3 (#1 #2 and #3) are byte-interleaved into a Virtual

Container VC-4 (=HIGHER ORDER VC).

Fig. 27

3.7 Interleaving of the VC-4 into the STM-1

The Virtual Container VC-4 is transmitted directly in the STM-1 frame

(Payload).

In this case, the pointer (PTR) embedded in the STM-1 frame contains an

address indicating the beginning of the VC-4 in the payload.

Chapter 4: Mapping

24

Fig. 28

That component of the STM-1, inside which the VC-4 can "float" and which

comprises the two blocks PTR and Payload is designated.

Administrative Unit 4 = AU-4.

In the AU-4, the pointer is abbreviated AU-4 PTR.

To supplement the STM-1 frame, the Section OverHead (SOH) is added to the

AU-4.

Fig. 29

Chapter 4: Mapping

25

3.8 Creation of a VC-12 Multiframe

In order to transmit 63 containers C-12 (with an own 2 Mbit/s signal in each

case) in the STM-1 frame, container-specific supplementing is necessary for

every C-12.

A "Path OverHead" (POH) with the size of 1 byte is added to every C-12.

The function of these bytes will be explained in chapter 6.

There can be four different POH bytes for one C-12:

Caution: A multiframe VC-12 is transmitted via four or five STM-1 frames!

Fig. 30

Chapter 4: Mapping

26

Fig. 31 Plesiochronous 2Mbit/s mapping

When a byte-synchronous 2Mbit/s signal is transmitted, the individual 64-

kbit/s channels occupy exactly one byte for each channel in the STM 1 frame.

Hence after the interpretation of the individual pointer levels, it is possible to

access a 64-kbit/s directly, irrespective of the transmission mode. However,

two pointers must be evaluated the AU and the TU pointers.

Chapter 4: Mapping

27

Fig. 32

3.9 Numbering of TU-12s in a VC 4

Each TUG-2 can comprise three TU-12s which shall be numbered #1 to #3(#K).

Thus any TU-12 can be allocated a two-figure address in the form #L, #M,

where L designates the TUG-2 number (1 to 7) and M designates the TU-12

number (1 to 3).

Thus TU-12 #1 (1, 1) resides in columns 10, 73, 136 and 199 of the VC-4, and

TU-12 #2(7, 3) resides in columns 71, 134, 197 and 260 of the VC-4. A full

listing of the location of the TU-12 columns with the VC-4 frame follows.

NOTE – The Time Slot number contained in the diagrams below should

not be interpreted as the tributary port number, as the time slots and port

numbers are independent from each other. It is only during configuration

of the equipment that port is assigned a time slot.

Chapter 4: Mapping

28

An external tributary signal may be assigned to a particular payload capacity

using a connection function.

For example at the VC-12 level,

– Tributary #1 – TU-12 (1, 1, 1)

– Tributary #2 – TU-12 (1, 1, 2)

– Tributary #3 – TU-12 (1, 1, 3)

– Tributary #4 – TU-12 (1, 2, 1)

•

•

•

– Tributary #63 – TU-12 (3, 7, 3)

Fig. 33

Chapter 4: Mapping

29

Fig. 34 3.10 Creating Tributary Units

That component, inside which the multiframe VC-12 can "float" with the aid

of a pointer, is termed multiframe TU-12. The four pointer bytes also count

as part of the multiframe TU-12. Every 125 µs one pointer byte is transmitted,

i.e. the transmission of the complete pointer takes 500 µs.

Three TU-12 (= 3 x 2 Mbit/s signals) from different multiframe TU-12 are

multiplexed byte-by-byte to form a "Tributary Unit Group-2" (TUG-2).

Chapter 4: Mapping

30

In a next step, seven TUG-2 (=21 x 2 Mbit/s signals) are combined to form a

TUG-3, i.e. byte-interleaved.

The three resulting TUG-3 (#1 #2 and #3) are byte-interleaved into a Virtual

Container VC-4 (=HIGHER ORDER VC ) and so on (see 3.6).

Fig. 35 4 Mapping of ATM Cells into the STM-1

ATM Cells -> STM-1

Fig. 36

Chapter 4: Mapping

31

Basic Structure and Contents of an ATM Cell

To account for the rapidly increasing need for broadband services and

applications (e.g. video conferences, multimedia etc.), the pieces of

information are no longer transported and switched through via channels with

a defined structure, but in the form of short packets with a constant length (=

Asynchronous Transfer Mode - cells).

Fig. 37

The tributary information, which normally comes in continuously:

• constant bit rates 64 kbit/s

2 Mbit/s 34 Mbit/s

• data packets • variable bit rates

Is written into the PAYLOAD bit by bit and supplemented by the HEADER.

Chapter 4: Mapping

32

Fig. 38

4.1 ATM Characteristics

Until now „synchronous time-division multiplex methods“ (cf. narrowband

ISDN) or packet-oriented multiplex methods (e.g. in conformity with X.25)

were used to assign band width to the connections within the framework of the

existing transmission capacity.

Although the time-division multiplex method, also known as „asynchronous

transfer mode“(ATM), used in modern telephone networks is ideally suited for

signals with constant bit rates (e.g. PCM-coded speech), but not suited at all if

a variable band width is required.

On the other hand, the conventional handshaking packet-oriented methods

(X.25) are flexible in terms of the throughput per connection, but are

unsuitable for the communication of constant bit-rate signals because of the

propagation times (e.g. speech or video signals); furthermore, the conventional

protocols are designed only for bit rates up to approx. 2 Mbit/s.

Chapter 4: Mapping

33

Therefore, a new approach was made with B-ISDN known as the

„asynchronous transfer mode (ATM) “. This mode is a packet-oriented, non-

handshaking multiplex mode. The ATM principle is bit-rate-independent and

may basically be employed for any digital transmission path which is

sufficiently free of errors.

The ATM method is so simple that the user packets can be conveyed purely

by hardware (table-controlled) after a software-controlled call setup and not

by slow software as in current packet networks. This makes ATM as efficient

as the STM method and far superior to conventional packet-oriented methods

(e.g. X.25).

Current international standards envisage the transport bit rates 155 and 622

Mbit/s for ATM. Due to these high rates and the hardware-controlled

switching, the delay times (otherwise typical for packet methods) are

significantly reduced. Thus ATM seems suitable for all information types:

both for fixed and variable bit rate signals and for packet-oriented signals.

However the flexibility of ATM is balanced by increased complexity. Thanks

to advances in modern microtechnology, however, system costs do not

increase in line with module complexity (gateway total). Otherwise, ATM

would stand no chance when competing with the more inflexible and hence

less complex STM.

Chapter 4: Mapping

34

Fig. 39

With ATM, fixed-length packets known as „cells“are continuously transmitted

in every transmission section.

These cells consist of 48 octets for payload and a 5-octet cell head. If no

payload is to be transmitted, specially labeled blank cells are sent.

ATM allows connections with any net bit rate. The latter is very low if almost

no information cells are being sent, but approaches the transport bit rate

(approx. 130 Mbit/s for 155-Mbit/s transport bit rate) if information cells are

being sent almost exclusively. By a label in the cell head each cell is assigned

to a specific virtual transmission path (short form: path) and to a (virtual)

channel routed in this path. This principle allows the capacity of the

transmission sections in the entire ATM network to be flexibly allocated to

narrowband and broadband connections in any desired combination.

Chapter 4: Mapping

35

The ATM network functions trunk-oriented, i.e. it retains the cell sequence for

every connection. When the connection is set up the network user notifies the

network of the desired bit rate via a (virtual) signaling channel; the network

subsequently reserves the appropriate band width on all transmission paths.

Should a user exceed the agreed bit rate on a connection, the network detects

this at the network input and takes precautions against possible overload, e.g.

by ignoring excess cells.

With ATM, a very simple protocol is used. The protocol works without

acknowledgements, flow control or error correction. This results in a rapid,

service-independent basis switching service. Further performance features can

be added, if necessary, in more advanced protocol levels. Section-by-section

error correction can be omitted due to the high quality of the digital and

optical transmission techniques employed in the network.

Correspondingly, the structure of the cell head is simple fig. 1 shows the cell

structure defined by CCITT for the user-network interface. The essential

elements in the cell head are the "virtual path identifier" (VPI; 8 bits) and the

"virtual channel identifier" (VCI; 16 bits). Together VIP and VCI provide for

the unambiguous assignment of a cell to a virtual connection on a section-by-

section basis.

The field "payload type" (PLT: 3 bits) is used to differentiate useful cells and

blank cells. The field "cell loss priority" (CLP: 1 bit) is used to differentiate

the cells the loss of which is more or less acceptable. The field "header error

control" (HEC: 8 bits) is used to protect the cell head against transmission

errors. Finally, the field "generic flow control" (GFC: 4 bits) is only relevant

in the subscriber area and is discussed for flow control in the case of multiple

access of terminal equipment in the subscriber area.

Chapter 4: Mapping

36

This field is not significant network-internally; its place in the cell head is

therefore used network-internally to extend the virtual path identifier (to 12

bits).

Like all packet-oriented methods ATM has several peculiarities when

compared to STM. Apart from the conventional bit errors in the information,

entire cells can be lost in ATM. However, the probability that this actually

happens is low (e.g. 10-8). Possible causes are, for example, incorrigible

errors in the cell head or overflow of network-internal queues which have to

be established at every multiplex point in the network due to the statistical

multiplex principle.

Further peculiarities relate to the runtime. The procedure used means that in

an ATM network the end-to-end runtime for cells will be lower than in a

narrowband ISDN. A considerable "packeting time" may arise for filling the

ATM cells, however, particularly with lower bit rates. This packeting time

arises whenever the information appears at the source as continuous signal (as

is the case with most computer applications) rather than in packet form. The

packeting time for PCM-coded speech (64 kbit/s) is as much as 6 ms. Thanks

to the low runtime in the ATM network, this delay hardly disturbs speech

quality as long as no additional packeting procedures arise through transitions

to STM networks. Special measures such as the use of echo suppressor

equipment, guarantee the standard high speech quality in such cases.

Finally, unlike in STM, statistical runtime fluctuations (delay jitter) arise in

ATM due to the intra-network queues. In the case of continuous signals (e.g.

speech) the receiver must balance fluctuations through an anticipated delay.

Chapter 4: Mapping

37

4.2 Transmission of ATM Cells in SDH

Fig. 40

4.3 "Mapping" of ATM Cells to the Container C-4

Prior to their transmission in the STM-1 frame, the ATM cells are interleaved

into the container C-4 (= Mapping).

Fig. 41

Chapter 4: Mapping

38

Fig. 42

A C-4 is provided every 125 µs as network-synchronous transmission

capacity.

A comparison between the number of possible, usable bits per container C-4

260 byte x 9 = 2340 byte

and the number of ATM cells to be transmitted per C-4

2340 byte: 53 byte = 44,15

reveals that an ATM cell can also be transmitted via two C-4.

4.4 Interleaving the C-4 into the STM-1

In order to transmit the container C-4 in the STM-1, container-specific

supplementing is necessary:

1) The C-4 receives a "Path OverHead" (POH) with a size of 9 byte.

The block resulting from the C-4 and POH is called Virtual Container-4 = VC-4.

Chapter 4: Mapping

39

Fig. 43

2) There is a "floating" embedding of the Virtual Container VC-4 to the

STM-1frame of the payload. Part of the VC-4 is transmitted in one STM-1

frame, and another part in the next frame.

The pointer indicates the start of the Virtual Containers (VC-4) in the

payload.

NOTE where there is a need to carry an ATM signal greater than

140Mbit/s such as a 600Mbit/s, then it is carried in an STM 4 frame as

shown below. This signal has its 1st VC 4 with a normal AU4 pointer, and

the other 3 AU4's have concatenated pointers.

Fig. 44

Chapter 4: Mapping

40

That component of the STM-1, inside which a VC-4 is able to "float" and

which consists of the blocks PTR and payload, is termed Administrative Unit

4 = AU-4.

In the AU-4 there is an AU-4 Pointer abbreviated by AU-4 PTR.

Fig. 45

3) To complete the STM-1 or STM 4 frame, the Section OverHead (SOH)

is added to the AU-4.

Fig. 46

Chapter 4: Mapping

41

5 Concatenation of Payloads

5.1 Introduction

There is an increasing need to have ultra high capacity interfaces which

require several 155Mbit/s channels in today's Data applications.

Therefore if we can transmit data in one channel rather than several individual

channels, then we would have better utilization of the available channel

bandwidth. This would also prove to be a much less expensive solution for our

customers.

These large channels are called "Clear Channels". Concatenation is the merger

of multiple channels (say 155Mbit/s) into one large Clear Channel.

There are TWO distinct methods of CONCATENATION

1. CONTIGUOUS CONCATENATION.

2. VIRTUAL CONCATENATION.

An example of where this feature is used is when there is a requirement to

carry IP packets over the SDH network.

This is a detailed subject which is not covered in this course.

For further reading, an introduction, in the form of a tutorial, can be found in

the Appendix of this training manual. Please take some time later to read this

as it gives good background information on this topic.

Chapter 4: Mapping

42

5.2 Contiguous Concatenation of Payloads

To illustrate the form of concatenation we will use the concatenation of VC4's

initially, then describe the TU 2's AU4's can be concatenated to form and AU4

Xc which can transport payloads requiring greater than one Container 4

capacity.

The concatenation indication is used so that the multi container VC4 Xc

payload should be kept together, and is part of the VC 4 pointer. The X

indicates the number of VC 4 concatenated, e.g. VC 4 4c means 4 VC 4

concatenated i.e. capacity of 599.040 Mbit/s.

The first AU 4 of an AU4 Xc has the normal range of pointer values. All

subsequent AU 4 within the AU4 Xc will have their pointer set to

Concatenation Indication (CI) "1001" in bits 1 to 4, with bits 5 & 6

unspecified, and the 10 bit decimal pointer values will have all 1's.

The CI indicates that the pointer processors will perform the same operations

as performed on the first AU4 of the AU4 Xc.

Recommendation G707 also describes the contiguous concatenation of TU 2

in VC3. The term VC 2mc is used where "m" indicates the number of

concatenated Tu 2's carried.

The first TU 2 of an TU 2mc has the normal range of pointer values. All

subsequent Tu 2's within the TU 2 mc will have their pointer set to

Concatenation Indication (CI) "1001" in bits 1 to 4, with bits 5 & 6

unspecified, and the 10 bit decimal pointer values will have all 1's.

The CI indicates that the pointer processors will perform the same operations

as performed on the first TU2 of the TU 2mc.

Chapter 4: Mapping

43

5.3 Virtual Concatenation of Payloads

The standard G707 describes the use of Virtual Concatenation of TU 2

payloads only. The Virtual Concatenation of VC 4 payloads is understudy,

however, it is expected that the process will remain the same.

This method of concatenation has been initially developed for the transport of

a single VC 2 mc, m times TU 2 without the use of CI of the pointer bytes.

This method only requires the path termination equipment to provide

concatenation functions.

Virtual Concatenation requires that all TU signals to be concatenated at the

origin of the path are to have the same pointer value. These TU's are then

carried in one VC4. When the VC4 is then terminated, all the concatenated

TU's must be passed unaltered from one interface to another and remain

within the VC4 with their time sequence unchanged.

With Virtual Concatenation the available capacity is lower than that for a

Contiguous Concatenation, therefore, care should be taken to base the required

capacity of VC 2mc's to the lower value to allowed the interconnections of

both types of concatenation. The reasons for this is because Virtual

concatenations need a separate POH for every VC 2 whereas contiguous

concatenation only requires a POH for the first VC signal. Byte stuffing

techniques are used to fill up the spare capacity in the Contiguously

concatenated TU 2mc.

Chapter 4: Mapping

44

Fig. 47 Two methods of concatenation

Fig. 48

Chapter 4: Mapping

45

Fig. 49

The diagram above shows the following:

1. The normal multiplex structure that has been already introduced.

2. The STM 0 multiplex structure are used Radio Relay products, such as:

• SRT1S.

• SRA1S.

3. The concatenated multiplex structure, also it can be seen that an STM 4/16/64/256 can carry both concatenated and non-concatenated signals.

Chapter 4: Mapping

46

6 Summary

In the SDH, containers with a fixed transmission capacity are provided every

125 µs.

Fig. 50

Container-Terminology (140 Mbit/s/ATM Cells)

The incorporation of the 140 Mbits/s signals/ATM cells into the STM-1 is

performed as follows:

Fig. 51

Chapter 4: Mapping

47

Container-Terminology (34 Mbit/s)

The interleaving of the three 34 Mbits/s signals into the STM-1 looks like this:

Fig. 52

Container-Terminology (2 Mbit/s)

The interleaving of the 63 x 2 Mbit/s signals to the STM-1 looks like this:

Fig. 53

Chapter 4: Mapping

48

The following PDH signals can be transmitted in a VC-4.

Fig. 54

Synchronous Digital Hierarchy accord. to ETSI

Fig. 55

Chapter 4: Mapping

49

7 Exercise

Fill in the missing components of the STM-1 signal in order to complete the

mapping function.

Chapter 4: Mapping

50

8 Solution

Chapter 5: Pointer

1

Chapter 5 Pointer

Aim of study This chapter introduces pointer functions, types and structure, pointer addressing scheme

and pointer justification.

Contents Pages

1 Pointer Functions 2

2 Pointer Types 3

3 Pointer Structure 6

4 Pointer Addressing Scheme 9

5 Pointer Justification 11

6 Exercise 15

7 Solution 16

Chapter 5: Pointer

2

Chapter 5

Pointer 1 Pointer Functions

The pointer is used for synchronization of tributaries and the higher-order

frame. Packed in the virtual container, the tributary signal can be transmitted

with a phase decoupled from that of the frame. The phase relationship

between frame and virtual container is recorded in the pointer bytes. The

pointer bytes are embedded in the frame at a fixed position and contain the

address of the first byte of the VC (1st POH byte) in the frame.

The pointer technique allows the tributary signals, which are packed in VC, to

be inserted in the higher-order frame without elaborate and time-consuming

buffering. Any phase and bit rate fluctuation can be compensated through

pointer value alignment together with byte-by-byte positive, zero and negative

justification.

Access to the higher-order virtual container (HO VC) is possible immediately

after evaluation of the AU pointer. A further pointer must be interpreted

before access to LO VC is possible.

The pointer allows single user channels to be dropped from and added to the

overall signal without the signal having to be demultiplexed completely.

Chapter 5: Pointer

3

2 Pointer Types

3 types of pointer can be distinguished:

a) AU pointer.

b) TU-3 pointer.

c) TU-1/TU-2 pointer.

The contents of H1 and H2 are as follow:

• Pointer value (address of container POH).

• New data flag.

• Justification opportunity digits.

• AU3/AU4/TU3-type.

H3 contains:

Pointer action byte

(for transmission of information with negative justification method)

Example of (a) and (b) type of pointers

Fig. 1 Basic structure of AU-x/TU-3 pointer

Chapter 5: Pointer

4

Example of (c) type of pointer

Fig. 2 Basic structure of TU-1x/2 pointer General pointer structure:

Fig. 3 General pointer structure Concatenation Indication (CI):

Application: A broadband signal divided into several subsignals is

transmitted in one STM-N.

In this case the standard pointer is set in STM-1 #1, while

CI is set in STM-1 #2. Thus, the phase relationship

between both STM-1 remains locked.

Chapter 5: Pointer

5

Fig. 4 Concatenation indication pointer structure

2.1 AU Pointer

The following AU pointer exists:

• AU-4 pointer.

• AU-3 pointer.

AU-x (x = 3, 4) pointer allow the phase and frequency adaptation of the VC-x

to a particular AU-x frame. This corresponds to a direct alignment of the

payload (VC) to the section overhead (SOH).

The following containers can be directly transported in the STM-1 frame:

1 x VC-4 (1 x 140 Mbit/s) by means of one AU-4 pointer.

3 x VC-3 (3 x 45 Mbit/s) by means of 3 AU-3 pointers.

2.2 TU-3 Pointer

The VC-3 containers can also be transported indirectly via a VC-4 container in

the STM-1 frame.

Chapter 5: Pointer

6

For indirect transmission, the VC-3 containers are initially aligned to the VC-4

frame by means of the TU-3 pointers; the VC-4 container is subsequently

aligned to the STM-1 frame with the AU-4 pointer.

3 x VC-3 can be carried in the VC-4 with 3 TU-3 pointers.

2.3 TU-12 Pointer

TU-12 pointer allows the VC-12 to be aligned in phase and frequency to the

higher-order frame (VC-3 or VC-4).

The VC-12 is transported in a multiframe, with merely one TU-12 pointer

byte being transmitted per 125 µs subframe. After three 125 µs subframes, the

transmission of the 3 pointer bytes is completed and the fourth subframe

carries a pointer reserve byte.

For transmission of the TU-12, several TU are combined in a group (tributary

unit group TUG) and the respective TUG are subsequently transferred to a

VC-4 or VC-3 container.

3 Pointer Structure

Fig. 5 AU-4 pointer

Chapter 5: Pointer

7

Fig. 6 TU-3 pointer

Chapter 5: Pointer

8

Fig. 7 H1, H2, H3 bytes as used in the AU-4/TU-3 pointer structure

Fig. 8

Chapter 5: Pointer

9

Fig. 9 TU-12 pointer structure

4 Pointer Addressing Scheme

Fig. 10 AU-4 pointer offset numbering

Chapter 5: Pointer

10

Fig. 11 TU-3 pointer addressing scheme

Fig. 12 TU-12 pointer offset numbering

Chapter 5: Pointer

11

5 Pointer Justification

When existing virtual containers are inserted into a higher-order frame, it is

possible to adjust phase and bit rate fluctuations by means of byte-by-byte

positive/zero/negative justification. This is necessary for example if several

STM-1 signals which are not network-synchronized meet at the network node.

If several STM-1 are multiplexed to a STM-N, for example, the higher-order

VC contained in the STM-1 are adapted to the STM-N frame.

Zero Justification:

If the VC to be inserted and the higher-order frame are in synchronism, no

justification is required. The phase difference (recorded in pointer value)

between frame and start of VC remains unchanged. This is termed zero

justification.

Positive Justification:

If the VC bit rate is too low compared to the frame transmission capacity, - i.e.

the available transmission capacity is higher than the one effectively required -

3 justification bytes (without information content) are, if required transmitted

instead of 3 VC information bytes at a defined position in the frame in order to

align the bit rates.

This corresponds to a positive justification operation. The start of the VC (1st

byte of POH) is consequently delayed in time by 3 bytes in relation to the

frame. This operation delays the start of the VC concerned by 3 bytes in time

and the pointer value must be incremented by 1.

Chapter 5: Pointer

12

Negative Justification:

If to the VC bit rate is too high compared to the frame transmission capacity -

i.e. the transmission capacity is inadequate - supplementary capacity must be

provided in the frame if required. This is accomplished by the transfer of 3

bytes of the VC content to the pointer action bytes. The phase difference

between frame and VC is thus decreased by 3 bytes and accordingly the

pointer value must be decremented by 1.

Pointer corrections are only permitted in every fourth frame, i.e. at least 3

consecutive frames with unchanged pointers must exist between 2 pointer

corrections.

Fig. 13 AU-4 positive pointer justification

Chapter 5: Pointer

13

Fig. 14 AU-4 negative pointer justification, standard case

Chapter 5: Pointer

14

5.1 Pointers and Signal Labels

Fig. 15

Chapter 5: Pointer

15

6 Exercise

1. What is the function of a pointer?

2. Which pointer exists in ETSI?

3. Which pointer value range exists for the AU-4 pointer and which bytes can

be addressed?

4. How is positive/zero/negative justification performed if the bit rate to be

transmitted is?

Identical with the transmission capacity?

Higher than the transmission capacity?

Lower than the transmission capacity?

5. What is the value of an AU-4 pointer originally set to 782 after positive

justification?

Chapter 5: Pointer

16

7 Solution

1. What is the function of a pointer?

The pointer indicates the address of the first POH bytes of the VC and thus

the location of the VC within the frame. It therefore serves to synchronize

payload information which can be accessed directly.

2. Which pointer exists in ETSI?

Pointer of the SDH: AU-4, TU-12, TU-3.

3. Which pointer value range exists for the AU-4 pointer and which bytes

can be addressed?

Value range 0 - 782 decimal; only every third byte can be accessed.

4. How is positive/zero/negative justification performed if the bit rate to

be transmitted is?

Identical with the transmission capacity? Zero justification

Higher than the transmission capacity? Negative justification

Lower than the transmission capacity? Positive justification

5. What is the value of an AU-4 pointer originally set to 782 after positive

justification?

To 0 (zero)

Chapter 6: Overhead

1

Chapter 6 Overhead

Aim of study This chapter introduces overhead functions & Section Overhead (SOH).

Contents Pages

1 Overhead Functions 2

2 Section Overhead (SOH) 3

3 Exercise 25

4 Solution 26

Chapter 6: Overhead

2

Chapter 6

Overhead

1 Overhead Functions

The functions of the overhead channels include:

• Frame formation.

• Status monitoring.

• Error monitoring.

• Error localization.

• Maintenance functions.

• Control functions.

The structure of the STM-1 or STM-N frame is such that the overhead is

always an entity separate form the useful information. The advantage of this

arrangement is that the individual overhead bytes can be interrogated, changed

or added at any time without the individual signal first having to be

demultiplexed.

A distinction is made between the section overhead (SOH) and path overhead

(POH).

Chapter 6: Overhead

3

Fig. 1 Affected area of overhead function

2 Section Overhead (SOH)

2.1 Basic Information

Fig. 2 STM-1 frame structure

Chapter 6: Overhead

4

The SOH block is composed of eight 9-column rows. The first 9 bytes of rows

1-3 respectively contain the RSOH (regenerator section overhead), while the

first 9 bytes of rows 5-9 contain the MSOH (multiplex section overhead). The

first 9 bytes of the 4th row are used by the AU pointers and are not a

component of the SOH.

2.2 Regenerator Section Overhead

Fig. 3 SOH structure of STM-1 highlighting the regenerator section overhead

2.2.1 Byte Description

A1 and A2 Framing bytes – These two byte types indicate the beginning of

the STM-N frame. The A1, A2 bytes are unscrambled. A1 has the binary

value 11110110, and A2 has the binary value 00101000. The frame alignment

word of an STM-N frame is composed of (3 x N) A1 bytes followed by

(3 x N) A2 bytes.

Chapter 6: Overhead

5

J0 Regenerator Section (RS) Trace message – It’s used to transmit a Section

Access Point Identifier so that a section receiver can verify its continued

connection to the intended transmitter. The coding of the J0 byte is the same

as for J1 and J2 bytes in the path overheads. This byte is defined only for

STM-1 number 1 of an STM-N signal.

Z0 - These bytes, which are located at positions S[1,6N+2] to S[1,7N] of an

STM-N signal (N > 1), are reserved for future international standardization.

B1 RS bit interleaved parity code (BIP-8) byte – This is a parity code (even

parity), used to check for transmission errors over a regenerator section. Its

value is calculated over all bits of the previous STM-N frame after

scrambling, then placed in the B1 byte of STM-1 before scrambling.

Therefore, this byte is defined only for STM- 1 number 1 of an STM-N signal.

E1 RS orderwire byte – This byte is allocated to be used as a local orderwire

channel for voice communication between regenerators.

F1 RS user channel byte – This byte is set aside for the user’s purposes; it

can be read and/or written to at each section terminating equipment in that

line.

D1, D2, D3 RS Data Communications Channel (DCC) bytes – These three

bytes form a 192 kbit/s message channel providing a message-based channel

for Operations, Administration and Maintenance (OAM) between pieces of

section terminating equipment. The channel can be used from a central

location for control, monitoring, administration, and other communication

needs.

Chapter 6: Overhead

6

DCC-channels

Fig. 4 Principle use of DCCM and DCCR channels

2.3 Multiplex Section Overhead

Fig. 5 Multiplex section overhead bytes

Chapter 6: Overhead

7

2.3.1 Byte Description

B2 Multiplex Section (MS) bit interleaved parity code (MS BIP-24) byte –

This bit interleaved parity N x 24 code is used to determine if a transmission

error has occurred over a multiplex section. It’s even parity, and is calculated

overall bits of the MS Overhead and the STM-N frame of the previous STM-

N frame before scrambling. The value is placed in the three B2 bytes of the

MS Overhead before scrambling. These bytes are provided for all STM-1

signals in an STM-N signal.

K1 and K2 Automatic Protection Switching (APS channel) bytes – These

two bytes are used for MSP (Multiplex Section Protection) signaling between

multiplex level entities for bi-directional automatic protection switching and

for communicating Alarm Indication Signal (AIS) and Remote Defect

Indication (RDI) conditions. The Multiplex Section Remote Defect Indication

(MS-RDI) is used to return an indication to the transmit end that the received

end has detected an incoming section defect or is receiving MS-AIS. MS-RDI

is generated by inserting a “110” code in positions 6, 7, and 8 of the K2 byte

before scrambling.

See Appendix for more details of the K1 and K2 bytes.

D4 to D12 MS Data Communications Channel (DCC) bytes – These nine

bytes form a 576 kbit/s message channel from a central location for OAM

information (control, maintenance, remote provisioning, monitoring,

administration and other communication needs).

Chapter 6: Overhead

8

S1 Synchronization status message byte (SSMB) – Bits 5 to 8 of this S1

byte are used to carry the synchronization messages. Following is the

assignment of bit patterns to the four synchronization levels agreed to within

ITU-T (other values are reserved):

Bits 5-8

0000 Quality unknown (existing sync. network)

0010 G.811 PRC

0100 SSU-A (G.812 transit)

1000 SSU-B (G.812 local)

1011 G.813 Option 1 Synchronous Equipment Timing Clock (SEC)

1111 Do not use for synchronization. This message may be emulated by

equipment failures and will be emulated by a Multiplex Section AIS signal.

M1 MS remote error indication – The M1 byte of an STM-1 or the first

STM-1 of an STM-N is used for a MS layer remote error indication (MS-

REI). Bits 2 to 8 of the M1 byte are used to carry the error count of the

interleaved bit blocks that the MS BIP- 24xN has detected to be in error at the

far end of the section. This value is truncated at 255 for STM-N >4.

E2 MS orderwire byte – This orderwire byte provides a 64 kbit/s channel

between multiplex entities for an express orderwire. It’s a voice channel for

use by craftspersons and can be accessed at multiplex.

Chapter 6: Overhead

9

Fig. 6 STM-4 frame

Fig. 7 STM-16 frame

Chapter 6: Overhead

10

2.4 Forward Error Correction FEC

Forward Error Correction (FEC) has been developed to improve the

transmission quality, by reducing the bit error rate. This is achieved by

correcting bit errors produced during optical to electrical conversion at the

receive side of the transmission path. Conversely, improvements can also be

used to reduce the necessary optical power required at the transmitter, thus

allowing a longer transmission path for the same primary bit error rate.

There are two types of FEC:

• Inband FEC.

• Outband FEC.

Inband FEC makes use of the spare bytes to be found in the Section Overhead

of the STM-4, 16, 64, 256. The previous figure of the STM-16 frame shows

the bytes reserved for this function.

Outband FEC uses an extra Overhead. This produces a higher improvement of

the Bit Error Rate; however, the signal bit rate will need to increase. This

method will probably not be used, however.

Currently FEC is not finalized by a specific standard. G707 standard has

allocated bytes and is under study (as at Jan 2001), therefore, Siemens are

currently using a proprietary version in their SLD 16 version 2.5 equipment,

until such time as the standard has been finalized.

On the transmit side the STM-4 part signal is put through an arithmetic-logic

unit which calculates the FEC Parity bytes from the 4 STM-1's signals.

These calculated parity FEC parity bytes are then inserted into the SOH of the

STM- 1 signals # 2, 3 and 4 (STM-1 # 1 is not used to carry FEC signals).

Chapter 6: Overhead

11

On the receive side of the STM-4 part signal, the signal is delayed and put

through an arithmetic-logic unit which calculates the correction information

by means of the received STM-4 part signal and the received FEC parity

bytes. This correction information is then used to correct the delayed STM-4

signal. A similar process is also used for the STM-16, STM-64, and STM-256

line rates.

This feature will be described in more detail during specific product training,

in products which supports this feature.

Some examples of improvements of BER and power gains by using FEC are:

Thus you may improve your BERPRIM to BERFEC for the same distance, or

for the same BERPRIM have more power to go an increased distance.

2.5 VC-3 and VC-4 Path Overhead (POH)

Basic Information

The POH is added to the container C. Both form together the virtual container

VC which is carried as unchanged entity in the network path. The POH

contains all information required for reliable transportation of the container.

Information about the status of the entire path can be obtained by evaluating

the POH data. Fig. 1 shows the scope of validity of the POH.

Chapter 6: Overhead

12

Description of VC-3/VC-4 POH bytes

VC-3, VC-4 POH

Fig. 8 VC-3, VC-4 path overhead

Byte Description

J1 Higher-Order VC-N path trace byte – This user-programmable byte

repetitively transmits a 15-byte, E.64 format string plus 1-byte CRC-7. A 64-

byte free-format string is also permitted for this Access Point Identifier. This

allows the receiving terminal in a path to verify its continued connection to the

intended transmitting terminal.

Chapter 6: Overhead

13

B3 Path bit interleaved parity code (Path BIP-8) byte – This is a parity

code (even), used to determine if a transmission error has occurred over a

path. Its value is calculated over all the bits of the previous virtual container

before scrambling and placed in the B3 byte of the current frame.

C2 Path signal label byte – This byte specifies the mapping type in the VC-

N. Standard binary values for C2 are:

MSB LSB Hex Code Interpretation

Bits 1-4 Bits 5-8

0000 0000 (00) Unequipped or supervisory-unequipped

0000 0001 (01) Equipped – non-specific

0000 0010 (02) TUG structure

0000 0011 (03) Locked TU-n

0000 0100 (04) Asynchronous mapping of 34,368 kbit/s or 44,736 kbit/s into

the Container-3

0001 0010 (12) Asynchronous mapping of 139,264 kbit/s into the Container-4

0001 0011 (13) ATM mapping

0001 0100 (14) MAN DQDB (IEEE Standard 802.6) mapping

0001 0101 (15) FDDI (ISO Standard 9314) mapping

0001 0110 (16) Mapping of HDLC/PPP (Internet Standard 51) framed signal

0001 0111 (17) Mapping of Simple Data Link (SDL) with SDH self

synchronizing scrambler

Chapter 6: Overhead

14

0001 1000 (18) Mapping of HDLC/LAP-S framed signals

0001 1001 (19) Mapping of Simple Data Link (SDL) with set-reset scrambler

0001 1010 (1A) Mapping of 10 Gbit/s Ethernet frames (IEEE 802.3)

1100 1111 (CF) Obsolete mapping of HDLC/PPP framed signal

1110 0001 (E1) Reserved for national use

1111 1100 (FC) Reserved for national use

1111 1110 (FE) Test signal, O.181 specific mapping

1111 1111 (FF) VC-AIS

G1 Path status byte – This byte is used to convey the path terminating status

and performance back to the originating path terminating equipment.

Therefore the bidirectional path in its entirety can be monitored, from either

end of the path. Byte G1 is allocated to convey back to a VC-4-Xc/VC-4/VC-

3 trail termination source the status and performance of the complete trail. Bits

5 to 7 may be used to provide an enhanced remote defect indication with

additional differentiation between the payload defect (PLM), server defects

(AIS, LOP) and connectivity defects (TIM, UNEQ). The following codes are

used:

Chapter 6: Overhead

15

Fig. 9 G1 byte "Path Status"

F2 Path user channel byte – This byte is used for user communication

between path elements.

H4 Position and Sequence Indicator byte – This byte provides a multi frame

and sequence indicator for virtual VC-3/4 concatenation and a generalized

position indicator for payloads. In the latter case, the content is payload

specific (e.g., H4 can be used as a multiframe indicator for VC-2/1 payload).

For mapping of DQDB in VC- 4, the H4 byte carries the slot boundary

information and the Link Status Signal (LSS). Bits 1-2 are used for the LSS

code as described in IEEE Standard 802.6. Bits 3-8 form the slot offset

indicator. The slot offset indicator contains a binary number indicating the

offset in octets between the H4 octet and the first slot boundary following the

H4 octet. The valid range of the slot offset indicator value is 0 to 52. A

received value of 53 to 63 corresponds to an error condition.

F3 Path user channel byte – This byte is allocated for communication

purposes between path elements and is payload dependent.

Chapter 6: Overhead

16

K3 APS signaling is provided in K3 bits 1-4, allocated for protection at the

VC-4/3 path levels. K3 bits 5-8 are allocated for future use. These bits have no

defined value. The receiver is required to ignore their content.

N1 Network operator byte – This byte is allocated to provide a Higher-Order

Tandem Connection Monitoring (HO-TCM) function. N1 is allocated for

Tandem Connection Monitoring for contiguous concatenated VC-4, the VC-4

and VC-3 levels. Bits 1-4 Incoming Error Count (IEC).

1001 0

0001 1

0010 2

0011 3

0100 4

0101 5

0110 6

0111 7

1000 8

1110 Incoming AIS

NOTE: To guarantee a non all-zeroes N1 byte independent of the incoming

signal status, it is required that the IEC code field contains at least one “1”.

When zero errors in the BIP-8 of the incoming signal are detected, an IEC

code is inserted with “1”s in it. In this manner, it is possible for the Tandem

Connection sink at the tail end of the Tandem Connection link to use the IEC

code field to distinguish between unequipped conditions started within or

before the Tandem Connection. Bit 5 Operates as the TC-REI of the Tandem

Connection to indicate errored blocks caused within the Tandem Connection.

Bit 6 Operates as the OEI to indicate errored blocks of the egression VC-n.

Chapter 6: Overhead

17

Bits 7-8 Operate in a 76 multiframe as:

• Access point identifier of the Tandem Connection (TC-APId); it complies with the generic 16-byte string format.

• TC-RDI, indicating to the far end that defects have been detected within the Tandem Connection at the near end Tandem Connection sink.

• ODI, indicating to the far end that AU/TU-AIS has been inserted into the egression AU-n/TU-n at the TC-sink due to defects before

or within the Tandem Connection.

• Reserved capacity (for future standardization).

Chapter 6: Overhead

18

2.6 Low Order Path Multiframe

Fig. 10 Example for 500-_s multiframe of a TU-1/2 multiframe indicator using H4 bytes

Chapter 6: Overhead

19

2.7 VC-12 Path Overhead (POH)

Basic Information

In floating mode transmission of VC-12 four bytes (V5, J2, Z6, Z7) per 500 _s

are provided as POH.

Fig. 11 Path overhead of the VC-12

Description of the V5 byte

The V5 byte fulfills the following functions:

• Bit error monitoring.

• Signal labeling.

• VC-12 path status indication.

Chapter 6: Overhead

20

Fig. 12 V5 byte of VC-12

Byte Description

V5 VT path overhead byte.

Bits 1-2

Allocated for error performance monitoring. A Bit Interleaved Parity (BIP-2)

scheme is specified. Includes POH bytes, but excludes V1, V2, V3, and V4.

Bit 3

A VC-2/VC-1 path Remote Error Indication (LP-REI) that is set to one and

sent back towards a VC-2/VC-1 path originator if one or more errors were

detected by the BIP- 2; otherwise set to zero.

Chapter 6: Overhead

21

Bit 4

A VC-2/VC-1 path Remote Failure Indication (LP-RFI). This bit is set to one

if a failure is declared, otherwise it is set to zero. A failure is a defect that

persists beyond the maximum time allocated to the transmission system

protection mechanisms.

Bits 5-7

Provide a VC-2/VC-1 signal label. The Virtual Container path Signal Label

coding is:

000 Unequipped or supervisory-unequipped

001 Equipped – non-specific

010 Asynchronous

011 Bit synchronous

100 Byte synchronous

101 Reserved for future use

110 Test signal, O.181 specific mapping

111 VC-AIS

Bit 8

Set to 1 to indicate a VC-2/VC-1 path Remote Defect Indication (LP-RDI);

otherwise set to zero.

Chapter 6: Overhead

22

Path Trace J2

Byte J2 is used to transmit repetitively a Low Order Path Access Point

Identifier so that a path receiving terminal can verify its continued connection

to the intended transmitter. This Path Access Point Identifier uses the format

defined in clause 3/G831. A 16-byte frame is defined for the transmission of

Path Access Point Identifiers.

Network Operator Byte: N2

This byte is allocated to provide a Tandem Connection Monitoring (TCM)

function.

Fig. 13 N2 byte of a VC-12

Bits 1-2 Used as an even BIP-2 for the Tandem Connection.

Bit 3 Fixed to “1”. This guarantees that the contents of N2 is not all zeroes at

the TCsource. This enables the detection of an unequipped or supervisory

unequipped signal at the Tandem Connection sink without the need of

monitoring further OHbytes.

Chapter 6: Overhead

23

Bit 4 Operates as an “incoming AIS” indicator.

Bit 5 Operates as the TC-REI of the Tandem Connection to indicate errored

blocks caused within the Tandem Connection.

Bit 6 Operates as the OEI to indicate errored blocks of the egression VC-n.

Bits 7-8 Operate in a 76 multiframe as:

• The access point identifier of the Tandem Connection (TC-APId); it complies with the generic 16-byte string format.

• The TC-RDI, indicating to the far end that defects have been detected within the Tandem Connection at the near end Tandem Connection sink.

• The ODI, indicating to the far end that TU-AIS has been inserted at the TC-sink into the egression TU-n due to defects before or within the Tandem Connection.

• Reserved capacity (for future standardization).

Chapter 6: Overhead

24

Automatic Protection Switching (APS) channel: K4 (b1-b4)

These bits are allocated for APS signaling for protection at the lower order

path level.

Reserved: K4 (b5-b7)

Bit 5 to 7 of K4 are reserved for an optional use. If this option is not used,

these bits shall be set to "000" or "111". A receiver is required to be able to

ignore the contents. The use of the optional function is at the discretion of the

owner of the trail termination source generating the K4-byte.

Spare: K4 (b8)

This bit is allocated for future use. This bit has no defined value. The receiver

is required.

Chapter 6: Overhead

25

3 Exercises

1. What is the function of the bytes A1, A2?

2. What is the function of the byte B1 and in which type of network elements

(multiplexer, regenerator) is it evaluated?

3. What is the function of the bytes B2 and in which type of network elements

(multiplexer, regenerator) is it evaluated?

4. Which byte and bits are used to transmit the signal multiplex section remote

defect indication?

5. Which byte is used for path trace?

Chapter 6: Overhead

26

4 Solution

1. What is the function of the bytes A1, A2?

Framing

2. What is the function of the byte B1 and in which type of network

elements (multiplexer, regenerator) is it evaluated?

Error monitoring on regenerator sections

Multiplexer

Regenerator

3. What is the function of the bytes B2 and in which type of network

elements (multiplexer, regenerator) is it evaluated?

Error monitoring on multiplexer sections

Only multiplexer

4. Which byte and bits are used to transmit the signal multiplex section

remote defect indication?

Byte K2, bit 6, 7, 8

5. Which byte is used for path trace?

J1

Chapter 7: Monitoring, Maintenance and Control in the SDH

1

Chapter 7 Monitoring, Maintenance and Control

in the SDH

Aim of study This chapter introduces alarm interactions overview, bit error monitoring and AIS.

Contents Pages

1 Alarm Interactions Overview 2

2 Bit Error Monitoring 5

3 Error Reports REI, RDI 16

4 AIS 18

5 Examples 19

6 Exercises 32

7 Solution 33


2

Chapter 7

Monitoring, Maintenance and Control in the SDH 1 Alarm Interactions Overview

With the help of the overhead data bytes, we can send forward and backward

from the reporting network element certain alarm conditions. This information

helps to localize the fault as quickly as possible.

Using a process of prioritization, and elimination, we can determine where the

fault is, what is possibly causing it and what needs to be done to fix it.

The following diagram attempts to show the Alarms raised, their subsequent

actions, destinations in the forward and backward directions.

The following description should help to read the diagram:

(J0), (C2), (H4) etc are bytes to be found in the RSOH, MSOH, POH High

and low order.

Description

• The line shows the direction the Alarm is sent, with a description of the alarm event for example Loss of Signal, or Loss of frame.

• The following show all the alarms that cause the forwarding onwards or backwards of the next alarm indication as required.

• The alarm names indicate the alarms that all cause the subsequent alarm indication and the "1" indicates the contents of the STM frame contains all "1"s in the AU4.


3

Fig. 1

Fig. 2


4

1.1 Abbreviations


5

2 Bit Error Monitoring

Specific bytes in the individual overheads are provided for bit error

monitoring and fault localization. These bytes contain information indicating

the bit error rate and thus the quality of the transmission sections concerned.


6

Fig. 3

2.1 Operational Principle

Fig. 4 Bit error monitoring


7

On the transmit side, an n-bit code word is generated over a bit stream of

specific length in conformity with a fixed code protocol. This code word is

carried supplementary to the useful information in the overhead.

The bit stream is coded according to the same rules on the receive side and a

code word is regenerated. The new code word is compared with the

transmitted one. Any discrepancy between the code words indicates bit errors

in transmission. The precise number of bit errors is not determined with this

audit. However, a statistical evaluation of the incorrect code words allows

conclusion to be drawn about the transmission bit error rate.

2.2 BIP-n Code

Fig. 5 Bit-interleaved parity (BIP-n)

A special parity code known as the BIP-n code is provided for bit error

monitoring in the synchronous hierarchy.

BIP-n: bit interleaved parity n


8

BIP-n code generation:

Here the bit stream of the multiplex unit under test (e.g. STM-N, VC) must be

envisaged as divided into sequences n bits in length.

Parity is now generated over the first bit of each sequence respectively and

even parity is produced at the end of the multiplex unit being tested. The even-

parity bit corresponds to the 1st bit of the n-bit long code word.

To „produce even parity“means that there must be an even total (including the

parity bits in the code word) of „1s“in the particular bit stream must exist.

The same procedure is applied to the 2nd bit in each sequence with the result

that the 2nd bit of the code word is generated. This continues in the same

manner until all n-bits are generated. The n-bit long code word is then inserted

and carried in the appropriate overhead.

2.3 Monitoring Sections

Fig. 6 Monitoring sections


9

Different BIP-N code words are used to monitor individual route sections in

the synchronous hierarchy:

Regenerator Section: B1 in RSOH

One BIP-8 code word (1 byte) is provided for bit error monitoring. This code

word is generated over all bits in the STM-N frame after scrambling. The BIP-

8 byte is subsequently inserted in the allocated position B1 of the RSOH in the

next frame before scrambling starts. This byte is evaluated and regenerated in

every multiplexer and regenerator.

Fig. 7 B1, B2 generation


10

Multiplexer Section: B2 in the MSOH

A BIP-N x 24 code word (N x 3 bytes) is provided for error monitoring on the

individual multiplexer sections. This BIP-N x 24 code cord is generated prior

to scrambling over the entire STM-N frame, but not on the first 3 rows of the

SOH. The N x 3-byte BIP-N24 x 24 code word is inserted prior to scrambling

in the N x 3 bytes B2 provided for this purpose in the MSOH of the next

frame. These B2 bytes are not modified in the regenerator.

VC-4 and VC-3 Path: B3 in the POH

A B3 byte is provided for error monitoring of the individual VC-3 and VC-4

transmission paths. One BIP-989 (1 byte) code word is generated over the

entire bit stream of the virtual container and inserted in the appropriate byte

B3 of the POH of the follow-on VC. The B3 code word is generated over the

entire VC bit stream including the POH but without pointers. In the case of

negative justification it must be noted that the pointer action byte contains

useful information of the VC and is therefore incorporated in the B3

generation.

VC-2 and VC-1 Path: Bit 1 and 2 in the V5 POH

The first 2 bits in the POH byte V5 of the respective VC are provided for bit

error monitoring of the individual VC-1 and VC-2 transmission paths. A BIP-

2 (2 bits) code word is generated over the entire VC block in the 500 _s

multiframe and inserted in the first two bit positions of the POH (V5) of the

next VC.


11

2.4 Tandem Connection Monitoring

2.4.1 Where do the Errors come from?

End-to-end quality is monitored by checking BIP parity. This gives an

indication of whether errors have been generated somewhere in the entire

path.

It is not possible to determine in which part of the path the error occurred.

If a sub-network provider is present (provider 2), there will always be disputes

over who produced the errors on the way through the network.

An additional possibility allowing sub-network providers to demonstrate the

quality of their networks from the point of receiving to the point of

transmitting the signal from their network limits was therefore looked for.

Fig. 8


12

2.4.2 Principle of Tandem Connection Monitoring

"Tandem Connection Monitoring" was introduced for this reason.

The principle is very simple:

The incoming and outgoing data streams (SINK and SOURCE) are each

monitored at the network limits.

This allows network provider 2 to monitor own errors in the path layer

independently of any received errors.

Fig. 9


13

2.4.3 Check TCM Sub-Layer

Path parity errors are checked at the input to the sub-network. If errors are

present, they are copied into N1/N2 bytes in the POH. The data now passes

through the sub-network.

At the far end of the sub-network a check is made again: Path parity errors are

checked and compared with the extracted N1/N2 bytes.

If there is a difference, the sub-network produced additional errors. Otherwise,

provider 1 is responsible for the errors.

Fig. 10


14

2.4.4 Alarm and Error Handling with TCM

In addition to error monitoring, alarms are also signaled in the backward

direction in the same way as in the VCn layer. This allows monitoring of the

entire TCM systems in the forward and backward directions.

The following events are used to signal alarms:

SOURCE:

• Invalid VC-n? --> Insert AIS

• BIP errors detected? --> Insert errors in IEC (incoming error count)

• Alarms received from SINK --> TC-RDI : Remote Defect Indication

--> TC-REI: Remote Error Indication

--> ODI: Outgoing Defect Indication

--> OEI: Outgoing Error Indication

SINK:

• TC alarms detected? --> TC-RDI

• AIS detected? --> ODI

• BIP errors detected? --> Insert errors in OEI

• BIP = IEC? --> Insert difference in TC-REI


15

Fig. 11

2.4.5 Interaction between Generation and Analysis of N1/N2

Detected B3 or BIP-2 errors are indicated in bytes N1/N2 of the sub-network.

In the USA, only N1 (Z5) is taken into consideration.

The right-hand figure shows the TCM sink and source functions. There is an

exchange of errors and alarms in the incoming and outgoing data signals.

N1/N2 produces a multiframe with 76 frames which allows transportation of

different alarms. N1 also transports the number of B3 errors counted, and N2

transports the value of BIP-2 errors.

Recommendations G.707 and G.783 cover SDH.


16

T1.105 and T1.105.05 apply to SONET (Bellcore GR-253 only refers to the

ANSI recommendation).

Fig. 12

3 Error Reports REI, RDI

3.1 REI Remote Error Indication

a) Path REI

The POH of the individual virtual containers contain one byte (VC-3 and

VC-4) or 2 bits (VC-1 and VC-2) for bit error monitoring. As mentioned

previously the BIP-8 or BIP-2 codes are used respectively. If bit errors are

detected at the path end when the BIP end words are evaluated, a REI code is

inserted in the opposite direction (to the path start) in order to notify the

source of the detected failure.


17

Bits 1-4 in POH byte G1 are used for REI transmission by VC-3 and VC-4.

The parity of 8 bit sequences is checked with the BIP-8 code employed.

Maximum 8 parity violations can thus be detected. The REI code contains the

total number of parity violations, with the values 0 to 8 being transmitted.

Should a different value appear in the REI code, however, it must be

interpreted as 0.

Bit 3 in POH byte V5 is used for REI transmission by VC-1 and VC-2. The bit

is set to 0 if no parity violation is detected with the BIP-2. A parity error is

indicated by the value 1.

b) Section REI

The M1 byte in the MSOH is used for relaying the number of parity violations

occurred in the B2 bytes to the far end side.

Depending on the multiplex signal (STM-N), the M1 REI code can have

values between 0 and N x 24.

3.2 RDI Remote Defect Indication

a) Path-RDI

If no valid signal or an AIS is present when the individual VCs are received,

the distant end is notified through the remote alarm.

This remote alarm is set to „1“in the event of a fault; in normal operation its

value is „0“.

The remote alarm is carried in POH byte G1 (bit 5) for VC-3 and VC-4.

For VC-1 and VC-2, the remote alarm is carried in POH byte V5 (bit 8).


18

b) Section-RDI

If the STM-N multiplexer receives an AIS or no valid signal, it inserts the RDI

code in the opposite direction.

The RDI code (110) is inserted in byte K2 at bit positions 6-8.

4 AIS

AIS = alarm indication signal

a) Definition

If a device detects an error, e. g. no valid signal or loss of frame alignment, it

sends an alarm indication signal (AIS) in the ongoing direction in the same

manner as a normal signal is relayed by the follow-on equipment. The purpose

of this signal is to prevent the activation of alarms in the follow-on equipment.

The reception of an AIS signal triggers direct functions (such as channel

blocking) only in specific terminal equipment.

The AIS signal is an all-one-signal in the plesiochronous hierarchy. The frame

alignment signal and service word are likewise set to „1“, so that the frame

alignment signal is no longer detectable as such.

In the synchronous hierarchy, the STM-1 frame is fully retained even in the

event of an AIS. A distinction is made between the section AIS and the path

AIS.

b) Path AIS

Path AIS is set if a virtual container fails.

The entire TU-n (n = 1, 2, 3) including pointer is set to „1“ in the case of

a TU-path AIS.


19

The entire AU-n (n = 3, 4) including pointer is set to „1“ in the case of

a AU-path AIS.

These permanent one signals are carried in the STM-1 as valid tributaries.

c) Section AIS

Section AIS is set if the entire STM-1 or STM-N has failed. It is indicated in

byte K2, of which bits 6, 7 and 8 are set to „1“.

5 Examples

Fig. 13

The three B1 are generated in the Regenerator Section, and monitor the STM –N

frame after scrambling. This B1, BIP 8 code is subsequently inserted in the B1

position of the RSOH in the next frame, before scrambling. this byte is then

evaluated and regenerated along the route at each Multiplexer and Regenerator.

Fig. 14


20

The three B2 are generated prior to scrambling over the entire STM-1 in the

STM-N signal, but not on the first three rows of the SOH (=RSOH). The B2

monitor individual STM-1 signals on the multiplex section; they are only

generated (TRANSMITTER) and evaluated (RECEIVER) in MUX systems.

Fig. 15

The B3 is provided over the entire VC-4 / VC-3. It is generated at the

beginning of the path and evaluated at the end of the path.

5.1 Bit Error Monitoring Concept (Examples)

B1 - code errors are only indicated in those MUX/REG systems which are

contained in the faulty regenerator section.

B2 - code errors are only indicated in those MUX systems which are

contained in the faulty multiplex section.

B3 - code errors are indicated in those MUX systems, in which a VC-4/VC-3

is evaluated (access to PDH signals).


21

Fig. 16

Example 1

Fig. 17

Example 2

Fig. 18


22

Example 3

Fig. 19

Example 4

Fig. 20

Example 5

Fig. 21


23

5.2 Error and Failure Reports

Error report "REI" (Remote Error Indication):

REI is sent in the backward direction, if there are code errors (bit errors) in

the incoming signal of the local receiver (MUX).

Fig. 22

There are two types of REI:

Path REI

Path REI if a code error was determined in the B3 byte.

Fig. 23


24

Section REI

Section REI if a code error was determined in the B2 byte.

Fig. 24

Example REI

Fig. 25

Since MUX 2 detects a bit error rate SD in bytes B2 and B3, there is an error

report due to which Path REI and Section REI is indicated in MUX1.


25

Failure Report "RDI" (Remote Defect Indication

RDI is reported in the backward direction in the case of urgent line alarms.

Fig. 26

A distinction is made between:

PATH RDI SECTION RDI

Bit 5 of the G1 byte in VC4 POH is

set to "1".

Bits 6, 7 and 8 of the K2 byte in

MSOH are set to "110".

B3 SD B2 LOS

AIS in VC-4 SECTION AIS

no signal in the VC-4 loss of STM-N signal

wrong path trace in the J1 byte loss of frame alignment


26

Example RDI

Fig. 27

Since MUX 2 detected a LOS or SD in the B2 byte and, implicitly in the B3

byte, a RDI (Section + Path) report is sent to the MUX 1.

5.3 Alarm Indication Signal "AIS"

AIS is sent to the forward direction, if urgent line alarms were detected in the

MUX/REG.

Fig. 28


27

There are two types of AIS:

Path AIS

e.g. with VC-4 the entire AU-4 including the pointer is set to "1".

Section AIS

Fig. 29 Path AIS

Fig. 30 Section AIS


28

Bits 6, 7 and 8 of the K2 in the MSOH are set to "111".

AIS is sent in the forward direction if the following conditions were detected

in the MUX/REG:

PATH AIS SECTION AIS

B3 SD section AIS already received

(in regenerators)

NO signal in the VC – 4 NO signal in the STM – N

(in regenerators)

Wrong path trace J1 byte loss of STM-N signal

(in regenerators)

Path AIS already received internal functional disturbances

in the MUX/REG systems

Example Path AIS

A cable break in the regenerator by the alarm "Loss of Signal" (LOS).

The regenerator cannot re-generate the STM-4 signal and sends

"Section AIS".

The MUX 2 transmits all-one-signals in channels 1, 2 and 3. The PDH

devices interpret these signals as AIS.


29

Fig. 31

MUX 2 sends Path AIS (Pointer + VC-4 = "1") in channel #4.

The last SDH MUX (MUX 3) also sends AIS to the PDH device.

Error reports are issued in the backward direction, too, of course.

MUX 2 sends Section RDI and Path RDI for paths #1, #2 and #3 to MUX 1.

MUX 3 sends Path FERF for channel #4 , which is switched through until the

end of the path (MUX1).

5.4 Summary

Bit Error Monitoring

In order to monitor an STM-N signal, the "Bit Interleaved Parity" (BIP)

procedure is used.

• B1 (BIP8) = monitoring of the entire STM - N signals on

regenerator sections.


30

• B2 (BIP N x 24) = monitoring of an STM - 1 signal in an

STM - N on multiplex sections.

• B3 (BIP 8) = monitoring VC-4/VC-3 on path sections.

Error Report REI Remote Error Indication (previously called FEBE Far

End Block Error)


• Section REI is sent in the backward direction of the STM-N signal, if a code error was detected via the B2

bytes.

• Path REI is sent in the backward direction of the path if a code error was detected via the B3 byte.

Failure Report RDI Remote Defect Indication (previously called FERF

Far End Receive Fail)


• Section RDI is sent in the backward direction of the respective STM-N in

the following cases:

- -> Section AIS was received

- -> no signal (loss of STM-N)

- -> loss of frame alignment

• Path RDI is sent in the backward direction of the respective path in the

following cases:


31

- -> Path AIS in the VC

- -> no signal in the VC

- -> wrong path trace in the VC POH

AIS Alarm Indication Signal


• Section AIS is sent by the regenerators in the on-going direction in the

following cases:

- -> Section AIS was received

- -> no signal (loss of STM-N)

- -> loss of frame alignment

- -> internal functional disturbances (of the MUX, too)

• Path AIS is sent by MUX in the on-going direction of a path in the

following cases:

- -> SD

- -> no signal in the VC

- -> wrong path trace in the VC POH

- -> Path AIS already received in the path


32

6 Exercises

1. Which method is used for bit error monitoring?

2. Which byte is used to monitor the regenerator section with BIP?

3. Which byte is used to monitor the multiplex section with BIP?

4. Which byte is used to monitor the VC4 and VC3 path with BIP?

5. Which byte and bits are used to monitor the VC12 path with BIP?

6. List the indications for remote alarms.

7. List the types of AIS in the synchronous hierarchy.


33

7 Solution

1. Which method is used for bit error monitoring?

Bit-interleaved parity

2. Which byte is used to monitor the regenerator section with BIP?

B1

3. Which byte is used to monitor the multiplex section with BIP?

B2

4. Which byte is used to monitor the VC4 and VC3 path with BIP?

B3

5. Which byte and bits are used to monitor the VC12 path with BIP?

V5, bits 1 and 2

6. List the indications for remote alarms.

Remote error indication REI

Remote defect indication RDI

7. List the types of AIS in the synchronous hierarchy.

Path AIS

Section AIS

Chapter 8: Appendix

1

Chapter 8 Appendix

Aim of study This chapter introduces ITU-T Recommendation list, multiplex section overhead bytes K1

& K2, SONET and IP over SDH.

Contents Pages

1 ITU-T Recommendation List 2

2 Multiplex Section Overhead bytes K1 & K2 6

3 SONET 7

4 IP over SDH 10

Chapter 8: Appendix

2

Chapter 8

Appendix

1 ITU-T Recommendation List

A Selection of ITU-T Recommendations has been given below as a guide for

further reading:

• Recommendation G.652 (10/00) - Characteristics of a single-mode optical fiber cable - To be published

• Recommendation G.653 (10/00) - Characteristics of a dispersion-shifted single-mode optical fiber cable- To be published

• Recommendation G.654 (10/00) - Characteristics of a cut-off shifted single-mode optical fiber cable - To be published

• Recommendation G.655 (10/96) - Characteristics of a non-zero dispersion shifted single-mode optical fiber cable - To be published

• Recommendation G.662 (10/98) - Generic characteristics of optical fibre amplifier devices and subsystems

• Recommendation G.663 (04/00) - Application related aspects of optical amplifier devices and subsystems - To be published

• Recommendation G.664 (06/99) - Optical safety procedures and requirements for optical transport systems

• Recommendation G.671 (11/96) - Transmission characteristics of passive optical components

• Recommendation G.681 (10/96) - Functional characteristics of interoffice and long-haul line systems using optical amplifiers, including optical multiplexing

• Recommendation G.691 (10/00) - Optical interfaces for single channel STM-64, STM-256 systems and other SDH systems with optical amplifiers - To be published

Chapter 8: Appendix

3

• Recommendation G.692 (10/98) - Optical interfaces for multichannel systems with optical amplifiers

• Recommendation G.701 (03/93) - Vocabulary of digital transmission and multiplexing, and pulse code modulation (PCM) terms

• Recommendation G.702 (11/88) - Digital hierarchy bit rates

• Recommendation G.703 (10/98) - Physical/electrical characteristics of hierarchical digital interfaces

• Recommendation G.704 (10/98) - Synchronous frame structures used at 1544, 6312, 2048, 8448 and 44 736 kbit/s hierarchical levels

• Recommendation G.705 (10/00) - Characteristics of Plesiochronous Digital Hierarchy (PDH) equipment functional blocks - To be published

• Recommendation G.706 (04/91) - Frame alignment and cyclic redundancy check (CRC) procedures relating to basic frame structures defined in G 704

• Recommendation G.707/Y.1322 (10/00) - Network node interface for the synchronous digital hierarchy (SDH) - To be published

• Recommendation G.708 (06/99) - Sub STM-0 network node interface for the synchronous digital hierarchy (SDH)

• Recommendation G.711 (11/88) - Pulse code modulation (PCM) of voice frequencies

• Recommendation G.732 (11/88) - Characteristics of primary PCM multiplex equipment operating at 2048 kbit/s

• Recommendation G.736 (03/93) - Characteristics of a synchronous digital multiplex equipment operating at 2048 kbit/s

• Recommendation G.773 (03/93) - Protocol suites for Q-interfaces for management of transmission systems

• Recommendation G.774 (09/92) - Synchronous digital hierarchy (SDH) management information model for the network element view

• Recommendation G.780 (06/99) - Vocabulary of terms for synchronous digital hierarchy (SDH) networks and equipment

• Recommendation G.781 (06/99) - Synchronization layer functions

Chapter 8: Appendix

4

• Recommendation G.783 (10/00) - Characteristics of synchronous digital hierarchy (SDH) equipment functional blocks - To be published

• Recommendation G.784 (06/99) - Synchronous digital hierarchy (SDH) management

• Recommendation G.801 (11/88) - Digital transmission models

• Recommendation G.803 (03/00) - Architecture of transport networks based on the synchronous digital hierarchy (SDH) - To be published

• Recommendation G.804 (02/98) - ATM cell mapping into plesiochronous digital hierarchy (PDH)

• Recommendation G.805 (03/00) - Generic functional architecture of transport network To be published

• Recommendation G.806 (10/00) - Characteristics of transport equipment - Description methodology and generic functionality - To be published

• Recommendation G.810 (08/96) - Definitions and terminology for synchronization networks

• Recommendation G.811 (09/97) - Timing characteristics of primary reference clocks

• Recommendation G.812 (06/98) - Timing requirements of slave clocks suitable for use as node clocks in synchronization networks

• Recommendation G.813 (08/96) - Timing characteristics of SDH equipment slave clocks (SEC)

• Recommendation G.821 (08/96) - Error performance of an international digital connection operating at a bit rate below the primary rate and forming part of an integrated services digital network

• Recommendation G.822 (11/88) - Controlled slip rate objectives on an international digital connection

• Recommendation G.823 (03/00) - The control of jitter and wander within digital networks which are based on the 2048 kbit/s hierarchy - To be published

• Recommendation G.824 (03/00) - The control of jitter and wander within digital networks which are based on the 1544 kbit/s hierarchy - To be published

Chapter 8: Appendix

5

• Recommendation G.825 (03/00) - The control of jitter and wander within digital networks which are based on the synchronous digital hierarchy (SDH) - To be published

• Recommendation G.826 (02/99) - Error performance parameters and objectives for international, constant bit rate digital paths at or above the primary rate

• Recommendation G.831 (03/00) - Management capabilities of transport networks based on the synchronous digital hierarchy (SDH) - To be published

• Recommendation G.832 (10/98) - Transport of SDH elements on PDH networks - Frame and multiplexing structures

• Recommendation G.841 (10/98) - Types and characteristics of SDH network protection architectures

• Recommendation G.842 (04/97) - Interworking of SDH network protection architectures

• Recommendation G.852.1 (11/96) - Management of the transport network - Enterprise viewpoint for simple sub network connection management

• Recommendation G.853.1 (03/99) - Common elements of the information viewpoint for the management of a transport network

• Recommendation G.853.2 (11/96) - Sub network connection management information viewpoint

• Recommendation G.950 (11/88) - General considerations on digital line systems

• Recommendation G.957 (06/99) - Optical interfaces for equipments and systems relating to the synchronous digital hierarchy

• Recommendation G.958 (11/94) - Digital line systems based on the synchronous digital hierarchy for use on optical fiber cables

Chapter 8: Appendix

6

2 Multiplex Section Overhead bytes K1 & K2

2.1 K1 Byte

Bits 1-4 Type of request

1111 Lock out of Protection

1110 Forced Switch

1101 Signal Fail – High Priority

1100 Signal Fail – Low Priority

1011 Signal Degrade – High Priority

1010 Signal Degrade – Low Priority

1001 (not used)

1000 Manual Switch

0111 (not used)

0110 Wait-to-Restore

0101 (not used)

0100 Exercise

0011 (not used)

0010 Reverse Request

0001 Do Not Revert

0000 No Request

Bits 5-8 indicate the number of the channel requested

Chapter 8: Appendix

7

2.2 K2 Byte

Bits 1-4 Selects channel number

Bit 5 Indication of architecture

0 1+1

1 1: n

Bits 6-8 Indicate mode of operation

111 MS-AIS

110 MS-RDI

101 Provisioned mode is bi-directional

100 Provisioned mode is unidirectional

011 Future use

010 Future use

001 Future use

000 Future use

3 SONET

3.1 Introduction

SONET (Synchronous Optical NETwork) is a standard for optical

telecommunications transport. It was formulated by the Exchange Carriers

Standards Association (ECSA) for the American National Standards Institute

(ANSI), which sets industry standards in the U.S. for telecommunications and

other industries.

Chapter 8: Appendix

8

3.2 Background

Before SONET, the first generations of fiber optic systems in the public

telephone network used proprietary architectures, equipment, line codes,

multiplexing formats, and maintenance procedures. The users of this

equipment – Regional Bell Operating Companies and inter-exchange carriers

(IXCs) in the U.S., Canada, Korea, Taiwan, and Hong Kong – wanted

standards so they could mix and match equipment from different suppliers.

The task of creating such a standard was taken up in 1984 by the Exchange

Carriers Standards Association (ECSA) to establish a standard for connecting

one fiber system to another. This standard is called SONET for Synchronous

Optical NETwork.

3.3 Basic SONET Signal

SONET defines a technology for carrying many signals of different capacities

through a synchronous, flexible, optical hierarchy. This is accomplished by

means of a byte-interleaved multiplexing scheme. Byte-interleaving simplifies

multiplexing, and offers end-to-end network management. The first step in the

SONET multiplexing process involves the generation of the lowest level or

base signal. In SONET, this base signal is referred to as Synchronous

Transport Signal level-1, or simply STS-1, which operates at 51.84 Mb/s.

Higher-level signals are integer multiples of STS-1, creating the family of

STS-N signals. An STS-N signal is composed of N byte-interleaved STS-1

signals. The table includes the optical counter (carrier)-part for each STS-N

signal, designated OC-N (Optical Carrier level-N).

Chapter 8: Appendix

9

3.3.1 SONET HIERARCHY

3.3.2 NON-SYNCHRONOUS HIERARCHY

3.4 STS Frame Structure

STS-1 is a specific sequence of 810 bytes (6480 bits), which includes various

overhead bytes and an envelope capacity for transporting payloads. It can be

depicted as a 90 column by 9 row structure. With a frame length of 125 µs

(8000 frames per second), STS-1 has a bit rate of 51.840 Mb/s. The order of

transmission of bytes is row-by-row from top to bottom, left to right (most

significant bit first). The first three columns of the STS-1 frame are for the

Transport Overhead. The three columns each contain nine bytes. Of these,

nine bytes are overhead for the Section layer (for example, Section Overhead),

and 18 bytes are overhead for the Line layer (for example, Line Overhead).

Chapter 8: Appendix

10

The remaining 87 columns constitute the STS-1 Envelope Capacity (payload

and path overhead).

Fig. 1

4 IP over SDH

4.1 Overview

The objective is quite clear. Service Providers require a mechanism that lets

them deliver the proper services where and when their customers need them.

Data traffic is increasing dramatically whilst carriers will enjoy to use the

existing circuit-switched equipment. Although the combination of data and

voice sounds to be incompatible, the reality of both networks is changing.

New services like voice and video over IP on the one hand and the rapidly

increasing bandwidth supply on the other suggest this combination.

The existence of a widespread, commonly used and highly reliable network

well proven over years is in store for the carriers that tread new paths towards

IP. Standardization ensures a large degree of interoperability. Bandwidth

demand is no longer a real obstacle, but low transit-delay (latency) is. Lowest

latency however is the favorite domain of SDH and Sonet.

Chapter 8: Appendix

11

Starting with point to point connectivity solutions and ending up with an

entirely new generation of SDH-integrated IP-Routers, Siemens presents the

way how to combine SDH/Sonet and IP in the most efficient manner.

To understand how this can be achieved requires knowledge of both, basics of

data traffic as well as SDH precautions taken for it.

4.2 Data Traffic in Short

4.2.1 Local Area Networks

LANs (Local Area Networks) are mostly based on CSMA/CD (carrier sense

multiple access/collision detect) generally referred to as Ethernet. Ethernet

speed has been increased from 10Mbps to 100Mbps (Fast Ethernet).

Most of today’s equipment automatically adjusts to the right data rate (auto-

sensing) and is designed to work with twisted pair (UTP) and fiber media

(100BaseT-FS).

Gigabit Ethernet builds on top of the Ethernet protocol but increases speed

tenfold over Fast Ethernet to 1000Mbps. All participants of a network are

grouped into so called LAN segments or Collision Domains. Physically all

stations within a LAN segment are connected to one common Hub.

Thus each station contends with all others for access to the network. If

multiple stations send out packets simultaneously, a collision occurs, which

corrupts the data. The more participants in a collision domain are, the more

collisions occur and the lower the data throughput of a LAN segment is.

Chapter 8: Appendix

12

4.2.2 Bridges

Fig. 2

Bridges keep local traffic within a particular LAN segment while allowing

packets destined for other segments to pass through. This process is called

filtering. To increase the throughput within a dedicated LAN segment, the

segment can be subdivided into two sub segments combined via bridges,

creating two separated collision domains and thus minimizing the probability

of collisions.

Ethernet specifies the data link (layer 2) (with the MAC sublayer) of the ISO

protocol model, while IP (Internet Protocol) and TCP (Transfer Control

Protocol) in turn specify the network (layer 3) and transport (layer 4) portions

and allow communication services between applications.

Chapter 8: Appendix

13

4.2.3 Switches

Fig. 3

Switches are based on traditional bridges’ capabilities to segment busy

networks by providing multiple dedicated connections. After decoding the

address, the switch sends the packet directly toward its destination. In Ethernet

switching, the MAC address (Media Access Control) defined in Layer 2

determines the switch port to which the packet has to go. Two nodes

connected via a full-duplex, switched path can simultaneously send and

receive packets.

Chapter 8: Appendix

14

4.2.4 Router

Fig. 4

In traditional IP networks, each router working at layer 3 calculates the

appropriate hop to the next router for each destination. Packets are forwarded

“hop-by-hop” rather than travelling along a set-up end-to-end connection. IP

delivers a “connectionless” service contrary to a transport network. When the

network is congested, packets are stored inside the router, referred to as

queuing, and forwarded according to special rules e.g. FIFO or priority. Thus

bursty sources can cause high delays in delivering time-sensitive application

traffic.

Chapter 8: Appendix

15

4.2.5 Latency

The characteristic that is most harmful to the performance of multimedia

applications is latency. This is the amount of delay that affects all types of

communications links, including those used for the public Internet and private

Intranets. Demand for immediate backbone bandwidth will inevitably force

some traffic to be queued in routers independent of the provided bandwidth.

Most increases in latency occur because network devices (mainly routers) get

overloaded and to make things even worse, large data packets are occasionally

queued ahead of shorter packets, thus introducing longer-than-average delays

and creating jitter.

Such delays as the result of queuing are variable because of the bursty nature

of IP traffic. The higher those bursts are, the longer the delays.

In fact, studies of IP networks show that traffic patterns are “self linear”, like a

fractal. Traffic still has the same burstiness, no matter how large or how small

the aggregate channel is. There is no smoothing of traffic peaks and valleys as

with the combination of large numbers of voice phone calls.

Increases in latency cause packets to arrive at their destination out of their

sequential order, especially during peak traffic periods. The packets are stored

in a buffer at the receiving device until all packets arrive to be put in the right

order. Although these delays do not affect e-mail and file transfers, which are

no real-time applications, excess latency does affect multimedia applications

arriving out of voice and video synchronization like a badly dubbed movie.

Chapter 8: Appendix

16

Fig. 5

4.3 To Overcome the IP Corset ...

The bandwidth limitations of the Internet, as well as its high latency and slow

response time, have to be overcome. Network managers have to employ

appropriate routing protocols that conserve bandwidth and/or reserve network

resources and implement flow control.

4.3.1 Tunneling

Tunneling is a method of using an inter-network (e.g. SDH backbone network

or another IP network) to transfer data e.g. frames of another protocol (i.e. IP

packets) as payloads from one network over another network. The payload is

encapsulated in a PPP (Point to Point Protocol) frame to be sent across the

inter-network.

POS (Packet over SDH/SONET) is a high speed WAN transport, that leaves

LAN traffic in its native format. It is a serial link between two access points

like any other - only much more reliable and a whole lot faster.

Chapter 8: Appendix

17

With POS, IP Traffic runs over PPP with the resulting frame embedded into

HDLC-like framing (High-level Data Link Control), just it would like any

other type of WAN- circuits-like leased lines. These link layer protocols in

turn run directly over SDH.

Fig. 6

4.3.2 Multi Link PPP

Fig. 7

Chapter 8: Appendix

18

Or “all roads lead to Rome”. MLPPP (Multi-Link PPP) co-ordinates multiple

independent links between a fixed pair of participants, providing a virtual link

with greater bandwidth than each had by itself. In order to establish

communications over a point to point link, each end of the PPP link must first

send Link Control Protocol packets to configure the data link. Once the link is

established, the source is free to send the payload encapsulated with the multi

link header.

4.3.3 Multiprotocol Label Switching

MPLS (Multiprotocol Label Switching) is a technique that brings many of the

qualities and attributes of switched networks to IP networks. It introduces the

concept of paths to the routed network. MPLS works by building engineered

paths across the core of a network. Like trains go along predefined railway

tracks from one switch/point to the next, the IP packets go along predefined

paths from one router to the next. An Ingress Router forms the starting point

of a path. Then the packets are sent encapsulated and labeled along those

predefined LSP (label switched paths) hopping from one Label Switch Router

to the next.

Fig. 8

Chapter 8: Appendix

19

The termination of the LSP is built by an Egress Router. At the Ingress

Router, the IP-packets are packed into MPLS Layer 2 frames. A so called

Shim-Header is added and the resulting packet is sent to the next Transit

Router on the LSP. At the end of the LSP the Egress Router takes the MPLS

encapsulation away and directs the packet to the next destination in the IP

network. Thus a network will seamlessly use native IP packet forwarding at

the edge, and LSP switching in the core. Labels can be used to identify traffic

that should receive special treatment to meet QoS (quality of service)

requirements.

Fig. 9

4.4 ... and Pep it up with SDH

Most carriers have made very substantial investments to build SDH/Sonet

networks, and they attract more customers and implicitly new as profitable as

possible services. IP market is the highest increasing one and after all, carriers

know that it is better to customize their network than to leave fibre buried in

the ground unused. Obviously a combination of both SDH and IP would be

the most desirable solution.

Chapter 8: Appendix

20

With the increasing rate of data streams Gigabit Ethernet comes along with,

coping with the biggest SDH containers, which only offer 155 Mbit/s,

problems are encountered, which need to be resolved. Standard SDH-

Interfaces for IP use multiple independent Virtual Containers which allow the

transport of data via established SDH-networks. Those solutions are offered

by any important IP-vendor.

Fig. 10

The biggest advantage is, that the transport of data is supported by any

established SDH-network even via third party networks (Cross Domain).

Because of the independence of the different transport channels, transport

planning remains highly flexible. Unfortunately additional processing effort in

the appropriate data machines comes along, which reduces data throughput

and subsequently leads to loss of performance on the one hand and to highly

priced equipment on the other.

Chapter 8: Appendix

21

Also SDH specification bodies acknowledged these arising problems and

directed their specification work to make SDH more convenient to data traffic.

Concatenation provides a mechanism for transporting payloads greater than

the capacity of a VC-4. Under normal circumstances the “float” of containers

within the supporting higher order virtual container is controlled by the

respective pointer mechanisms. The VC-4s are linked together by setting the

concatenation indicator at each container but the first, which takes the

respective pointer. A set of “n” in this way contiguously concatenated VC-4s

is designated as VC-4-nc. The whole structure looks like one coherent

container.

Fig. 11 The advantages are lower pricing and better performance due to higher

throughput. But contiguous concatenation requires support in each

intermediate network element, which is less supported in established SDH

networks and nearly impossible to maintain in multi carrier topologies.

Chapter 8: Appendix

22

Fig. 12

The solution to overcome these problems is virtual concatenation. The

appropriate concatenated payload is designated as VC-4-nv. According ITU-T

G.707 the virtual concatenation is identical to the contiguous concatenation

but breaks the contiguous bandwidth in individual VCs for transportation. As

a consequence the Cross Domain transport of data is ensured, and the

advantages of contiguous concatenation (lower pricing and better

performance) still remain.

Current IP routers often demand contiguous concatenation for the transport of

high bandwidth signals through SDH. Virtual Concatenation can cope with

this problem by easy conversion of VC-4-nc to individual VC-4-nv and vice

versa.

Chapter 8: Appendix

23

Fig. 13 With all these features SDH is the ideal transport network for IP wide

area connectivity.

4.5 How it should be

A successful strategy to support IP with the reliable and powerful SDH

backbone network must be aimed at the augmentation of carriers’ profit. The

flexibility to offer different services at any time to their customers under

highly optimized exploitation of the available resources will be the most

sophisticated criteria.

Chapter 8: Appendix

24

The solution to maximize network utilization and flexibility is Traffic

Engineering. Obviously, this is the basis which offers the required varied

services. It allows control of traffic streams to avoid congested data paths and

to switch it to sparsely used ones.

Thus it increases the degree of network utilization allocating transport

channels specifically after the demands of the services asked for. MPLS as

described above is a powerful aid to support Traffic Engineering. Mainly

developed for core networks, it meets all the necessary features, e.g. path

switching and QoS indication.

Fig. 14

But to become effective this needs label switch routers entirely integrated in

the network elements, working directly on the switched circuit paths (VC-n).

This enables Traffic Engineering to use the knowledge of path utilization

within the network to divert traffic avoiding congestion and overload of paths.

Chapter 8: Appendix

25

Normal router cores suffer from the lack of scalability and bad exploitation of

the network capacity. IP-over-ATM topologies are expensive and complex

caused by the additional cell tax and the co-ordination and management of two

separate networks. Scalability should be proven up to STM-16 and higher.

Traffic Engineering as a basis for offering varied services should be possible.

All these features and more are gained by an SDH backbone network with

integrated label switch routers.

So don’t plug away at IP, plug it to SDH.

Part 4

Introduction to data

Contents

1 The Tasks of Reference Models 3

1.1 The Purpose of a Reference Model 4

1.2 A Reference Model Example 4

1.3 The Basic Principles of a Reference Model 6

2 The OSI Reference Model 9

2.1 What is the Purpose of the OSI Reference Model 10

2.2 OSI Model Properties 10

2.3 Communication According to the OSI Model 14

3 The Tasks of the Individual Layers 23

3.1 The Physical Layer 24

3.2 The Data Link Layer 28

3.3 The Network Layer 32

3.4 The Transport Layer 36

3.5 The Application Layers - Layers 5 to 7 40

4 The TCP/IP Protocol Suite 43

4.1 Why does TCP/IP Exist? 44

4.2 The Structure of the TCP/IP Suite 44

5 Exercise 57

6 Solution 63

Reference Models

1

mohamed.abdelrahman

Rectangle

1 The Tasks of Reference Models

Tasks of Reference Models

Fig. 1

2

1.1 The Purpose of a Reference Model

When dealing with the Internet, we are initially faced with a multitude of terms. The myriad of protocols, hardware, transmission methods, addressing concepts and much more make it rather difficult to concentrate on the mode of operation of networks.

When dealing with data networks, an important tool is a suitable structuring of their representation, which provides an overview and makes the subject easier to tackle.

We can use so-called reference models to obtain structural descriptions of communications systems. The most important reference model is the OSI reference model, with the TCP/IP reference model also being very significant. Both models will be introduced during the first part of the course.

You may think it is unnecessary to look at the theory of the reference models, but as you have more to do with communication networks, you will find that they form a useful basis for learning. You will find that many of the principles you have encountered in one area will reappear in other contexts. The reason for this is that new developments in particular are based to a great extent on these reference models.

1.2 A Reference Model Example

The OSI reference model is a relatively detailed model to describe technical communication processes. To give you an idea of what a reference model does, we will first take a look at a reference model that outlines the communication process between people. This simple example is sufficient to demonstrate the most important principles of a reference model.

The next steps will be to provide a structured description of the overall process based on this example, and to introduce some terms that will generally appear in the context of networks.

3

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Layer Communication

Guten Morgen Bonjour

Guten Morgen

=

Good morning

Bonjour

=

Good morning

Fig. 2 Solving a comprehension problem in the layer model

4

1.3 The Basic Principles of a Reference Model

Horizontal communication:

Let us assume the above gentlemen are diplomats exchanging important diplomatic messages. As you know, the higher layer uses a sophisticated code to exchange messages: the diplomatic protocol. Generally, we can say that with the aid of a protocol, equal partners can exchange messages on any layer, but these messages are only significant for this particular layer. This principle is referred to as horizontal communication.

Vertical communication:

If we look at the actual path of a message, we can see that it is transferred horizontally on each side from top to bottom or vice versa. This is referred to as vertical communication. Actual horizontal exchange of information takes place on the lowest layer only.

Layer independence:

As you can see, the layers in the example are organized in such a way that they can be exchanged or altered independently. It would, for example, be possible for the translators to communicate comprehensibly in a new common language, without affecting the other layers. Likewise, the physical message transmission method could also be changed without affecting other layers.

NOTE It is true, however, that non-technicians would probably not have solved a comprehension problem in this way.

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Layer Communication

Layer 3

Layer 1

Layer 2

Speaker Protocol

Translator Protocol

Transmission ProtocolVe

rtic

al C

om

mu

nic

atio

n

Horizontal Communication

Fig. 3 Horizontal and vertical communication in the layer model

6

2 The OSI Reference Model

OSI Reference Model

Fig. 4 OSI Reference Model

7

2.1 What is the Purpose of the OSI Reference Model

We have just looked at a very simple reference model. Reference models used to describe technical communication systems are actually much more detailed. The most important widely used model is the OSI reference model, which was designed by the International Standards Organization (ISO) in the late 70s.

OSI stands for Open System Interconnection, which already gives some indication as to the purpose of this standardization. The OSI model standardizes communication processes to enable different (heterogeneous) computer worlds and networks to communicate with each other.

2.2 OSI Model Properties

Our introductory example illustrated that a communication process should be divided into different tasks. The objective of this is to exchange individual procedures without affecting the overall system. The OSI model applies the same principle.

Seven layers are used to describe communication processes. Clearly defined functions are allocated to each layer. Subfunctions are hereby combined in such a way as to reduce the amount of information exchanged between the layers to a minimum.

The model does not specify how these functions are implemented within the layer, i.e. which protocols or transmission methods are used. It is sufficient to define interfaces between the individual layers for the transfer of information. The OSI model clearly differentiates therefore between protocols and services.

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ApplicationApplication

PresentationPresentation

SessionSession

TransportTransport

NetworkNetwork

Data LinkData Link

PhysicalPhysical

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Logical combination of sub-

functions within one layer

Logical combination of sub-

functions within one layer

Defined interfaces between

layers:

offer services

Defined interfaces between

layers:

offer services

Implementation of a service is

not predefined

Implementation of a service is

not predefined

Seven layers for the

description of communication

processes

Seven layers for the

description of communication

processes

PropertiesProperties ofof thethe OSI ModelOSI Model

Fig. 5 OSI 7-layer model

9

Practical problems

The theoretical structuring of the OSI model works very well. You will, however, note that in reality, the protocols actually used do not always match this description 100%. Typically, this will be when you come across questions such as: ‘Which layer does this protocol belong to?’ or ‘Is this a layer 5, 6 or 7 function?’.

Some layers of the OSI model may well be subdivided again into sublayers, or one protocol could combine several layers of the model.

In these instances, you should look to the OSI model as a tool for better understanding rather than a straightjacket for a protocol. This applies mainly to older protocols. Fortunately, more recent developments have a stronger affiliation with the OSI model.

Other reference models

You will find that unfortunately, there is not only one reference model. Many standardization organizations, e.g. the ATM forum, have developed their own reference models, which sometimes differ considerably from the OSI model. The most well-known rival is the TCP/IP model, which will be discussed later. The TCP/IP model uses only four layers to describe systems.

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SessionSession

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NetworkNetwork

Data LinkData Link

PhysicalPhysical

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Application LayerApplication Layer

Host-to-Host LayerHost-to-Host Layer

Internet LayerInternet Layer

Network Access LayerNetwork Access Layer

44

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OSI ModelOSI Model TCP/IP ModelTCP/IP ModelTCP/IP Model

Fig. 6 OSI reference model and TCP/IP reference model

11

2.3 Communication According to the OSI Model

We have already outlined the differences between horizontal and vertical communication. You can draw on this knowledge to help you understand the description of the data flow in the OSI model, without having to know the tasks of the individual layers.

Remember that information is transferred between (at least) two partners. This means that the layers of the OSI model are present on both sides. Communication is between

the same layers of both sides (horizontal communication) and

the layers of one side (vertical communication).

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SessionSession

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Computer 1Computer 1



SessionSession

TransportTransport

NetworkNetwork

Data LinkData Link

PhysicalPhysical

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Computer 2Computer 2

Horizontal Communication

Vert

ical C

om

munic

ation

Fig. 7 Horizontal and vertical communication

13

2.3.1 Vertical Communication

We can use interfaces and services to describe the communication between different layers. For their part, individual layers either provide services or use the services of other layers. Generally, a layer communicates with two partners:

with its service user. This is the layer above. The service user is offered certain clearly defined services (”service primitives“),

with its service provider. This is the layer below. Now the layer concerned is the service user itself and requests ”services“ from the layer below.

A practical example:

A layer offers a layer above connection setup to the other side, this service is known as connect. The higher layer can then request connection setup from the layer below (connect.request). On the opposite side, the respective higher layer can indicate that a connection request has been made (connect.indication).

Interfaces:

Interfaces are used to exchange service requests between layers. These interfaces are also called service access points (SAP). If a layer offers a number of different services, it can also have several SAPs. In this context, we often refer to entities.

Entity

An entity is a single specific application within a layer. It can receive and send information. An end system layer can have several entities.

NOTE The corresponding entities of the same layer in different end and transit systems are referred to as peer-to-peer entities.

14

... uses the services of the

lower layer as ‘service user‘...

Each layer performs specific tasks ...

... provides services to the

upper layer as ‘service provider‘.

Service UserService User

Service ProviderService Provider

Service Access Point

(SAP)Service Access Point Service Access Point

(SAP)(SAP)

Service UserService User

Service ProviderService Provider

EntityEntity

Fig. 8 Service access points

The ‘service user‘ uses the services of the layer below

The ‘service provider‘ provides the service, e.g. connection setup

connect.requestconnectconnect..requestrequest connect.indicationconnectconnect..indicationindication

Fig. 9 An example for the service concept in the layer model

15

Data transfer across layers

When using the services of a subordinate layer, the higher layer transfers its data to the subordinate layer. The lower layer adds layer-specific information – mostly in the form of a header – and, if necessary, transfers the data further down.

We must keep this principle in mind when we look at the communication between two different partners.

Connecting two systems

When connecting two end systems, the data is first transferred downwards from layer 7 to 1 in the sender, then upwards in the reverse direction in the receiver.

In each layer, the data receive a header (cell header) and, if necessary, a trailer (information at the end of the frame). By “packing” the user data layer by layer, i.e. adding new information in each layer, the data packet increases in size.

16

Data Transfer across Layers

Send Receive

Layer n+1

Layer n

n

Headern Data

n + 1

Headern + 1 Data

n

Headern Data

n + 1

Headern + 1 Data

Fig. 10 Transfer of data across layers

Sending and Receiving Data in the Layer Model

Send Receive

1 2 3 4 5-7 Data 2

2 3 4 5-7 Data 2

3 4 5-7 Data

4 5-7 Data

5-7 Data

Data

1 2 3 4 5-7 Data 2

2 3 4 5-7 Data 2

3 4 5-7 Data

4 5-7 Data

5-7 Data

Data

Fig. 11 Sending and receiving data in the layer model

17

2.3.2 Horizontal Communication

The mechanism described above enables corresponding layers to communicate with each other. Information required on one layer can be added to the layer-specific header and transferred to the other side, for example.

This would enable a layer 3 entity to directly contact its counterpart on the other side and exchange messages. Since no direct physical link is involved, this exchange of information is also referred to as virtual communication. Only on the lowest layer do both sides communicate directly with each other.

Too abstract?

When you receive a packet (in this instance, you are part of the application layer) you are generally interested in its content and will remove its superfluous packaging. You are usually also not interested in the addresses on the packaging. However, they are of interest to the post office, your service provider. The post office could add additional information to the package that would affect the quality of service, for example, but that would only be relevant internally. (Information such as “Post Office Mail”, “Express” or “Handle with Care” can influence the success of the transmission considerably.)

If you wanted to continue this analogy with the OSI model, you would have to demand that customers bring their items to the post office for packaging.

Network performance

Information to be transmitted in addition to the user data is referred to as overhead.

The overhead is an important parameter for the evaluation of network performance. The data must be packed in the transmitter and unpacked in the destination station. These functions, as well as the additional volume of data, influence the performance of a network.

18

Horizontal / Virtual Communication in the Layer Model

Send Receive

1 2 3 4 5-7 Data 2

2 3 4 5-7 Data 2

3 4 5-7 Data

1 2 3 4 5-7 Data 2

2 3 4 5-7 Daten 2

3 4 5-7 Data

Virtual Communication:

Layer 3 can communicate with

Layer 3. The exchange of

information takes place in the

respective header.

Fig. 12 Horizontal / virtual communication in the layer model

Overhead and Payload on Layer 1:

1 2 3 4 5-7 Data 2

Overhead Payload

Fig. 13 Overhead and payload

19

3 The Tasks of the Individual Layers

Network Header

SegmentHeader

FrameTrailer

Data

SegmentHeader

Data

Data

FrameHeader

NetworkHeader

SegmentHeader

Data

0111111010101100010101101010110001

E-mail

Data

Segment

Packet

Frame(Hardware-dependent)

Bits

Fig. 14 The tasks of the individual layers

20

3.1 The Physical Layer

Services

In the service model, the task of the physical layer is to enable bit-by-bit transmission between adjacent systems. It provides the following services for layer 2:

setup, maintenance and release of physical connections

transparent bit transfer (physical transmission) between two adjacent systems.

Definitions of the physical layer

The physical layer defines all electrical and mechanical properties required for the transmission of signals. It also describes the physical properties of the transmission medium below the physical layer, as well as voltage levels, bit duration and the design of the plug contacts.

The physical layer establishes in particular

the type of physical network structure (topology),

the mechanical and electrical specifications for the use of the transmission medium,

which physical transmission, coding and timing rules are used.

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Determines physical topology

Mechanical specification of the transmission medium

Electrical specification of the transmission medium

Determines line coding

Determines other bit presentation rules

(e.g. voltage level, timing rules)

Physical Layer1

Fig. 15 Physical layer tasks

22

Too abstract?

Maybe the following questions will help you understand the purpose of the physical layer.

Which plugs are used? Which transmission properties (attenuation, side-to-side crosstalk, etc.) are required for the transmission medium? What category of cable is required?

Is the result a physical ring or a star topology?

Which modulation method is used? What are the transmission frequencies? Which line voltage levels are permissible? What is the maximum length of a pulse? To what extent may it be deformed?

Which line coding is used for the transmission of a 1 or a 0?

NOTE

The tasks of layer 1 are implemented in hardware modules within the systems (e.g. boards, buffers, controllers, network cards, …). The following network hardware is usually associated with the physical layer:

single-port or multiport repeaters

mechanical interfaces for the connection of transmission media, e.g. T-plugs

23

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Which plugs are used?

What category of cable is required?

Is the result a physical ring or a star topology?

Which modulation method is used?

Which voltage levels are permissible?

What is the maximum length of a pulse?

Which line coding is used for

the transmission of a 1 or a 0?

Questions of the

Physical Layer1

Fig. 16 Questions of the physical layer

24

3.2 The Data Link Layer

The transmission of bits in layer 1 does not involve any measures to prevent the modification or loss of individual bits, i.e. the transmission is not reliable. However, technical systems require reliable transmission in the sense that data mutilation can at least be detected. This is why the data link layer is required.

Setting up connections and reliable transmission

The task of the data link layer is to set up and release connections to other computers within a network and guarantee the error-free transmission of data frames. In the transmitting station, the data link layer divides the network layer’s data packets into frames and transmits them sequentially to the destination station once a connection has been set up. The frame contains checksums that are used for error detection or error recovery. The receiver’s data link layer analyzes the checksum, extracts the user data from frames that have been transmitted error-free and transfers them to the layer above.

NOTE

The data link layer also offers different services, such as the transfer of only one frame from one side to the other, or complete monitoring of the data transport, for example, by acknowledging each transmitted frame. Which service is used depends on the service user.

Media Access Control

The tasks of the data link layer, for shared mediums for example, also include a rule governing how individual connected stations can access the transmission medium. This rule is very dependent on the procedure used on the physical layer. In many cases, it is therefore useful to divide the data link layer into two sublayers: the MAC layer (MAC = Media Access Control), which is very much related to the hardware, and an LLC layer, which is not explicitly dependent on the hardware and which can provide the higher protocols with a uniform interface.

25

Layer 2 structures the bit sequence of layer 1:

1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0

Layer 2

HeaderData of the Higher Layers

Check-

sum

Fig. 17 The grouping of bits in frames

Possible Tasks of Layer 2: Media Access Control and Connection Setup

Data

Link

Phy.

Media Access Control Layer =

controls access to the various

transmission media

Different transmission methods and

transmission media of the physical layer

Logical Link Control Layer =

layer for setting up and releasing

reliable communications

802.3 802.5 ISO9314

CSMA/CD

MAC

Token Ring

MAC

FDDI

MAC

802.2 LLC

Fig. 18 A possible layer 2 task: media access control

26

Data link layer tasks

The data link layer is responsible for:

bit organization and structuring the information in logical groups also referred to as frames,

setting up and releasing reliable connections,

addressing the receiver system using its hardware address (e.g. MAC address),

access control for sharing the transmission medium,

forming information for error detection,

possibly error monitoring and correction; (Ethernet: collision handling),

possibly flow control.

Hardware of the data link layer

The following network hardware is usually associated with the data link layer:

bridges

switches

network interface cards and drivers.

27

Data Link Layer

Organization of individual bits in frames,

adding information for error detection

Setting up and releasing

reliable connections

Media Access control for the sharing of a

transmission medium

Addressing systems using their MAC

address (hardware-related address)

Possibly error handling

(e.g. collision handling in Ethernet)

2

Possibly flow control

Fig. 19 Data link layer tasks

Data Link Layer and

Physical Layer

Fig. 20 Data link layer and physical layer

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3.3 The Network Layer

Layers 1 and 2 are only sufficient for small local area networks

Layers 1 and 2 guarantee a reliable connection between adjacent systems. Larger networks can also be implemented using layers 1 and 2, but the maximum size of these networks is limited for reasons that will be discussed later. Such networks can be classified as LAN networks (LAN = Local Area Network).

Large networks can be set up by connecting subnetworks ...

Large networks can be set up by connecting many smaller subnetworks. As a consequence, functions and protocols are required that allow the connection of two terminals across several subnetworks. These functions are allocated to a further layer, the network layer.

... which is a task of the network layer

The network layer is therefore responsible for the transport of data across networks. As service provider, the network layer enables the service-using protocols of layer four to set up connections between two end systems (e.g. a server and a PC). Higher layers will therefore not be affected by problems such as path selection and switching. Path selection across several subnetworks is also referred to as routing. Routing is for this reason a layer 3 task.

Network addresses

The use of new address types, the network addresses, is closely associated with the task of routing. We have already encountered the hardware-related MAC addresses of layer 2. On layer 3, network addresses are introduced that are no longer dependent on the hardware and that allow the logical structuring of a network. This structuring will be discussed in the following chapters.

29

For protocols of higher layers, there is a direct

connection between both stations.

Layer 3 is also tasked with finding

a way to the destination.

This task is called „routing“.

Within a LAN,

methods of layer

one and two will

be used for data

transmission.Large networks are set up by

the interworking of small

networks. The connection

of networks is a

task of layer three.

For delivery of messages

the network address is used.

?

Fig. 21 Routing as a layer 3 task

Network Layer

Path selection based on network addresses

(e.g. IP addresses)

Multiplexing of network connectionsSegmentation and reassembly

(block formation)

Error detection and possibly correction

Possibly flow control at network level

3

Fig. 22 Network layer tasks

30

SUMMARY

The main tasks of layer 3 are as follows:

path selection across subnetworks (routing) based on network addresses (e.g. IP addresses),

segmentation and reassembly (block formation),

error detection and possibly correction,

possibly flow control at network level.

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Provision of the “best“ route

Network Layer

Fig. 23 Network layer

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3.4 The Transport Layer

What does layer 3 do?

With layers 1 to 3, it is possible to transmit data transparently from one end system (e.g. a PC) to another (e.g. a server). Generally, this transmission is not reliable, e.g. a service that tries to transport datagrams from one side to the other without providing a delivery guarantee.

How does the transport layer enhance this service?

The transport layer enhances this delivery service by introducing additional quality features. Connectionless and unreliable communication on layer 3 can, for example, be upgraded to connection-oriented protected communication.

As service provider, the transport layer terminates all transmission-related processes vis-à-vis the applications that use it. Applications can now exchange information independently of the structure and properties of the underlying communication networks.

Addressing of applications

Since many different applications use these services, the transport layer must provide a mechanism for addressing different applications.

NOTE Transport layer protocols can offer a reliable connection but need not do so. This depends on the requirements of the application. When we look at the TCP protocol later, you will learn about the mechanisms for establishing a reliable connection.

33

The transport layer provides a

connection for the applications ...

... and hides the details

of the networks below

Reliable Connection

is possible

Different protocols of the

application-oriented layers

Fig. 24 Function of a transport connection

Transport Layer

Setting up a connection between applications

on different end systems

A reliable connection is possible

Reaction to errors is possible,

e.g. by repeating transmission

Addressing applications using the port address

4

Fig. 25 Transport layer tasks

34

SUMMARY

The most important tasks of the transport layer are:

provision of a (often reliable) connection for the transport of data between applications on physically separated end systems.

the identification of a particular application by using an addressing mechanism.

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Provision of an end-to-end connection

Transport Layer

Fig. 26 Transport layer

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3.5 The Application Layers - Layers 5 to 7

Layers one to four describe the structure of a transport system that can transmit data in such a way that the technical transmission details remain hidden from the applications. This is where the network operator can draw a useful line and refer to everything above these layers as applications. This is exactly what the TCP/IP model does.

Here too, the OSI model provides a more sophisticated subdivision and differentiates between three further layers located above the transporting layers. In reality, this subdivision comes up against limiting factors. The functional scope of a relatively large number of lower layer protocols reflects the respective OSI model layer very well. However, applications often implement functions of layers five, six and seven simultaneously.

Layer 5 – the session layer

Task: Agreement between two partners concerning a session

Layer 5 describes activities required to set up a session and governs issues such as password inquiries and the like.

Layer 6 – the presentation layer

Task: Agreement of the subscribers concerning the presentation of the data to be exchanged

This layer governs matters such as character representation syntax and semantics, e.g. the conversion of ASCII characters into EBCDIC codes.

Layer 7 – the application layer

This layer contains programs that use the services of layers one to six below, such as file transfer programs or programs for e-mail or directory inquiries. These programs can again offer interfaces for other programs, for example, for a program that uses the services for file transfer.

NOTE: ASCII - EBCDIC

ASCII American Standard Code for Information Interchange – standard character representation code for computers and terminals

EBCDIC Extended Binary Coded Decimal Interchange Code

37

Application Layers - Layers 5 to 7

Session Layer

Presentation Layer

5

6

Application Layer7

Fig. 27 The application layers of the OSI model

Application Layer

Network applications, such as:

FTAM,

MAIL,

and much more...

Fig. 28 Application layer

38

• Text

• Data

ASCII

EBCDIC

Encrypted

• Sound

• Video

MIDI

MPEG

QuickTime

• Graphics

• Visual ImagesTIFF

JPEG

GIF

Presentation Layer

Fig. 29 Presentation layer

Vocabulary of two processes

Service Request

Service Reply

Session Layer

Fig. 30 Session layer

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2.1 Circuit Switching

Circuit switching is what is known from telephony. At the beginning of a data exchange, the network sets up a physical connection between the participants. This connection with a fixed reserved bandwidth is maintained until the end of the data exchange when it is released again. As a consequence of the dedicated physical connection between the calling parties, once the setup signaling has completed, the only delay for data is the propagation delay of the electrical signal. It is therefore constant without any variation.

The procedure of connection setup and release takes some time and is done by a system called signaling. This time is only worth while when the connection lasts a certain amount of time.

The advantages of circuit switching

Every connection is allocated a bandwidth of N×64 kBit/s, which cannot be used by any other connection.

There are no delay variations. For this reason, Circuit Switching is particularly well suited to real-time applications.

A very small overhead is used, the line rate can be used almost exclusively for the transmission of user data.

The disadvantages of circuit switching

During the connection setup, the bandwidth is reserved for the exclusive use of the subscriber. Following the connection setup, no changes can be made to react to changes in bandwidth requirements.

Even if the subscriber does not require his full bandwidth allocation, the free bandwidth can still not be used by other subscribers.

There are no extra security measures - no fault identification is carried out.

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The Principle of Circuit Switching

A-Subscriber

B- Subscriber

Switching Equipment

• Signaling establishes and releases a physical connection

between the two calling parties

• Bandwidth is fixed allocated for this connection

Fig. 14 The Principle of Circuit Switching

Bandwidth Allocation with Circuit Switching

Unused Bandwidth

Still available

SIEMENS

NIXDORF

SIEMENS

NIXDORF

SIEMENS

NIXDORF

SIEMENS

NIXDORF

SIEMENS

NIXDORF

SIEMENS

NIXDORF

Fig. 15 Bandwidth Allocation in Circuit Switched Networks

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2.1.1 TDM Networks

A transmission medium has much more bandwidth than a single connection could ever use. It would be a big waste to have one pair of wires, one optical fiber or one geographically separated location on the air interface per connection. To share the transmission media, people invented multiplexing, i.e. concurrent use of one physical medium by more than one connection.

Different multiplexing techniques have been developed. The best-known is probably frequency multiplexing used in radio and TV broadcasting applications (modulation).

The second technique is time multiplexing, used in modern digital technologies. The idea behind time multiplexing is to send short samples instead of a continuous signal. These samples are sent at regular time intervals (synchronous) separated by a pause that can be used to send samples of other connections. Every sample is called a timeslot and is allocated to a specific connection and provides a fixed bandwidth. This timeslot-oriented, synchronous transport technology is called

Time Division Multiplexing or TDM.

TDM is used in telephone networks and in leased line services providing different data rates.

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Time Division Multiplexed Networks

S1

S2

S3

S4

BB

t4 t3 t2 t1

t4 t3 t2 t1

S1

S2

S3

S4

AA

t4 t3 t2 t1

Fig. 16 TDM Networks

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2.1.2 Public Switched Telephone Network - PSTN

The PSTN is based on the Synchronous Transfer Mode. That means synchronous time division multiplexing (STDM) techniques are used for transmission and switching. Connections are typically set up on demand using signaling procedures.

PSTN Characteristics:

The combination of physical line and time slot defines the channel for a user information connection

The bandwidth available per channel is 64 kBit/s and is made available to the connection during the connection setup.

This channel is only used for one connection, and it is permanently allocated to the connection until connection release occurs. In the example, a time slot is reserved for both lines to and from the switching equipment.

Features POTS

Reserved channel Yes

Available bandwidth Fixed

Potential for bandwidth wastage Yes

All information units take the same route Yes

Connection setup Required

Point of possible overload Connection setup

Message transfer unit size Constant

Delay Constant

Error detection None

Error correction None

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Public Switched Telephone Networks

A-Subscriber

B- Subscriber

Switching Equipment Switching Equipment

Line

Termination

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. . . . . . . . .Multiplexer

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1 Byte G.704-Frame

7 210 31 2 1 0

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plexer

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2.2 Packet Switching

Circuit Switching statically reserves the bandwidth in advance, if no data is transmitted, the bandwidth can not be used by a different connection. This is obviously a potential waste of bandwidth.

Statistical Multiplexing

With statistical multiplexing, bandwidth is occupied, and released, according to requirements. A packet can only be transmitted when the previous packet has been transmitted in its entirety. Through statistical bandwidth, unused bandwidth can be allocated to other connections as required. Information is only transmitted, if sufficient bandwidth is available on the line, otherwise transmission is paused. The line rate is used for the transmission of a packet.

Because packets are of varying lengths, there are varying delays for other packets. Short packets are more likely to be delivered on time than long packets. Variable delays are critical for real-time applications like voice.

A header is appended to every packet indicating the destination, so that the network can route the packet to a given point. At every network node, the packet is stored and forwarded according to its destination address. Packet switching can be connection less, (e.g. IP) or connection oriented (e.g. Frame Relay). In the later case a so called virtual connection (VC) is set up.

The advantages of packet switching

No permanent bandwidth is allocated, so unused bandwidth can be used by other connections.

Transmission reliability is possible through error detection and error correction.

Connectionless transmission is possible.

The disadvantages of packet switching

The packets do not arrive at the subscriber at regular intervals. There are delays, which depend, amongst other things, on the load. For this reason, real-time applications are typically not transmitted.

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Packet Switching

Unused

Bandwidth

Virtual Circuit

• Dynamic bandwidth allocation among VCs with „Statistical Multiplexing“

• Variable Delay depending upon:

– Load of the line

– Packet length

– Line speed

Synchronization

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Part 5

GSM Introduction

Sub ‐ Sections GSM Introduction

1 Introduction Pages (1-44)

2 Transmission Principles Pages (1-37)

3 GSM PLMN Pages (1-32)

4 Procedures Pages (1-38)

5 Radio Interface Pages (1-31)



Introduction

Introduction

Contents

2History 1 15 GSM 2 27Current Situation, Market & Trends 3

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Introduction Siemens

1 History

Introduction

History

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Siemens Introduction

History of Mobile Communications

“Mobile Communication” is much older than many people think. There have beendiverse "acoustic and optic means of remote information transfer" in the most variedcultures and stages of civilization on all populated continents. The range ofinformation transfer was very limited and the quality of the messages was affected byouter conditions such as the weather. In order to increase the range of informationtransfer in these times, transit stations were in part systematically constructed.

Beginnings of Electronic Communications

Telegraph: S.F.B. Morse: 1843 First experimental telegraph line: Washington -Baltimore

Telephone: Phillip Reis 1861: First speech transmission by cable / A. G. Bell: 1876World Exhibition, Philadelphia

At first electronic communications was possible only via wire i.e. by means of fixed(immobile) connections, forerunners of today's Fixed Network Connections. Initiallyan operator ("switchboard girl") was needed to establish these fixed physicalconnections for the caller manually at the central office. The first automaticexchanges were first put into service in the mid-1920s.

Radio Communications

Radio connections were first used for Wireless Communications in the late 19thcentury; information was sent via "ether".

1873: J.C. Maxwell - electromagnetic wave theory

1887: H. Hertz - experimental proof of the existence of electromagnetic waves

1895: A. Popow - first receiver with antenna for weather reports

1895: G. M. Marconi - first wireless transmission using spark inductor generatedHF waves (Morse code)

1897: “Marconi Wireless Telegraphy Company" founded

1901: First transatlantic transmission (Marconi)

1903: "Deutschen Telefunken GmbH" founded by AEG and Siemens & Halske

1906: First speech & sound transmission (Lorenz AG / Deutsche TelefunkenGmbH)

1909: First radio broadcast (New York, Caruso)

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The beginnings: "archaic mobile communication"

• visual transmission (smoke/light signals,...)

• audible transmission (drums, horns,...)

Electronic

communication:

"terrestrial network"

• Telegraph 1st telegraph line 1843

Washington - Baltimore

• Telephone

P. Reis 1861

A.G. Bell 1876 World Exhibition Philadelphia

Radio transmission:1873 Maxwell‘s theory of electromagn. waves

1887 H. Hertz: experimental proof1895 Marconi: 1st wireless transmission1901 1st transatlantic transmission

1903 Dt. Telefunken GmbH: AEG, Siemens& Halske1906 1st speech and sound transmission1909 1st radio broadcast

1917 1st mobile transmission: radio station - train

History of Mobile Communications

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Connection Types

There are two principles for radio connections:

Simplex Connection

Simplex connections are a "one-way street" for communication in the form of (mostlyfixed) transmitters and mobile receivers. This has been realized as e.g. (broadcast)radio and television. But simplex connections are also used for direct communicationexchange i.e. two-way communication using stations which can be used both as atransmitter and a receiver (e.g. walkie-talkies). However the equipment (transmitting /receiving stations) cannot transmit and receive simultaneously. The call cycles or callintervals are determined by prior agreement or personal code words ("over").

Duplex Connections

Duplex connections signify two-way communication. Users can transmit and receivemessages simultaneously. An example of an early duplex connection is radiotelegraphy.

Simplex Connection:transmit or receive

Duplex Connection:simultaneous

transmission and reception

Over

Fig. 3

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Single Cell Systems

The first Mobile Telephone Service to offer duplex connections comparable to fixednetwork based telephone services started in 1946 as a car phone service in St.Louis, Missouri. Comparable mobile telephone services appeared in post-war Europesome years later.

Problems in early mobile (car) telephone services (late 1940s/early 1950s):

An operator was needed to connect calls within the wireless network.

The equipment required was extremely heavy, bulky (therefore only feasible as acar phone service) and expensive.

The service range was limited to the area that could be covered by a singletransmitting or receiving station (single cell system).

The HF frequency range available was (is) very limited; it had to be (and still hasto be) distributed among competitors (e.g. the military, radio, and television).

The result was limited capacity, rapid market saturation, high equipment costs andlow service quality.

• Car telephone service

• Since the late 40‘s

• Low service and speech quality

• Heavy, bulky and expensive equipment

• Small coverage area

• No handover

• Manual exchange

• Low capacity

First Mobile

Services:

Single Cell Systems:

Fig. 4

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Innovations in Mobile Radio Communications

Technical Innovations / Equipment

Fast development of new technologies such as semiconductor technology, diodes,transistors, integrated circuitry, microprocessors,...

automatic switching

reduction of hardware costs

reduction of size and weight of equipment (in the 1950s/1960s a car phone tookup half of a car trunk; 1988: introduction of the mobile phone)

but:

very limited telephone network capacity.

During the 1970s large-scale integrated, electronic applications and the developmentof microprocessors made the configuration of more complex systems possible. Oneresult of this was the development of single-cell transmitter systems with multiplereceiving stations. This made it possible to extend the range of the supply area, i.e.the operational range of the subscriber because the mobile station's transmitterpower limits the size of the cell in Single Cell Systems. However no increase incapacity resulted from this.

Cellular Mobile Radio Systems

The breakthrough in capacity, which resulted in a significant increase in the numberof subscribers, was achieved with the introduction of the Cellular Radio System in thelate 1970s/early 1980s. The coverage of the supply area of a mobile communicationoperator involves many radio cells with cellular radio systems, in which theaforementioned limitation of the available HF frequency range is neatly circumventedthrough the repeated use of the HF channels.

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Quantum Leap in Mobile Communications:

Single Cell Systems Cellular Systems

radius

r

re-use distance

r

Single Cell

System

Cellular

System

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First Generation (1G) Cellular Mobile Radio Systems

Information transmission of first generation cellular mobile radio system takes placevia analogue radio interface. These systems were tested in many countries in the endof the 70s.

In 1979, mobile services were introduced for commercial operation; in the USA,AMPS (Advanced Mobile Phone Service), and in Japan, NTT-MTS (NipponTelegraph & Telephone Co.).

In the early 80s, the NMT (Nordic Mobile Telephone) was introduced in Scandinavia,in 1985 TACS (Total Access Communication System) was introduced in England andthe C450 System in Germany.

First Generation Cellular Mobile Radio Systems

Country System Frequency range[MHz]

Introduced

in year

USA AMPS 800 1979

Japan NTT-MTS 800 1979

Sweden, Norway,Finland, Denmark

NMT 450, 900 1981 - 86

Great Britain TACS 900 1985

Germany C450 450 1985

France Radiocom2000

NMT

450

900

1985

1989

Italy RTMS

TACS

450

900

1985

1990

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Second Generation (2G) Cellular Mobile Radio Systems

A further and very significant innovation in mobile radio communications took placewith the introduction of the second generation cellular mobile radio system (e.g.GSM) in the early 90s. Transmission via radio interface is now digital. Along with asignificant improvement of transmission quality and expansion of services, there hasbeen a considerable increase in capacity. The increase in subscribers led to moreconvenient, lighter and less expensive equipment with a wide range of possibilitiesfor use.

Portable Mobile Equipment

Mobile phones were first introduced in 1988. The weight of the equipment decreasedfrom 1 kg to less than a

100 g within few years. At the same time, mobility clearly improved despitedecreasing weight owing to improvements in rechargeable batteries. Standby timesof more than 5 days can be achieved.

2nd Quantum Leap:

Analog (1st Generation) Digital (2nd Generation)

Different Generations of Mobile Stations

Second generation

GSM mobile telephones Second generationGSM mobile telephones

Digital GSM technology.Terminal devices are handier

and have greater battery capacity.

Digital GSM technology.Terminal devices were less

bulky, but still too heavy(battery capacity problems).

Analog technology.Terminal devices were

bulky and heavy.

First generation

mobile telephones

for fixed vehicle installation and

analog mobile telephones

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Example: Mobile Subscriber in Germany

Since the early 50s there have been several regional networks at 30, 80, 100 MHz.They were allocated only to public authorities and organizations with security tasks.The regional networks (DBP) were combined in the so-called A-network in 1958 al-lowing private use for the first time.

A-network: in operation: 1958 - 1977; frequency range: 156 - 174 MHz; in thebeginning 16, later 37 radio carrier; analogue transmission, manual switching; max.11,000 users (1971); closed in 1977; its frequencies were transferred to the B-network.

B-network: in operation: 1972 - 1994; frequency range: 146 - 164 MHz; from 1977 to174 MHz (from A-network); in the beginning 38, later 75 radio carrier; analoguetransmission, automatic switching; max. 27,000 users (1986); problem: max.capacity, no further channels; closed in 1994.

C-network (C450): in operation: 1985 - 2000; frequency range: 451.3 - 455.74 MHz& 461.3 - 465.74 MHz; 222/287 radio charier; system technology: Siemens. TheC450 system was the first German cellular system and led to an enormous increaseof subscribers (max. 850,000 users). The C-network was similar in structure tomodern digital networks.

D-networks (GSM900): Introduction in 1992 (D1 & D2); 900 MHz frequency range (+minor extensions in the 1800 MHz range from 1999 on; system technology partlyfrom Siemens (D900).

E-networks (GSM1800): Introduction in 1994 (Eplus) and 1998 (E2); 1800 MHzfrequency range; System technology partly from Siemens (D1800).

The digital D and E networks, being GSM900 / GSM1800 networks, led to a rapidand steady increase of the number of subscribers in Germany. In 12/2000, a total of46 million mobile subscribers were registered in the 4 networks, D1, D2, Eplus & E2.

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0,01

0,1

1

10

100

Su

bs

cri

be

r [M

.]

1978

1980

1982

1984

1986

1988

1990

1992

1994

1996

1998

2000

Year

Germany

Subscriber trends (Example): Germany 1978 - 2000

B-n

etw

ork

in

tro

du

ctio

n

C-n

etw

ork

in

tro

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n GS

M (

D1,

D2)

intr

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GS

M (

Ep

lus)

intr

odu

ction

GS

M (

E2

) intr

odu

ction

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Limits of the First Generation Mobile Radio Systems

1. Capacity: The capacity limits of analogue technology are reached quickly evenwith cellular networks. The demand increases with the offer and the sinkingprices. A number of 850,000 subscribers, i.e. the maximum capacity of theanalogue C-network, corresponds to less than 7 % of the mobile subscribers in1998 (only 6 years after introducing digital networks). The capacity of digitalnetworks has not yet been exhausted.

2. Quality: A second problem was the often inadequate transmission quality of theanalogue systems, which increased with the distance of the mobile subscriber. Adetailed description and discussion of the problems regarding the transmissionquality or the disadvantages of the analogue system in comparison to digital onecan be found in the next chapter.

3. Incompatibility: One or more analogue networks on frequency bands 450/900MHz existed in most European states in the late 1980s. Every one of thesenetworks formed a mobile communication island since the individual standards ofthese networks were incompatible in most cases (or still are, as far as they stillexist); they prevented mobile phone traffic across borders (InternationalRoaming). Europe thus looked liked a rag rug of incompatible systems.

The limits of existing analogue systems

1. Capacity: the number of potential mobile phone customers is larger than theexpected capacity of analogue systems,

2. Quality: insufficient transmission quality with increasing distance between themobile station and the base station,

3. Incompatibility: between different national standards,

were already recognized since the early 80s and were discussed on an internationalEuropean level. The need to develop a new, standard cellular system for Europe wasacknowledged.

The GSM Standard was developed for this purpose.

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Capacity Quality Incompatibility

European mobile

communication marketearly 90‘s

1G Limitations

Fig. 9

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2 GSM

Introduction

GSMGlobal System for

Mobile Communications

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The GSM History

The foundation for the GSM Standard was laid already in 1978, four years before thename GSM was established. In 1978 the CEPT reserved a frequency range round900 MHz for mobile communications in Europe. The limits of analog mobilecommunications in Europe were recognizable in the early 80s. At that time the firstanalog cellular networks were just beginning their operation and were still far fromtheir maximum capacity. Despite this a group of experts was formed to establish thelonger-term challenges of mobile communications and to develop a new bindinginternational standard for digital mobile communications in Europe. Thus the GSMStandard became undoubtedly one of the most successful European products of thepast decades; its sphere of influence is extended far beyond the originally plannedEuropean scope.

Milestones of the GSM Standard

1982: The CEPT forms a team of experts, the Group Special Mobile (GSM) withthe purpose of developing a binding international standard for mobilecommunications in Europe.

1984 – 86: Various technical possibilities are compared in order to achieve anoptimal utilization of the predefined frequency ranges.

1986: A permanent core of experts is employed.

1987: Main transmission principles are selected; 13 countries agree in the MoU(Memorandum of Understanding) to start GSM networks until 1991.

1988: The ETSI (European Telecommunication Standards Institute) is founded;most of the standardizing activities of the CEPT, including GSM, are assumed bythis new body. Along with state-owned operators, industry, private networkoperators and consumer groups participate in the ETSI, too.

1989: GSM is renamed from "Group Special Mobile" to "Global System for MobileCommunications".

1990: GSM900 Standard (Phase 1) is adopted. DCS1800 Standard (Phase 1) isdeveloped as first GSM adaptation. The first GSM systems are in test operation.

1992: Commercial introduction of many large GSM900 networks.

1993: Work begins on updating the GSM900/DCS1800 standards: GSM Phase 2.

1995: GSM-R (Railway): The ETSI reserves further frequency range for a railwaynetworks; first test projects are started. GSM Phase 2 work is completed.

1996: Worldwide success of GSM Standard; used in more than 50 countries.PCS1900 (Public Cellular Systems) as further GSM adaptation in the USA.

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GSM Milestones

1978 CEPT reserves 2 x 25 MHz in 900 MHz range

1982 CEPT founds "Groupe Special Mobile" GSM

1984-86 Comparison of technical possibilitiesGoals: - free roaming

- international accessibility under 1 number (international roaming)- large network capacity (bandwidth efficiency)

- flexibility ISDN- broad service offering- security mechanisms

1986 Core of experts meets continuously

1987 Selection of central transmission techniques

Memorandum of Understanding: MoU

1988 ETSI founded

1989 GSM Global System for Mobile Communication

1990 GSM900 Standard (phase 1)

1991 DCS1800 adaptation

Trials / "friendly user" operation

1992 Start of commercial operation

1993 Beginning of work on phase 2

1995 Completion of work on phase 2 (GSM900/DCS1800)

Reservation of GSM-R frequencies (ETSI)

1996 PCS1900 adaptation (USA)

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1997: GSM Phase 2+ Annual Release ‘96: CAMEL Stage 1, ASCI for GSM-R.DCS1800 / PCS1900 are renamed to GSM1800 / GSM1900. Dual bandequipment for GSM900 / GSM1800; 10 years of MoU: 109 countries; 239operators; 44 million GSM subscribers; 28 % share of the world market.

1998: Phase 2+ Annual Release ‘97: HSCSD, GPRS Stage 1, CAMEL Stage 2,...08/98: 100 million GSM subscribers in 120 countries; 35 % share of the worldmarket; GSM is quasi world standard. GSM-R networks in operation. World-wideservicing through co-operation with mobile satellite systems (IRIDIUM).

1999: Phase 2+ Annual Release '98; 250 million subscriber; 130 countries

2000: Phase 2+ Annual Release '99: GPRS Stage 2, CAMEL Stage 3, EDGE,Virtual Home Environment VHE, Adaptive Multirate speech AMR,...GSM Rel. '99services identical to UMTS Rel. '99 (first UMTS release); 410 million subscriber;161 countries; approx. 60% of world-market

1997 Phase 2+: Annual Release `96

DCS1800 / PCS1900 GSM1800 / GSM1900

Dual-band devices

GSM: practical world standard (109 countries/regions; 28 % market share)

1998 Phase 2+: Annual Release `97: GPRS, CAMEL,....

First GSM-R networks

World-wide accessibility using dual mode GSM/IRIDIUM

35 % of world market

1999 Phase 2+: Annual Release ‘98

250 M. subscriber, 130 countries

2000 Phase 2+: Annual Release ‘99: AMR, VHE,... identical to UMTS Rel. ‘99

60% of world market; 410 M. subscriber, 161 countries

GSM Milestones

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The GSM Technical Guideline

Objective (1982): Development of a unified, international standard for mobilecommunications. Guideline from the start:2 x 25 MHz frequency bands at 900 MHzare reserved by the CEPT for mobile communications in Europe in 1978. 1982:Roaming; the user can change location, keep the connection and be reached in theentire range of a PLMN and in the entire GSM range (International Roaming) as longas roaming agreements have been made. One user - one number; the subscribercan be reached at a single personal number in the entire GSM range, i.e. in variouscountries and PLMNs.

Late objectives: Maximum flexibility to other services, e.g. ISDN (Integrated ServicesDigital Network; 1984) Vast service offers, i.e. technical possibilities of the PSTN/ ISDN and special features of mobile communications Safeguarding frominterception and subscriber license fraud; data protection.

The GSM Recommendations

The GSM Standard is a consistent and open standard for cellular mobilecommunication systems established by the ETSI. All aspects of the realization of theGSM Standard have been established in now more than 150 recommendations(technical specifications). Subsystems, network components, interfaces, signaling,tests and maintenance aspects etc. are described. This allows a harmoniousinteraction of all elements of a mobile communication network designated as PLMN(Public Land Mobile Network). At the same time the Recommendations are flexibleenough for the different realizations of various vendors. The Recommendations areorganized into 12 series according to different aspects. This structure reflects thestructure of the PLMN system and its interfaces.

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GSM Recommendation

MSC

PSTN

ISDN BSS MS

Series 01: General

Series 02: Service Aspects

Series 03: Network Aspects

Register

Series 04:

MS/BS Interface

& Protocols

Series 05:

Um Radio

Transmission

Series 06:

Speech Coding

Series 067:

Terminal

Adaptors for MS

Series 08:

MSC-BSS Interface

Series 09:

Network Interworking

Series 10:

Service Interworking

Series 11: Equipment & Type Approval Specifications

Series 12: Operation & Maintenance

12 Series; each max. 100 Rec.:e.g. GSM Rec. 08.07

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The Evolutionary Concept

The GSM Standard consists of multiple of recommendations. They are organized byvarious aspects and already comprised 5230 pages when the first phase wasadopted in 1990. It was originally planned to comprise every specification in the GSMStandard (with the exception of “half rate speech") from the start, i.e. when thestandard was adopted. In 1988 it was recognized that not all of the planned servicescould be specified in the expected time frame. This led to the important decision toleave the GSM Standard incomplete and to leave space for further modifications andtechnical developments. This evolutionary concept secures for GSM the possibility ofpermanently adapting to the requirements of the market and thus ensures of notbecoming old-fashioned within a couple of years owing to the extremely fastdevelopment in this market sector.

GSM Phase 1

The Phase 1 standardization was closed in 1990 for GSM900 and in 1991 forGSM1800. The implementation of GSM systems Phase 1 comprises all of the mostimportant prerequisites for digital information transmission. Speech transmission is ofthe greatest importance here. Data transmission is also defined by data transmissionrates of 0.3 to 9.6 kbit/s. GSM Phase 1 comprises only a few supplementary servicessuch as call forwarding and barring.

GSM Phase 2

The Phase 2 standardization work started shortly after completion of Phase 1 andwas closed in 1995. In Phase 2 Supplementary Services comparable to ISDN(Integrated Services Digital Network) were included in the standard. Technicalimprovements have been specified, e.g. the Half Rate Speech. In Phase 2, thedecision on future downward-compatibility with older versions is of high importance.

GSM Phase 2+

GSM Phase 2+ refers to a “smooth” transition in contrast to Phase 2. A new completeupdate of the GSM Standard is not planned. Individual topics are discussedseparately and the update is added to the GSM standard in Annual Releases. Maintopics are new Supplementary Services as the ASCI services (Advanced SpeechCall Items). Furthermore, the IN feature Customized Applications for Mobile networkEnhanced Logic CAMEL and Virtual Home Environment VHE are very important.Especially the introduction of features to achieve higher data rates, i.e. HSCSD (HighSpeed Circuit Switched Data), GPRS (General Packet Radio Service) and EDGE(Enhanced Data rates for the GSM Evolution) has received much attention. GSMPhase 2+ thus paves the way to 3G (UMTS).

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Phase 1Phase 2

Phase 1

Phase 2+

Phase 2

Phase 1

Services

Year1991 1995 1997

Full Rate Speech (FR),

Standard services

Data: max. 9.6 kbit/s

New services e.g.

MTPy, CUG, AoC;

Half Rate Speech (HR)

New services e.g.

ASCI, SOR, UUS

EFR;

IN: CAMEL

Data: HSCSD, GPRS,

EDGE (> 100 kbit/s)

Annual Releases !

GSM: Evolutionary Concept

Downward compatibility

MTPy:

CUG:

AoC:

ASCI:

SOR:

UUS:

EFR:

IN:

CAMEL:

HSCSD:

GPRS:

EDGE:

Multiparty Service

Closed User Group

Advice of Charge

Advanced Speech Call Items

Support of Optimal Routing

User to User Signalling

Enhanced Full Rate Speech

Intelligent Network

Customized Applications for

Mobile network Enhanced Logic

High Speed Circuit Switched Data

General Packet Radio Service

Enhanced Data Rates for the GSM

Evolution

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Adaptations of the GSM Standard

The GSM adaptations GSM900, GSM1800, GSM1900, GSM-R and GSM400 differ inthe frequency ranges used and the resulting different technical implementations.

GSM900 (GSM, E-GSM)

Originally 2 x 25 MHz in the frequency range around 900 MHz (890 - 915; 935 - 960MHz) were provided for mobile communication applications. In an extension of thisrange, called E-GSM (Extended GSM) these ranges will be increased to 2 x 35 MHz(880 - 915; 925 - 960 MHz) on a national level when further operation licenses expire.

GSM1800 (DCS1800)

As an adaptation of the GSM900 Standard the DCS1800 Standard (Digital CellularSystem) was introduced in 1991. The DCS1800 was a British initiative with theintention of opening mobile communications to all sections of population as a “massmarket”, especially in urban areas. The GSM1800 has 2 x 75 MHz in the frequencyrange around 1800 MHz (1710 - 1785; 1805 - 1880 MHz). In 1997 the designationDCS1800 was changed to GSM1800 in order to clarify the common standard.

GSM1900 (PCS1900)

The PCS1900 Standard (Public Cellular System) is the American branch of the GSMStandard since 1995/96 in the frequency range around 1900 MHz. The frequencyrange available between 1850 - 1910; 1930 - 1990 MHz in the USA was split up in1995 and auctioned off to different net-work operators. In 1997 the PCS1900 wasrenamed GSM1900 in order to clarify the common standard.

GSM-R (Railway)

For mobile communication of railway operators 2 x 4 MHz in the frequency range of876 – 880 MHz & 921 – 925 MHz have been reserved.

GSM400

With Rel. '99 the frequency ranges between 450.4 – 457.6 MHz & 460.4 – 467.6 MHzrespectively the ranges (of former 1G systems) between 478.8 – 486 MHz & 488.8 –496 MHz are foreseen for GSM400. The GSM400 frequency range enables largearea cells for rural environment.

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876 880

890

GSM

900

915 921 925

935

960 1710 1785 1805 1850

1880

1910 1930 1990[MHz] [MHz]

GSM

900

E-GSM E-GSM

GSM

1800

GSM

1800

GSM

1900

GSM-R GSM - Adaptations

GSM

1900

Frequency Range[MHZ]

Useable HFchannels

Application Area

GSM400 450.4 – 457.6 / 460.4 – 467.6

478.8 – 486 / 488.8 - 496

35 rural environment

GSM900E-GSM

890 - 915 / 935 - 960880 - 915 / 925 - 960

124174

Worldwide exceptAmerica

GSM1800 1710 - 1785 / 1805 - 1880 374 Worldwide exceptAmerica

GSM1900 1850 - 1910 /1930 - 1990 299 America

GSM-R 876 - 880 / 921 - 925 19 Railway systems

Fig. 15

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The GSM-PLMN

In the GSM System there must be a distinction between network operator, provider oftelecommunication services, supplier of terminal equipment and manufacturer ofnetwork components. Especially the sale of telecommunication services and terminalequipment differs from the conventional fixed network and mobile communicationnetwork of the first generation, in which state-owned network operators, serviceproviders and equipment suppliers usually form a monopoly. In GSM the actualnetwork operator often transfers services to private providers who supply theservices to the mobile subscribers under different conditions. With the wide range ofproducts there is also great competition in the field of mobile equipment as well as ofmobile communication network components which should force further technicaldevelopment and keep the prices down.

PLMN - Public Land Mobile Network

A PLMN is a terrestrial mobile communication network set up and run by public andprivate operators. It is used to provide public mobile communication services.

General Objectives of a GSM-PLMN (with respect to service aspects):

a) Provision of a wide range of speech and non-speech services andcompatibility to those services offered in fixed telecommunication networkssuch as PSTN, ISDN and PDN;

b) Additional provision of specific services for mobile access environment;

c) Compatible access for mobile subscribers in all countries where the GSMSystem is operated;

d) Provision of roaming (roaming agreement) and automatic updating;

e) Location registration of mobile subscribers in these countries;

f) Provision of sufficient quality of service;

g) Provision of services with a wide range of mobile stations, e.g. permanently in-stalled in vehicles, so-called portables and hand stations (mobile phones).

General Objectives of a GSM-PLMN (with respect to performance aspects):

a) Guarantee of a high spectrum efficiency;

b) Provision of a system concept which will lead to attractive costs regardinginfra-structure and mobile equipment

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GSM-PLMN(Public Land Mobile Network)

Example:

Germany

Competition concept:different network operators,

providers and manufacturers

D1Telekom

D2Mannesmann

Eplus

E2Viag Intercom

Fig. 16

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3 Current Situation, Market & Trends

0,01

0,1

1

10

100

1000

1980

1982

1984

1986

1988

1990

1992

1994

1996

1998

2000

Introduction

Current Situation,

Market & Trends

Fig. 17

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Overview: Systems/Standards

At the time there is a wide spectrum of mobile communication systems of the first andsecond generation along with the GSM Standard and its adaptations. Importantexamples include:

Paging Systems

Cordless Telephone

Wireless Local Loop

Private Mobile Radio

Cellular Mobile Systems

Mobile Satellite Systems

These different systems differ in:

Target groups

Services offered

Prices

Coverage

Degree of mobility

Technical principles / realization

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analogue cordless telephone systems

e.g. CT1, CT1+

digitalpaging systems

e.g. ERMES

analoguepaging systems

e.g. Citycall

Cordless

telephone booth

digital cordless telephone systems

e.g. DECT, PACS, PHP

analoguePrivate Mobile Radio

PMR

Wireless Local Loop

WLL

digitalPMR

e.g. TETRA

digital cellular systems

e.g. GSM, D-AMPS,

PDC, IS-95

digital satellite systemse.g. IRIDIUM, ICO,

Globalstar

analoguecellular systems

e.g. C450, NMT, AMPS

analoguesatellite systemse.g. INMARSAT

Current

Mobile

Communication

Systems

Differences:• target groups

• services offered

• prices

• coverage

• degree of mobility

• transmission technique

• ...

1G 2G

Fig. 18

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1G Systems

C450: closed 12/2000

TACS (Total Access Communications System): closed 2001.

NMT (Nordic Mobile Telephone): closed 2001.

AMPS (Advanced Mobile Phone Service): The AMPS system was introduced in 1979in the USA. The system, operated in the frequency range of 800 MHz, was the mostsuccessful mobile radio system in the world until 1997. It still has an increasingnumber of subscribers, because of its large coverage in the USA. 12/2000, more than75 million AMPS subscribers were registered.

2G Systems

GSM (Global System for Mobile Communications): The GSM Standard wasadopted as the first digital mobile communication standard, as planned since theearly 80s. Commercial operation started in 1992. This led to the world-wide use ofGSM net-works, which were originally planned for the European system, in more than120 countries and regions. GSM uses a hybrid solution of FDMA and TDMA as anaccess technique. GSM used currently 900 / 1800 /1900 frequency ranges.

D-AMPS (Digital Advanced Mobile Phone System): The D-AMPS was conceivedas a supplementary system to the successful analogue AMPS in the USA andCanada. The commercial start was 1991/92. D-AMPS as IS-136 standard is basedon a combined FDMA/TDMA access technique. It shares the 800 MHz range withAMPS (824 - 849; 869 - 894 MHz). It expanded to the 1900 MHz range in 1995.Multimode / multiband equipment is used for AMPS/D-AMPS.

PDC (Personal Digital Cellular): With the influence of D-AMPS, PDC (originallycalled JDC - Japanese Digital Cellular) was standardized for the Japanese market.The commercial start was 1993/94. A combined FDMA/TDMA procedure, similarly tothe D-AMPS, is used as an access procedure. Mobile stations transmit at the higherfrequency with PDC, in contrast to all other systems. Frequencies around 900 MHz

(810 - 826; 940 - 956 MHz) & 1500 MHz (1429 - 1453; 1477 - 1501 MHz) are used.

IS-95 CDMA IS-95 CDMA was developed in the early 90s based on CDMA spreadspectrum digital technology and was declared IS-95 standard in 1993. Thecommercial start was 1995/96. IS-95 CDMA networks are emerging world-wide withemphasis on North America and Eastern Asia. Frequencies in the 800 MHz and 1900MHz range are used world-wide, and also in the 1700 MHz range in Korea.

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Cellular Systems

First generation:C450

NMT - Nordic Mobile Telephone

TACS - Total Access Communications SystemAMPS - Advanced Mobile Phone System

Second generation:

GSM D-AMPS PDC IS-95

Start 1992 1991/92 1993/94 1995

Coverage worldwide especiallyUSA, Canada

Japan especially USA,Canada, EasternAsia

Frequency

ranges [MHz]

900 / 1800 /1900 (America)

800 / 1900 900 / 1500 800 / 1700 (Korea) /1900

Multiple

Access

TDMA / FDMA TDMA / FDMA TDMA / FDMA CDMA

Speech [kbit/s] 13 / 5.6 7.95 6.7 9.4 / 13

Data (max.)

[kbit/s]

9.6(n•14.4; n = 1...8)

4.8 4.8 9.6 / 14.4

Subscribers

(02/2001)

~ 410 million ~ 35 million + 75 million (AMPS)

~ 55 million ~ 85 million

Fig. 19

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Mobile Satellite Systems MSS

Large areas of the earth's surface can not be covered by fixed or mobile networks.Mobile Satellite Systems MSS are offered for supplying scarcely populated regionsand areas with weak infrastructure. Satellite supported mobile communicationsystems are useful for high-sea ship transport, for catastrophe regions, and foremergency supply.

Satellite systems can be distinguished with respect to their orbits:

GEostationary Orbit - GEO, with approx. 36,000 km altitude;

High Elliptic Orbit - HEO;

Medium Earth Orbital - MEO, from 10,000 - 20,000 km;

Low Earth Orbital - LEO, from 700 - 1,500 km.

1G MSS

MARISAT (Maritime Satellite): MARISAT went into operation in 1976 as the firstmobile satellite system, initiated by the USA.

INMARSAT (International Maritime Satellite Organization): INMARSAT is taking adominant role in 1G MSS. Founded in 1979, it is used by more than 100 membershipcountries. The four INMARSAT (operation) satellites are in a geostationary orbit(about 36,000 km altitude). With the exception of a the pole caps, a globaltransmission to the world is achievable. Digital transmission is via INMARSATsatellites since 1995., i.e. INMARSAT has turned over to a 2G MSS system

2G MSS

Digital information transmission and a larger number of satellites in lower orbits (LEOand MEO satellites) allow considerably higher capacity. Several services similar tothose of GSM should be possible. A problem of the 2G systems is the comparablehigh price and fast extension of 2G terrestrial networks

Iridium (closed 2000)

Globalstar

ICO

Ellipso

ORBCOMM

Teledesic

Skybridge

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Supply to/ in case of:

- inaccessible, underpopulated areas

- poor infrastructure- high seas- catastrophe areas

- failure of other supplies

Supply to/ in case of:Supply to/ in case of:

- inaccessible, underpopulated areas- poor infrastructure

- high seas- catastrophe areas- failure of other supplies

GEOGEostationary Orbit

10,000- 20,000 km

700- 1,500 km

MEO MediumEarth Orbit

approx.36,000 km

LEOLow Earth Orbit

Mobile Satellite Systems MSS

HEOHigh Elliptic

Orbit

1G:

MARISAT (USA) since 1976

INMARSAT (International Maritime

Satellite Organisation):• since 1979; > 80 member countries• 4 GEO satellites;• global access

2G:• Iridium, ICO, Globalstar

• private MSS operator• speech- & low data rate services

Earth

Fig. 20

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The Mobile Market: Subscriber Trends 1980 - 2000

Before the introduction of first generation of cellular mobile communication systems,the mobile communication market was unimportant. One-cell systems had only a fewthousand subscribers and slow annual growth rates in Europe, North America, andJapan. Until the introduction of the first cellular systems in 1979 (AMPS: USA, NTT-MTS: Japan) fewer than a million subscribers were registered worldwide.

The introduction of the first generation (analog) cellular mobile communicationsystems led to a quantum leap on the mobile communication market. There wereannual growth rates of 10 to more than 50 %. In the early nineties, there were morethan a million subscribers registered in both the USA (AMPS) and Great Britain(TACS) each. Several hundreds of thousands of subscribers were registered in othercountries with systems such as NMT, C450, NTT-MTS. The number of worldwidesub-scribers exceeded 10 million in 1990. Simultaneously the limits of analoguecellular systems were apparent in many countries owing to capacity problems,especially in densely populated urban regions.

The introduction of GSM as the first mobile communication standard of the second(digital) generation allowed an improved transmission quality, a larger offer ofservice, various technical / organizational improvements, and a considerably moreefficient use of radio interface resources. A significant increase of capacity and thusfurther growth of the mobile communication market became possible. Already shortlyafter the start of GSM in 1992, subscriber numbers exceeded the million mark inmany countries. Other digital systems such as IS-95 followed. A development to agenuine mass market has been evident since the introduction of the secondgeneration of mobile communications.

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0,01

0,1

1

10

100

1000

Su

bs

cri

be

r [M

.]

1980

1982

1984

1986

1988

1990

1992

1994

1996

1998

2000

Year

Germany

World

Subscriber trends:

1980 - 2000

1G

IntroductionSingle cell

systems

2G

Introduction

Fig. 21

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Trends & Outlook

The mobile communication market will expand greatly in the future as well. Incontrast to the fixed network sector, which has developed slowly in the past decadesand has only recently become more dynamic, many predict unhindered growth for themobile communication sector beyond the year 2000. Only the growth of the Internetis expected to exceed the growth of the mobile communication sector. It is generallyexpected that the number of the mobile communication subscribers will rapidlyapproach that of the fixed subscribers, and that in regions with a poorly set up infra-structure, the number of mobile communication subscribers will clearly exceed that offixed subscribers within the foreseeable future.

Almost three billion mobile communication subscribers world-wide are expected by2015. This growth is apparent in the currently developing countries and newlyindustrialized countries of the Asian / Pacific region. A 50 % share of the worldwidemobile communication market is expected for the Asian / Pacific region by 2015; forindustrial nations in North America and Europe (EU15), a share of only about 7 % -11 % is expected.

0'

500'

1000'

1500'

2000'

2500'

1995 2000 2005 2010 2015

RoW

As ia / P ac ific

North A m eric a

EU 15

UMTS Forum

Report #1

Trends & Outlook

Su

bscri

ber

[M.]

Year

Fig. 22

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Mobile Trends

The mobile radio systems of the second generation have been optimized for speechtransmission. Data transmission is possible, but has previously been consideredsecondary. Taking the increasing mobility in the professional world (work outside theoffice, telework) into consideration, the need for mobile transmission of data is in-creasing. Comparatively user-unfriendly terminals (adapter solution) and relativelylow data transmission rates are problems for data transmission of the secondgeneration of mobile communications. The data rates for GSM are between 0.3 - 9.6kbit/s, the transmission rates of other cellular standards are comparable or less. Thefirst mobile satellite systems of the second generation also have only low datatransmission rates (Iridium max. 2.4 kbit/s, Globalstar max. 9.6 kbit/s). These ratesare considerably lower than those of ISDN (64 kbit/s).

A large variety of demands are being placed on future mobile communications. Alongwith improved world-wide service, user friendliness and cost reduction, mobile PCInternet connection with a high data transmission rate is required.

Many of these demands are taken into account in GSM Phase 2+.

In this way bearer services were standardized with transmission rates in order to in-crease data transmission rates as well as to realize “mobile computing” and accessto the Internet. Data transmission rates can be adapted to the transmission rates ofISDN and can be increased significantly further (up to more than 100 kit/s) by meansof these bearer services. User friendly equipment and cost-reduced features are alsoplanned, such as improvements in speech quality and world-wide availability bymeans of satellite roaming. Furthermore flexible services adaptable to customer re-quests and intelligent network services are planned.

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Trend:

Voice Data

Mobile Trends

Source:

UMTS Forum

0

20

40

60

80

100

Tra

ffic

[%

]

1996 2001 2005 2007

Year

Voice

DataRequirements:• high data rates

• user-friendliness

• improved service offering

• cost reduction

• worldwide accessibility

GSM Phase 2+• data rates > 100 kbit/s

• mobile computing, Internet

• new, integrating ME

• new flexible services + IN

• satellite roaming

• & much more

Fig. 23

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Mobile Forecast (Europe)

10 % of the traffic is expected to be on the data transport radio interface already in2001, 30 % in 2005.

If further capacities and higher data transmission rates are achieved, there are hardlyany limits to a further growth of the mobile communication market even after thenumber of subscribers reaches saturation.

The market share of speech transmission is as of 2007 expected to be less than 50% in the entire volume of traffic.

An enormous change in the proportion of speech transmission to data transmissionhas thus been predicted in the use of mobile communications in the first decade ofthe 21st century.

It will be expected

change from speech to data transmission

high data rate multimedia applications.

Predictions assume a minor but slowly increasing share of multimedia users inEuropean mobile communications after the implementation of GSM Phase 2+features, HSCSD and GPRS (as of 2000).

This is also the limit of GSM. Although the performance capacity of GSM Phase 2+far exceeds the original expectations for the second generation of mobilecommunications, neither the frequency ranges available nor the narrow-bandfrequency use in GSM suffice for the predicted increases and demands regardingdata transmission, especially multimedia use.

The third generation of mobile communications with GSM's successor, the UMTS(Universal Mobile Telecommunications System) is to deal with these applications anddemands as of 2002.

A considerable increase in multimedia use is expected with a wide-range expansionof UMTS as of 2005. Predictions of the UMTS forum assume that of the approx. 260million European mobile communication subscribers in 2010, approx. 90 million couldbe multimedia users, while the rest of the users use only speech and low data rateservices. Multimedia users will produce more than 50 % of the entire traffic rate.

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Mobile subscriber

(total)

Mobile subscriber

all applications from

voice to Multimedia

Mobile subscriber

Speech only/

low data rates

Mobile communication

forecast (Europa)

mobile Multi Media:

• Start with GSM Ph2+

• Breakthrough:

3G (UMTS)Source: UMTS-Forum

0'

50'

100'

150'

200'

250'

300'

1995 2000 2005 2010

Year

Su

bs

cri

ber

[M.]

Fig. 24

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The Third Generation (3G)

There are at the time many mobile communication standards of both the second and(still) first generations. Cellular mobile networks of the most different standardscomplement one another or compete with private mobile radio systems, cordlessstandards, paging systems and satellite systems, etc. Every one of these standardshas specific features, advantages and disadvantages, applications and user circles.Many of these systems exist only on a national level and/or are incompatible. To acertain extent this scenario reassembles on a world-wide level the situation of thecellular systems in Europe before the introduction of GSM.

IMT-2000 (International Mobile Telecommunications 2000)

The third generation of mobile communications represents a world-wide system ofcompatible standards, in which the most various current and future demands ontelecommunications have to be dealt with. The main task is to provide services to thecustomer, independently of his location and the specific available infrastructure.Smooth mobility should be guaranteed over all operator-dependent, national andgeographic borders at any location.

The demands on the third generation mobile communication systems have beendiscussed since the early 90s under the term FPLMTS (Future Public Land MobileTele-communications Systems). The term FPLMTS was changed into a term easierto pronounce, IMT-2000, in the mid 90s for countries in which English is not a nativelanguage. IMT stands for International Mobile Telecommunications 2000 indicatesboth the approximate date of introduction and the frequency range.

The International Telecommunications Union - ITU - is responsible for the IMT-2000specification. IMT-2000 is planned as the world-wide guideline of all standards of thethird generation of mobile communications. All of the "regional" standardization unitsfor developing standards must fulfil the ITU stipulations for IMT-2000. This ensures acompatibility of the standards to be specified without hindering innovative individualdevelopment and competition.

Many regional standardization committees create their own standards under the IMT2000 "roof". Nevertheless, UMTS (Universal Mobile Telecommunication System) asGSM successor system is expected to dominate the 3G market

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e.g. UMTS, cdma2000, UWC-136

2G(digital)

Paging Systems

e.g. ERMES

Cordless Telephonee.g. DECT, PACS, PHS

WirelessLocal Loops

WLL

PMRe.g. TETRA

Cellular systems

e.g. GSM, D-AMPS,IS-95, PDC

MSS

e.g. IRIDIUM, ICO, Globalstar

1G(analog)

Cordless Telephonee.g. CT1, 1+

Paging Systems,

e.g. City Call

wirelessTelephone cell

Private Mobile RadioPMR

Cellular systems


MSS

e.g. INMARSAT

3G

1 family of

standards

for all

• applications

• countries

different, incompatible standards for

different applications, countries & regions

IMT-2000

Fig. 25

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UMTS - Universal Mobile Telecommunications System

The ETSI (European Telecommunication Standards Institute) has specified UMTS asthe successor of GSM; a forum call Third Generation Partnership Project 3GPP, co-operating with the most important standardization organizations of the world isresponsible since 12/98. UMTS will fulfil the requirements for IMT-2000.

With UMTS world-wide multimedia access is possible at any time to all ranges whichare currently operated by various mobile communication systems of the first andsecond generations.

Data rates of 8 kbit/s to 2 Mbit/s are to be supported. UMTS will support zone 1 – 3 ofthe four zones of the IMT-2000 concept:

Zone 1 Indoor: for offices, private households,...; for low speed (stationary / up to10 km/h) max. data rates up to 2 Mbit/s are theoretically possible.

Zone 2 Urban: for city, shopping malls, railway stations, subways, airport halls forlow speed (stationary / up to 10 km/h) max. data rates up to 2 Mbit/s aretheoretically possible.

Zone 3 Suburban/Rural: For wide range mobility (car, train) with higher / highspeeds (up to 120 / 500 km/h), 384 kbit/s 144 kbit/s should be possible. (Remark:for UMTS only the lower speed value is currently planed)

Zone 4 Global: For rural, thinly populated areas with low user densities. All speedsfrom stationary (individual buildings, measuring stations), to intermediate speeds(car, train, ship), to 1000 km/h (airplanes). Mobile satellite systems (e.g.INMARSAT: Horizons) which ensure up to 144 kbit/s are planned for servicing.

For IMT-2000 the frequency ranges from 1885 - 2025 MHz and from 2110 - 2200MHz should be reserved (requested by ITU).

UMTS uses in Europe the frequency ranges of 1900 - 1980 MHz, 2010 - 2025 MHzand 2110 - 2170 MHz.

The frequency ranges of 1980 - 2010 MHz and 2170 - 2200 MHz are reserved for 3GMSS.

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Zone 4: Global

Zone 3:

Suburban / Rural

Zone 2:Urban Zone 1:

IndoorPicoCellMicro

CellMacro

CellMSS

max.

data rate144 kbit/s 384 kbit/s 2048 kbit/s144 kbit/s

UMTS - Universal Mobile Telecommunications System

1 8 5 0 1 9 0 0 1 9 5 0 2 0 0 0 2 0 5 0 2 1 0 0 2 1 5 0 2 2 0 0 2 2 5 0

cellular MSS cellular MSS

1885

2010

2110

1980

2025

2170

2200

Frequency range [MHz]

Fig. 26

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Transmission Principles


Contents

2GSM Network Structure 1 14 Duplex Transmission & Multiple Access 2 21 GSM - Fixed Network Transmission 325GSM Air Interface 4

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Transmission Principles Siemens

1 GSM Network Structure


GSM Network Structure

Fig. 1

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GSM: The Network Structure

The international GSM service area covers all countries in which there is a GSMnetwork.

Networks provisioned by an operator on a national level for public mobilecommunication are called Public Land Mobile Networks PLMN. PLMNs builttogether with public fixed networks, i.e. "conventional" PSTN (Public SwitchedTelephone Network) or ISDN (Integrated Services Digital Network) networks thetelecommunication infrastructure of a country.

A Public Land Mobile Network is divided into mobile and fixed network components.They are connected via air interfaces.

Fixed Network Components of the PLMN

The fixed network components of a GSM-PLMN consist of:

Base Station Subsystem BSS: The BSS is the fixed network part of the PLMNradio access (Radio SubSystem RSS). It realizes the radio transmission via theradio interface. Several fixed radio station, so-called Base Stations BS are co-ordinated by one control unit.

Network Switching Subsystem NSS: The NSS forms the interface between theradio subsystem and the public fixed networks (PSTN, ISDN, PDN). It executes allsignaling functions for setting up connections from and to mobile subscribers. It issimilar to the exchanges of fixed network communication systems, but itfurthermore fulfils important mobile communication specific functions, e.g. keepingtrack of the users / mobile stations location.

Mobile components of the PLMN

The Mobile Stations MSs are regarded as mobile part of the PLMN. The air or radiointerface represents the connection between the MS and the PLMN fixed networkcomponents BSS and NSS. The organization of the radio interface is decisive foradvantages and disadvantages of different mobile systems.

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Mobile

terminal device

BSSBase Station

Subsystem

NSSNetwork Switching

Subsystem

control/switching of

mobile services

BSSBase Station

Subsystem

BSSBase Station

Subsystem

PLMNPublic Land Mobile Network

PSTNPublic Switched

Telephone Network

ISDNIntegrated Services

Digital Network

PDNPublic Data

Network

MSMobile

Station

Mobile

components

Fixed network

components

UmAir Interface

Fixed

network

GSM Network Structure: Concept

Fig. 2

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Mobile Components

Mobile components are the Mobile Stations MS which transmit the users speech anddata to the PLMN. The Mobile Station MS consist of:

ME: Mobile Equipment,

SIM: Subscriber Identification Module,

The MS is not necessarily the termination point for the users data transmission. ATerminal Equipment TE, e.g. laptop, fax machine,... can be connected to the MS forfinal data handling.

The Mobile Station MS

An important difference between fixed network communications and mobilecommunications is the separation of equipment and subscriber identity. It is possiblefor the mobile subscriber to use various mobile terminal equipment with a personalidentity by means of the SIM card, which includes his subscriber identity. The mobilestation is defined as: MS = ME + SIM.

The SIM card is allocated and activated by the provider upon completion of thecontract. It is realized by means of a chip which contains a variety of permanent andtemporary information for the subscriber (e.g. personal telephone register) and abouthim/her. Along with the personal (secret) ID numbers (IMSI - International MobileSubscriber Identity, TMSI - Temporary Mobile Subscriber Identity) these storedinformation are for example algorithms and keys for ciphering the transmission.

The PIN (Personal Identity Number) is important for the subscriber; it must beentered by the mobile subscriber before the start of the conversation in order toprevent fraud by unauthorized intruders. As a rule, calls cannot be made without aSIM card in the ME and without the PIN being entered. Emergency calls are anexception.

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SIMSubscriber Identification Module

MS = ME + SIM

Mobile Components

SIM card: „the heart of MS“

• Different equipments, one SIM (one bill)

• Security: PIN (exception: emergency call)• Chip with subscriber identification,

security algorithms,

personal phone book,...

Fig. 3

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The Cellular Network

The breakthrough in mobile communications with regards to subscriber numbers andcapacity was made possible by the introduction of the cellular radio system. Thecellular communication system was tested in various countries during the 1970s.

Cellular networks of the first generation were introduced, e.g.:

1979 in the USA: AMPS (Advanced Mobile Phone Service)

1981 in Scandinavia: NMT (Nordic Mobile Telephone)

1985 in Germany: C-450 (Siemens)

1985 in Great Britain: TACS (Total Access Communications System)

The successive digital systems of the second generation, and therefore GSMsystems, are structured as cellular communication systems in the same way as theanalogue systems.

Principle of the Cellular Communication System

PLMNs operating on a national level are divided by location into servicing areas, so-called cells, in which a Base Transceiver Station BTS supplies the mobile subscribersof the area concerned. The cells represent the smallest service area in the PLMNnetwork.

A variety of cells ensures service of the total PLMN service area. The cells aretheoretically arranged in a so-called honeycomb pattern. Adaptations to thepopulation/ traffic density and the topography of the service area lead to a moreirregular pattern.

The service areas of the individual cells partially overlap. In order to avoidinterference of different subscribers in surrounding cells the cell structure isorganized according to the principle of cellular systems, frequency re-use. Thenarrow available frequency range is divided into individual frequencies (channels).Only some of these channels are used in a certain cell, the remaining channels areused in the adjacent cells. The same frequency is used again in cells which aresufficiently far apart from each other to avoid interchannel interference. This meansthat any area can be covered and thus an enormous increase in network capacitycan be achieved with a small supply of channel frequencies.

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The Cellular

Network

Principle:• Many cells (BTS)

• Full coverage

• Partial overlap of cells

• Distribution of frequency resources

• Only a few frequencies per cell

• Frequency re-use

Solution:

cell,

radio cell

r = cell radius(cell parameter)

Principle:

~ 4 r

channels

u, v, w

channels

x,y,z

r

channels

x,y,zco-channel interference zone

= cluster area

re-use distance

for HF channel frequency

re-use distancefor

HF channel frequency

Fig. 4

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Cluster

A certain minimum distance must be maintained between cells using the samefrequencies in order to prevent interference or at least keep it to a bare minimum.This minimum distance, the so-called frequency re-use distance, depends on theconcrete network planning and corresponds to approximately 4 times the cell radius.On this principle, the available channels can be divided e.g. into 7 parts anddistributed over the PLMN area in such a way that each cell contains one of these 7sets of frequency channels. The minimum area in which the whole range of HFchannels is used is described as a cluster. Planning a concrete network implies thatthe population/traffic density, the topography of the area to be supplied, etc. must betaken into account. This network planning is an extremely difficult process; there isspecial network planning software for this purpose.

• Frequency re-use distance: avoid inter-channel interferences

• Cluster: smallest domain within which all frequency resource is used

(GSM900: typ. 7/9 cells)

• Network planning: difficult

The Cellular Network / Principles of Network Planning

Fig. 5

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The GSM Cell

The higher the traffic density, the smaller the cell area since a limited number of HFchannels can only cope with a limited traffic volume. This can be carried out via areduction of the cell radius or by dividing the cells into sectors.

Cell Size / Hierarchical Cellular Structures HCS

The size and shape of the cell depend on:

The range of the MS radio contact (MS output peak power); topography (e.g.mountains, buildings, vegetation etc) and climate play a role here.

Traffic density

The maximum radius of a cell broadcast channel is 35 km in the GSM900 system, 8km in the GSM1800 system. The possibility of setting up "extended range cells" witha radius of up to 100 km has been integrated into GSM Phase 2+ for GSM900systems. This should allow coverage of sparsely populated areas and especiallycoastal regions. The extended cell concept results in a reduced capacity.

Transmit power is limited for higher traffic densities in order to achieve a high degreeof re-use of frequencies over smaller cells: The size of clusters is inverselyproportional to the capacity of the radio system.

A Hierarchical Cell Concept (Rec. 05.22) is planned for towns, with an extremely highdensity of mobile subscribers.

Macro-Cell: The "normal" cells are called Macro Cells. They have ranges fromapproximately one km to several (extended cell concept: 100 km).

Micro Cell: Cells for the support of restricted areas with very high mobile userdensity, e.g. shopping malls, railway and subway stations, airport terminals. Theirradius ranges from some 100 meters to approximately 1 km.

Pico Cell: Cells for the support of indoor applications, e.g. offices. Their rangeshould be several 10m.

Velocity dependent Handover are necessary in the Hierarchical Cellular Structures.

Cell Coverage

Omni Cells: The BTS is equipped with omni-directional antennae and serves a360° angle.

Sector Cells: The BTS supplies the cells with directional antennae. The cell shapeis a circular segment. Sectors of e.g. 180° or 120° are covered.

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Cell Size and Coverage

Maximum cell size

GSM90035 km

(100 km)

8 kmGSM1800

Cell coverage

360°

180°180°

cell 1

cell 2

120°120°

cell 1

cell 2

cell 3

120°

omni cell

180°

sector cells

120°

sector cells

(extended cell)

Hierarchical Cellular Concept:

• Macro cells: min. 500 m

• Micro cells: some 100 m

• Pico cells: some 10 m

speed-dependent allocation

Fig. 6

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Roaming / Location Registration / Handover

Roaming

A further innovation of the cellular system was so called Roaming. This means that asubscriber can move freely within the PLMN and remain reachable on a singlepersonal telephone number anywhere in this area. With GSM this concept of roamingcan be expanded to the international area (international roaming). A subscriberwhose home PLMN has a roaming agreement with other countries' GSM-PLMNs canalso be reached in these PLMNs (Visited PLMN - VPLMN) without dialing thecorresponding VPLMNs code; calls can also be made from that VPLMN. Aprerequisite is of course that subscriber’s authorization for international roaming.

Location Registration / Location Update / Location Area

The subscriber has to be located in the respective cellular network. A procedureknown as Location Registration or Location Update Procedure LUP carries outthis function. It is important that the subscriber's temporary location area is recorded /registered with this procedure when the subscriber's mobile station is switched onand checked in, to forward calls to him. The temporary Location Area LA is the areain which the MS can move freely without having to carry out a location update. As arule, the location area consists of a multiple cells and is configured by the operatoraccording to the traffic or population density.

Handover

In cellular networks, it is not necessary for the subscriber to have his call interruptedwhen changing from one cell's service area to the area of a surrounding cell, as longas the cell areas overlap. This overlapping should be guaranteed with good planning.If the MS can receive better supply from another cell than the one currently in useduring a call, the MS connection will be diverted to the relevant cell. This proceduredesigned for system quality maintenance ideally takes place without the user beingable to notice and is known as handover.

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Roaming, Location Update

& Handover

BS

BS

Location Update:• Location Area: most precise location information

stored in the network

• Location Registration: initial registration

• Location Update: update of registration

MS

Handover

Fig. 7

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2 Duplex Transmission & Multiple Access


Duplex Transmission

& Multiple Access

FDD TDD

UL DL

Duplex

transmission

Multiple

Access

FDMA

TDMA CDMA

Fig. 8

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Duplex Transmission and Multiplex Procedure

In a cell for access to a network two different principles have to be co-ordinated: Theway of co-ordinating UL and DL, i.e. the Duplex Transmission, and the way ofenabling the simultaneous access of several user to the same Base Station, i.e. themultiple access principle.

Duplex Transmission: FDD & TDD

Modern cellular mobile radio systems of the first (1G) and second generation (2G)enable full duplex transmission. Simultaneous communication on both sides, i.e.(virtually) simultaneous transmission and reception is thus possible.

The transmission directions are designated as Uplink UL (MS to BTS) and DownlinkDL (BTS to MS).

There are two duplex transmission principles:

Frequency Division Duplex FDD: Transmission and reception take place indifferent frequency ranges. The distance between the Uplink UL and Downlink DLfrequency range is designated as duplex distance.

Time Division Duplex TDD: Transmission and reception take place in the samefrequency band. Uplink UL and Downlink DL transmission take place at differenttimes. There is fast switching between UL and DL transmission, so that the userhas the impression of simultaneous transmission and reception.

receive

transmit receive

transmit

transmit

transmitreceive

receiveMS

BS

UL ULDL DL

time t

T

frequency f

Duplex distance

UL / DLseparated by

frequency !

Same

frequency

UL / DLseparated by

time!

FDDFrequency

Division Duplex

Uplink UL

Downlink DL

Base Station BS Mobile Station MS

TDDTime

Division

Duplex

Fig. 9

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Multiplex Access: FDMA, TDMA and CDMA

Several subscribers in one cell must be able to use the frequency range available formobile communications together. Thus there must be procedures for regulatingsimultaneous access of different subscribers without disturbances. There are threedifferent general procedures, partially in combination, which are used for co-ordinating the frequency resources:

FDMA - Frequency Division Multiple Access

TDMA - Time Division Multiple Access

CDMA - Code Division Multiple Access

FDMA - Frequency Division Multiple Access

FDMA is a multiple access principle used widely in the first (analogue) generation 1Gof mobile communications. It is however also used in the second (digital) generation2G of mobile communications, usually in combination with TDMA and in the thirdgeneration 3G together with CDMA.

The available frequency reserves are divided into channels of the same bandwidthfor FDMA. A certain frequency uplink and downlink is made available to an individualsubscriber. Simultaneous calls and information transmissions of various subscribersthus take place on different frequencies. The transmitter and receiver must have acommon knowledge about the channel frequencies to use.

FDMAFrequency Division

Multiple Access

Multiplex Access

TDMATime Division

Multiple Access

CDMACode Division

Multiple Access

Co-ordination

of limited frequency resources

for different subscribers

Fig. 10

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TDMA - Time Division Multiple Access

The allocation of the available frequency range is made with respect to time forTDMA. A frequency band is not permanently available to one mobile station; it isused by several different mobile stations. Time is therefore split into individual timeslots. The individual mobile stations are assigned the frequency range for theduration of a TDMA time slot in a periodically exclusive manner.

A certain number of subscribers can use a certain frequency range virtuallysimultaneously with TDMA. The message information of a subscriber is taken apartand transmitted piece by piece to the corresponding time slots. The informationcarrying HF transmission in an individual time slot designated as a "burst".

CDMA - Code Division Multiple Access

In CDMA systems the users of one cell are not separated by frequency or time.Different to FDMA or TDMA simultaneously they take place in the same frequencyrange. The users are separated by unique Codes. The Base Station and MobileStation must have common knowledge of the Codes used. The information of asingle user is spread up from a narrowband signal to a wideband signal using a high-frequency code (high so-called "chiprate"). This spread information is transmitted viaradio interface. After receiving the information, it is de-spread using the same code toregenerate the original information.

The Codes in principal have orthogonal properties.

frequency f

time t

power

TS 1

TS 2

TS 3

TDMA

frequency f

time t

power

1 2 3

FDMA

frequency f

time t

power

1

2

3

CDMA

Multiple

method

BS & MS share

knowledge about

FDMA

TDMA

CDMA

Frequency

Time

PN code

P

P P

Multiple Access methods

Fig. 11

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Transmission via GSM Radio Interface Um

A combination of FDMA and TDMA is used for GSM. The GSM physical channels aredefined by a pair of frequency bands (for UL and DL) and a Time Slot TS.

FDMA in GSM

In the GSM system, a band width of 200 kHz is defined for one frequency band.These HF channel widths are perfectly suited to the demands for speechtransmission.

Allocation to (E-) GSM900, GSM-R, GSM1800 and GSM1900 is as follows:

GSM900: (880) 890 - 915 MHz; 925 (935) - 960 MHz; 124 (174) channel pairs ;with a duplex distance of 45 MHz

GSM-R: 876 - 880 MHz; 921 - 925 MHz; 19 channel pairs; with a duplex distanceof 45 MHz

GSM1800:1710 - 1785 MHz; 1805 - 1880 MHz; 374 channel pairs; with a duplexdistance of 95 MHz

GSM1900: 1850 - 1910 MHz; 1930 - 1990 MHz; common use along with otherstandards (e.g. IS-95; D-AMPS); with a duplex distance of 80 MHz

In GSM for DL the higher and for UL the lower frequency range is used in general.

Remark: In co-ordination with the frequency plan regulation, there is a 200 kHzprotective band inserted between the lower limit frequency and the first carrier ofevery sub-band, i.e. the corresponding channels are not used. This protective bandknown as the "guard band" is an accepted, virtually "unavoidable loss" for preventinginterference between different applications in the totally filled frequency range.

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FDMA in GSMGSM900 / 1800 Frequency Allocation

C - Radio Frequency Channel (RFC)200 kHz

UPLINK (UL) DOWNLINK (DL)

Guard band

(880) 890 MHz

1710 MHz

915 MHz

1785 MHz

(925) 935 MHz

1805 MHz

960 MHz GSM900

1880 MHz GSM1800

Duplex distance 45 MHz resp. 95 MHz

25 (35) MHz

75 MHz

25 (35) MHz

75 MHz

Transmit bandof the Base Station

C

124

(174)

374

C

124'

(174')

374'

C

1

C

2

C

3

C

1'

C

2'

C

3'

Transmit bandof the Mobile Station

Fig. 12

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TDMA in GSM

Each of the 200 kHz frequency bands is further sub-divided by TDMA into 8 so calledTime Slots TS. This produces 8 physical channels within one frequency band. InGSM a physical channel is thus defined by a determined frequency channel UplinkUL and Downlink DL and a determined time slot TS

In the GSM system, up to 8 (with half-rate transmission even 16) calls can betransmitted "simultaneously" on one frequency band.

A sequence of 8 time slots TS in one radio channel is referred to as a TDMA frame. ATDMA frame has a duration of 4.615 ms, an individual time slot a duration of approx.0.577 ms. The users data are transmitted virtually "piece by piece" on one specifictime slot every TDMA frame.

GSM:combined

FDMA/TDMA

TDMA

frame

FDMA

time

frequency200 kHz

0

1

3

2

4

5

7

6

1

0

1TS = 577 s

1 TDMA frame =8 TS = 4.615 ms

1TS = 577 s

1 TDMA frame =8 TS = 4.615 ms

Fig. 13

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3 GSM - Fixed Network Transmission

PCMPulse Code

Modulation

speech band 1

speech band 3

speech band 2common line

Multi-

plexerband

3 2 1

1 0 1 1

0 0 1 1

A/D conversion

1 1 0 0

GSM - fixed network transmission


Fig. 14

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PCM30: Transmission in GSM fixed network part

Information (conversations, data, signaling) is exclusively transmitted digitally viaPCM30 lines in the GSM-PLMNs fixed network part.

Pulse Code Modulation - PCM

Sampling values of a speech information are transmitted using binary code words(digitally) in PCM.

Due to the digital structure of the message, the PCM signals are less susceptible tointerference than analogue signals. Regenerators reconstruct the original digitalsignal at the receiving end. Analogue signals, on the other hand, can only beamplified (including noise peaks).

Amongst other things, during Pulse Code Modulation (PCM) an analogue oscillationis converted into a digital signal. A PCM signal can be transmitted alone or beembedded in a TDMA frame with other PCM signals (multiplexing).

The conversion of an analogue telephone signal into a digital signal is carried out inthree steps:

1. Band limitation: A bandpass filter restricts the incoming signal to the audiblefrequencies, i.e. to 300 to 3400 Hz.

2. Sampling: Sampling values are taken at fixed intervals from the limited telephonesignal. The sampling frequency must be greater than twice the highest frequencywithin the analogue signal (Shannon Theorem). Internationally specified: 8000 Hz.

3. 8-bit coding: Every amplitude value of the sampled (Pulse Amplitude Modulated -PAM) signal is transformed into an 8-bit word. The 8-bit word enables the analoguesignal to be represented in 256 quantization intervals.

Since the transmission of an 8-bit word requires only a portion of the sampling

interval (125 s) of the analogue signal, the 8-bit information is temporallymultiplexed (TDMA-procedure). 8 bits are transmitted in each time slot.

Using PCM30 transmission systems, a total of 30 digital user values can betransmitted in the time frame of the sampling period of an analogue value, i.e. in 125

s.

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1. Band limitation

(300-3400 Hz)

2. Sampling (8000 Hz)

3. 8-bit coding

Generation of a PCM Signal

transmission of the coded

sample value of signal 1

coded sample value

signal 2

time slot

0 1 0 0 1 1 0 1

signal 1

Fig. 15

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PCM30

PCM30 transmission systems use digital transmission lines or radio relay. A PCM30frame consists of 32 time multiplexed time slots.

The 32 time slots can contain pulse code modulated message information (speech,data) or signaling information in the form of 8-bit words.

The total bit rate of a PCM30 line is 2048 kbit/s

Time slot 0: alternately frame identification word and service word (alarms)

Time slots 1-15 and 17-31: calls or data

Time slot 16: signaling channel

The pulse frames are transmitted in a direct sequence.

PCM30: TDMA Principle

telephone channels 1 - 15 telephone channels 17 - 31

frame alignment/

service word channelsignaling channel

time

slot

PCM30PCM30

pulse frame pulse frame pulse frame

Fig. 16

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4 GSM Air Interface

GSM Air Interface

Advantage:

mobility

Single cell systems Cellular mobile communication systems

Limits:

1st generation 2nd generation incl. satellite roaming

cell national GSM service area unlimited

GSM (Ph1/2) (GSM Ph2+)


Fig. 17

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Radio Interface: Advantages, Problems and Solutions

The air or radio interface, i.e. the connection between the MS and fixed networkcomponents, represents the fundamental difference to a fixed networktelecommunication system. The radio interface has its specific advantages, but alsoshows problems and disadvantages inherent to mobile communications.

Advantage: Mobility

The main advantage of mobile communications is the unrestricted mobility which canbe achieved only via a radio interface. Mobility was extremely restricted, especially inthe early years of mobile communications (one-cell systems). Mobility only reachedas far as the radio coverage between the MS and the transmission/receivinginstallations would allow. These limits were stretched significantly by cellular mobilecommunication networks of the first generation (since the early 1980s). Nationalborders and the degree of area coverage of a PLMN within a country formed theborders. In the GSM system, national borders no longer represented restrictions tomobility owing to “inter-national roaming”. It is still the case that nation-wideconnectivity is only offered around urban areas and along main traffic routes in largeareas of central Europe. Unlimited world-wide mobility is possible in co-operationbetween GSM and MSS such as Iridium, Globalstar and ICO.

Problems & Solutions on the Radio Interface

Cost Aspect: Problem - The need to built up a new network architecture withthousands of BTS. But: Compared with the costs for a fixed network ISDN / PSTNinfrastructure, a GSM PLMN is comparable cheap, because there is no need formillions of lines into every private household.

Capacity: The capacity of transmission via radio interface is a great problem inmobile communications. Optimized usage of radio resources reducing the cellsizes, introducing sector cells and introducing the Hierarchical Cellular Structureswith Macro, Micro and Pico Cells solves this problem.

Data Rate: GSM (Phase 1/2) offers a maximum 9.6 kbit/s, compared to the 64kbit/s of ISDN. Introduction of HSCSD, GPRS and EDGE enhances the GSM datarates significantly.

Security Aspect: The radio interface can be intercepted with comparatively littletechnical expenditure. 1G could be intercepted without any problem, while thedigital transmission of the second generation offers protective measures againstinterception; the transmission is coded.

Health Aspect: The mobile radio frequencies lie near the resonance frequency ofwater (2.45 GHz). In order to keep thermal exposure to the mobile radio user aslow as possible there are maximum power limitations for mobile phones, 2 W forGSM900 and 1 W for GSM1800.

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The Air Interface Um:Problems of radio transmission and possible solutions

Cost Aspect:

Capacity:

Data Transmission Rate:

Security Aspect:

Health Aspect:

Construction of mobile

communication network

cheaper than terrestrial network

GSM900 / E-GSM: 124 / 174 frequency bands

GSM1800: 374 frequency bands

increasing subscriber numbers, data transmission

Resource optimization / protection !!!

GSM Ph1/2: 9.6 kbit/s

Ph2+: HSCSD, GPRS, EDGE > 100 kbit/s

Eavesdropping easy!

GSM offers encryption

H2O resonance frequency (2.45 GHz)

Thermal load

Pmax

= 2 / 1 W (GSM900/1800)

Fig. 18

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Problems of Physical Transmission

Screening: If there are hindrances between transmitter and receiver, the signalswill weaken. A connection can thus become problematic or impossible. In GSMthere is therefore the possibility of regulation of the transmitting power (PowerControl - PC) from mobile and base stations over several orders of magnitude.

Multipath Propagation: Multipath propagation through reflection and dispersionof radio waves leads to phase-shifted reception of signals of different paths. Theinterference can distort, amplify or erase the signal. An attempt to compensate fornegative effects of multipath propagation is given by power control, frequencyhopping, two antenna receivers for the base station (antenna diversity) andredundancy of the transmitted information.

Distance MS - BTS: The distance between MS and BTS has proved to beproblematic in several ways. The receive power sinks with increasing distancebetween transmitter and receiver theoretically with the square of the distance.Various physical effects such as atmospheric attenuation (weather-dependent)reduce the receive power even more. This attenuation depends on the frequencyand increases with increasing frequency in mobile radio relevant frequencyranges. The distance furthermore causes a reception de-lay, which may lead tointerference between neighboring time slots in TDMA. GSM responds to this delayby means of a regulation of the transmission time (Timing Advance TA). GSM900cells (GSM Phase 1/2) are limited to maximum 35 km, GSM1800 cells tomaximum 8 km radius as a result of the distance-related problems. There is thepossibility in GSM Phase 2+ to realize "Extended Range Cells" with a maximumradius of 100 km for GSM900.

MS Speed: Moving mobile stations can cause transmission distortions due toDoppler effect. A compensation for this effect up to a maximum speed of 250 km/h(130 km/h), for GSM-R a more powerful compensation for speeds of up to 450km/h was deloped.

Interference with external systems: The receive quality can also be disturbed byelectromagnetic waves from outside systems (e.g. car ignition, generators, PCs).A compensation is being tried out by means of the mechanisms described undermultipath propagation.

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Radio Transmission: Physical Disturbances

Mobility

• Screening

• Multipath propagation

• Distance MS-BS

• MS speed

• External system interferencetransmitted signal

received

signals

signal to

antenna

Digital systems offer manyerror recognition and

correction mechanisms( redundancy)

signal attenuation (Power Control PC) interference (PC, f-hopping, diversity, regeneration) power loss (f-dep.); delay (PC, TA, cell size) Doppler effect (corrections) quality loss (PC, f-hopping, regeneration)

Fig. 19

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Frequency Resources: Optimized Utilization

In order to be able to keep up with the increasing demands on mobilecommunications despite the limited resources of the radio interface differentapproaches are being pursued.

Additional Frequency Ranges: The simplest way to cope with the growingdemand for mobile communications is to expand the available frequency range.This approach was pursued with E-GSM and GSM1800. Any further futureexpansion would be problematic as other frequency ranges are already reservedfor other applications.

Speech Compression: Speech compression in GSM allows a reduction of voiceinformation from 64 kbit/s to 13 kbit/s in the so-called Full Rate FR speech and to5.6 kbit/s with the Half Rate HR speech. HR speech thus leads to a considerableincrease in capacity. Central aspects of HR speech are described in the GSM Rec.06.02, 06.20 - 22, 06.41 and 06.42.

Cell Size Reduction/Coverage: The most important measure for increasing thecapacity of GSM networks lies in a reduction of the cell size. The resources of aradio cell are available to a small geographical area through the reduction of thecell radius or through the limitation of the cell coverage (sector cell). By doing so,the density of mobile communication subscribers and consequently the systemcapacity can be considerably increased. By halving the cell radius, its capacity isincreased by a factor of four. Nevertheless the size of a (normal = macro) cell cannot be reduced indiscriminately. Hierarchical Cell Concepts (Rec. 05.22) withmacro, micro and pico cells are significantly enhancing efficiency.

OACSU (Off Air Call Set Up): Traffic channels are allocated only after a success-full call setup, that is after the called subscriber (delayed allocation). The OACSUprocedure thus serves to improve the frequency efficiency; it can be used foroverload handling.

Tariffs: Introduction of day- & night time tariffs can help to level down peak loads.

Discontinuous Transmission DTX: For a conversation, this will mean that justspeech phases are transmitted. Background noise, or so called comfort noise istransmitted with a greatly reduced bitrate (500 bit/s instead of 13 kbit/s as withspeech phase) in phases in which a subscriber is silent. The other subscribershould thus not worry that connection has been broken off. In order to makediscontinuous transmission possible, the presence of "useful" information fortransmission must be determined by means of Voice Activity Detection VAD. DTXaspects are included in GSM-Rec.06.31 and 06.41, VAD aspects in Rec. 06.32and 06.42.

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Frequency Resources: Expansion / Optimized Utilization

GSM900: 2 x 25 MHz

• Extension of frequency range:

E-GSM: 2 x 35 MHz

GSM1800 2 x 75 MHz+

Fixed network: 64 kbit/s

• Speech compression:

FR:

13 kbit/s

Digital speech information

HR:5.6

kbit/s

Half Ratespeech

Full Ratespeech

• Cell size

reduction:

(Radius reduction

and sectorization)

35 / 8 km 500 m

omnicell

180° / 120°

sector cell

• OACSU (Off Air Call Set Up)

• Time Balance / Tariffs

• DTX (Discontinuous Transmission) / VAD (Voice Activity Detection)

Fig. 20

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Advantages of Digital Transmission

Digital transmission has many advantages over analog transmission:

Network Capacity: The capacity of mobile communication networks can beconsiderably increased by the possibility of compressing digitalized speechinformation. The disadvantage of speech compression is a loss of information(reduction of speech quality).

Service Offer: Digital data transmission simplifies the transmission of signalinginformation. This makes the introduction of a wide, quickly growing range ofservices possible in GSM beyond pure speech or data transmission.

Cost Aspect: Digital equipment is less expensive to manufacture owing to betterpossibilities for use in highly integrated microelectronics. Purchase costs as wellas operation and maintenance costs are thus less expensive and have allowedGSM's breakthrough onto the mass market.

Miniaturization: Microelectronics used for digital information transmission allowsa relatively simple reduction of the hardware (in comparison to analogtransmission), especially of the mobile stations. Mobile phones have been usedwith GSM since the start; their weight has been reduced from over 500 g to some50g within a couple of years.

Security Aspect: Digital information can be ciphered much more easily thananalog information. Transmission via radio interface is protected from fraud andunauthorized interception in GSM by the ciphering the digital user data (speech,data) and signaling data.

ENCRYPTION

MODULE

Input data

(plain text)

Output data

(coded text)

Code

sequence

Advantages of Digital Information Transmission

• Network capacity speech compression

• Service offer signaling

• Cost aspect manufacture, operation, maintenance

• Miniaturization microelectronics

• Security aspect easily coded

• Transmission quality regenerability

Fig. 21

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Transmission Quality: Signal transmission via radio interface leads to consider-able distortions and weakening of the transmitted signals. Digital signals arefundamentally less susceptible to interference than analog signals and are bettersuited to regeneration. Analog speech connections become increasingly worsewith increasing distance from the transmitter until they eventually disconnect.Digital transmissions on the other hand maintain a constant good quality over along distance and then disconnect almost suddenly.

S / N

signal

quality

distance to transmitter r

analog signal

digital signal

Quality of Digital & Analog Signal Transmission

Fig. 22

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Reliable Transmission via Um: Channel Coding

Various measures are taken in GSM to protect transmissions via radio interface frominterference, distortions and loss of information. These measures are taken by meansof channel coding.

The transmission is protected in such a way that a certain number of transmissionerrors can be corrected by the error correction procedure, the so-called ForwardError Correction (FEC). By means of FEC the Bit Error Rates (BER) of the radiointerface transmission are reduced to a rate of 10-5 to 10-6 from an unacceptablevalue of 10-3 to 10-1. Redundancy is added to the information to be transmitted inorder to al-low recognition and correction of transmission errors.

Channel coding of information on the transmit side comprises three steps:

1. Adding of parity check bits and fill bits

2. Error protection (redundancy) with convolutional coding

3. Spreading by time: interleaving

The same steps are carried out in reverse order at the receiving side.

The added parity check bits serve to recognize incorrigible errors on the receivingside. The parity check bits are of special use in speech transmission. If incorrigibleerrors are indicated, the corresponding speech information is rejected and an attemptis made to interpolate the information from the preceding speech information.

Convolutional coding serves to create redundancy. The original information (speech,data, signaling) is coded along with the parity bits. Important information runs throughmathematical algorithms, where redundancy is added and the arrangement of theinformation is changed.

Interleaving serves to temporally spread information. Information is collected up to adetermined number of bits and is spread by time. The interweaving of the redundantinformation has the effect that information loss due to frequent short disturbances canbe compensated by means of temporal spreading of the information.

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Reliable Transmission via Um:

Channel Coding

Addition of:

parity

and filler

bits

transmission side

Convo-

lutional

coding

redundancy

Inter-

leaving

temporalspreading

Parity

check

Convo-

lutional

decoding

De-inter-

leaving

reception side

Um

Fig. 23

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Speech Coding: FR, HR and EFR

Speech transmission is of central importance in GSM. Speech information is handledespecially by the radio interface for secure and resource-preserving transmission.Speech information is compressed and then redundancy is added (channel coding).There are three different speech codecs available in GSM for compression of speechinformation: the Full Rate (FR) Speech Codec was specified for GSM Phase 1, i.e.from the start, in Phase 2 the Half Rate (HR) Speech Codec and in Phase2+ theEnhanced Full Rate (EFR) Speech Codec were added.

Full Rate FR and Enhanced Full Rate EFR Speech Codecs compress speechinformation from 64 kbit/s - used in digital line connected telephone networks such asISDN - to 13 kbit/s respectively 12.2 kbit/s. So 13 kbit/s / 12.2 kbit/s are the net datarate for speech transmission via the radio interface. The gross data rate after addingredundancy in channel coding is 22.8 kbit/s with FR and EFR.

Half Rate HR Speech Codec compresses speech information from 64 kbit/s to 5.6kbit/s. The gross data rate after adding redundancy is 11.4 kbit/s. The connectionsof two Half Rate speech using subscribers can be realized in one physical channeltogether, with a gross data rate of 22.8 kbit/s.

Models for speech generation are generally used for speech coding. Periodically re-turning elements of speech are identified as phonemata; redundancy is removedfrom the speech information. Even the attributes of hearing, especially the spectralcovering effect, are taken into account in different ways.

More efficient speech recognition mechanisms are of use for the HR introduced inGSM Phase 2 and EFR introduced in Phase 2+. The HR codec delivers a somewhatlower speech quality in comparison to the FR codec if transmission is undisturbed. Itis more robust against radio specific disturbances owing to the relatively strong errorprotection. The EFR codec offers a significant increase in quality in comparison to theFR codec. It sounds more natural and "smoother" according to subjective test results.

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Speech Coding: FR, HR, EFR

Speech coding models of speech and hearing• Removal of redundant information (periodic)

• Transmission of central speech information

• Reduction of speech information: 64 kbit/s 13 / 5.6 kbit/s (net data rate)

Full Rate (FR) CodecGSM Ph1;

13 kbit/s

Redundancy (channel coding)

9.8 kbit/s

Enhanced Full Rate (EFR) CodecGSM Ph2+;

12.2 kbit/s

Redundancy (channel coding)

10.6 kbit/s

Gross data rate via Um: 22.8 kbit/s

Half Rate (HR)Codec; GSM Ph2;

5.6 kbit/s

Redundancy

5.8 kbit/s

Gross data rate via Um: 11.4 kbit/s

HR & EFR: improved, acoustically optimized

speech coding

HR, FR almost the

same quality

Fig. 24

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GSM PLMN

GSM PLMN

Contents

2Overview1 7 Network Elements 2

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1 Overview

PLMN Public Land Mobile Network

PSTNPublic Switched

Telephone Network


Digital Network

PDNPublic Data

Network

MSMobile

Station

fixed

network

GSM-PLMN

BSSBase Station

Subsystem


Subsystem

OSSOperation SubSystem

RSSRadio

SubSystem

Overview

Fig. 1

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GSM PLMN: Subsystems

A GSM-PLMN is subdivided into the following subsystems:

Radio SubSystem RSS

Network Switching Subsystem NSS

Operation SubSystem OSS

Network Elements

The subsystems functions are grouped into functional units or network elements.Functional units may be realized either as standalone Hardware HW units orassociated with other GSM functional units in one HW unit.

The Radio SubSystem RSS consists of the Mobile Stations MS and the BaseStation Subsystem BSS, which is composed of the following functional units:

Base Station Controller BSC

Base Transceiver Station BTS

Transcoding and Rate Adaption Unit TRAU

The Network Switching Subsystem NSS (Phase ½) consists of the followingfunctional units:

Mobile services Switching Center MSC

Visitor Location Register VLR

Home Location Register HLR

Authentication Center AC

Equipment Identity Register EIR.

The Operation SubSystem OSS consists of Operation & Maintenance CentersOMC; in the Siemens solution:

Operation & Maintenance Center for the Base Station Subsystem OMC-B

Operation & Maintenance Center for the Switching Subsystem OMC-S.

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OMC- B OMC- S

MSC

HLR VLR

EIRAC

BSC

BTST

R

A

U

Mobile

Station

MS

Radio

SubSystem

RSS =

Base Station

Subsystem

BSS

Network

Switching

Subsystem

NSS

+

other

networks


PSTN

ISDN

Data

NetworksMS =

ME + SIM

GSM-PLMN

Fig. 2

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Interfaces

The individual network elements are connected to each other for user data and/orsignaling transfer. Some of the interfaces are specified by ETSI as open interfaces,allowing to connect equipment of different network manufacturer. Others are notspecified or "weakly" specified, so that only proprietary solutions are possible.

The following GSM Phase 1/2 interfaces are open interfaces:

Um: MS - BSS (Air interface)

A: MSC - BSS (BSC)

B: MSC - VLR

C: MSC - HLR

D: HLR - VLR

E: MSC - MSC

F: MSC - EIR

G: VLR - VLR.

The following interfaces are proprietary solutions:

Asub: BSC - TRAU

Abis: BSC - BTS

T: BSC, BTS, TRAU - Local Maintenance Terminal LMT

O: BSC - OMC-B

HLR - AC (no name)

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T

MS BTS

BSC

TRAU

VLR

AC

other networks

MSC/xxx interworking interface

LMT

LMT

LMT

OMC - B

Um Abis A

sub

C

B

F G

E DT T

O

A

GSM (Phase 1/2)

Interfaces

not specified

EIR VLR

HLRMSC

MSC

Fig. 3

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2 Network Elements

GSM-PLMN

PLMN Public Land Mobile Network

PSTNPublic Switched

Telephone Network


Digital Network

PDNPublic Data

Network

MSMobile

Station

fixed

network

BSSBase Station

Subsystem


Subsystem


RSSRadio

SubSystem

Network Elements

Fig. 4

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The Mobile Stations represent the mobile network components. They consist of theMobile Equipment ME and the Subscriber Identity Module SIM: MS = ME + SIM

The SIM card

The SIM consists of a microchip, which uses either a check card or a plate made of asynthetic material as a carrier. Without a SIM card, the use of an MS is normally notpossible. An exception is the emergency call, which should always be possible with afunctioning ME. The SIM card carries the subscriber-related information and codes,so that a GSM subscriber with a SIM card can use different ME. The main task of theSIM is the storage of data: permanent and temporary administrative data as well asdata concerning security. Personal telephone lists may be stored and using the SIMtoolkit with enhanced memory space, it is possible to enable applications such asMobile Banking, etc.

Important stored codes are e.g.:

Personal Identity Number - PIN

PIN Unblocking Key - PUK

Mobile Station ISDN number - MSISDN

International Mobile Subscriber Identity - IMSI

Temporary Mobile Subscriber Identity - TMSI

Important data relating to security are, e.g.:

the individual key - Ki

the cipher key - Kc

the algorithms for authorization and ciphering (A3, A8).

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ME:Mobile Equipment

•Hardware & Software for radio transmission•Cipher algorithm

SIM cardSubscriber Identity Module:


MS = ME + SIM

Subscriber license

Personal Identities

(e.g.MSISDN, IMSI, TMSI, PIN,...)

Subscriber key (Ki, Kc)

Algorithms (A3, A8)

Personal phone book

SIM toolkits,...

MSISDN: Mobile Subscriber ISDN no.

IMSI: International Mobile Subscriber Identity

TMSI: Temporary Mobile Subscriber Identity

PIN: Personal Identity Number

Ki: individual key

Kc: cipher key

Fig. 5

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The Mobile Equipment ME

The Mobile Equipment ME unites the tasks of many functional elements of the fixedGSM-PLMN network.

By using the data of the SIM card, the speech is digitalized, compressed, securedagainst loss of data (redundancy + interleaving), encrypted to prevent interceptionand modulated onto the Radio Frequency (RF) created by the mobile station. Directlyafter, the signal is amplified and transmitted.

In the opposite direction, the process runs inversely, beginning with the reception ofthe radio frequency (RF).

The MS represents the counterpart to BSC, MSC, HLR, VLR and EIR as regardssignaling. As a whole, ME and SIM cards are almost a complete GSM system asregards their functionality.

GSM Mobile Station

speechconversion

block diagram

Subscriber Identity Module SIM

Mobile Equipment ME

• securing• interleaving• burst block formation

ciphering

• HF generation• modulation• amplification

reverse speech conversion

• security check• de-interleaving• reformation

de-ciphering

• filtering• amplification• de-modulation

• Radio transmission counterpart to

BTS, BSC & TRAU

• Signaling counterpart to

BSC, MSC, HLR/AC, VLR & EIR

Fig. 6

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The Base Station Subsystem BSS

The BSS consists of the following network elements:

BSC: Base Station Controller

BTS: Base Transceiver Station

TRAU: Transcoding and Rate Adaption Unit

LMT: Local Maintenance Terminal

The BSS architecture shall be selected to achieve maximum flexibility with regards tothe various operator requirements. All BSS components can be installed in the samegeographical location or in different locations where the transmission paths can beused via public networks. The ability of the BSC to manage several BTSs in differentcell locations enables optimal adaptability to the traffic requirements in urban andrural areas.

In terms of function, the main task of the BSC is the handling of the call connections(switching), sampling of operational/maintenance information of all BSS (BSC, BTSsand TRAUs), as well as their transfer to OMC-B. The BTS handles the radio specificaspects.

BSCTRAU

LMT

BTS

BTS

BTS

TRAU

Base Station Subsystem BSS

Architecture

MSC

OMC-B

Fig. 7

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The Base Station Controller BSC is, as the controlling element, the heart and centerelement of the BSS.

BSC Location: between the interfaces Asub and Abis

BSC Functions:

switching of the user traffic between individual TRAUs and BTSs

control and monitoring of the connected TRAUs and BTSs

sampling of operation and maintenance information of BSC, TRAUs and BTSs aswell as transfer to OMC-B

evaluation of signaling information from MSC via TRAU and MS via BTS

Radio Resource Management for all connected BTSs

storage of the BSS configuration

back-up storage of the total BSS Software for fast system restart

TRAU

TRAU

TRAUBSC

BTS

•

•

•

BTS

BTS


•

•

•

• BSS control

• Switched between TRAU BTS

• Radio Resource Management

• Collecting error messages in BSS

• Contact to OMC-B

• Database storage, SW of BSS

• BSS control

• Switched between TRAU BTS

• Radio Resource Management

• Collecting error messages in BSS

• Contact to OMC-B

• Database storage, SW of BSS

OMC- B

Asub

Abis

Fig. 8

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Base Transceiver Station BTS

A BTS is the module which operates an individual cell and realizes the radiointerface. A BTS encompasses all applications concerning radio transmission(sending, receiving), as well as the air interface specific signal processing. The BTSis connected via the Abis interface with the BSC and via Um interface to the MSs.

Functions:

Channel coding: To protect the transmission, incoming information is provided withparity check bits and redundancy (convolutional coding) and spread in time overseveral HF bursts (interleaving).

Ciphering: After channel coding, the transmission of message information and thesubscriber data is coded to prevent illegal interception.

Burst block formation: The information is organized in blocks of a particular length(burst blocks). A so-called training sequence is added for synchronization andanalysis of transmission quality.

Modulation: The carrier frequency is created in the 900/1800/1900 MHz range andthe information is modulated upon this carrier.

Power Control PC: Control of the power level of the BTS and MS.

Timing Advance TA: Calculation of the distance of the MSs from the BTS; the MSsare informed of necessary transmission advance.

Frequency Hopping: a feature which enhances the reliability of information transfer

Synchronization: Providing of mobile stations with frequency and timesynchronization information.

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Um

parity

bits

convolutional

coding

inter-

leaving

channel coding

burst blocks

formation

burst

multiplexing

transmitHF generation modulation

modulation

user and signaling

information

Abis

receive

max. 16 carrier/cell

• Frequency hopping

• Synchronization

(time and frequency)

• Monitoring & optimization

of transmission quality

• Power Control PC

• Timing Advance TA

ciphering

Base Transceiver Station

BTS

+

Fig. 9

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Transcoding and Rate Adaptation Unit TRAU

The TRAU is used for speech compression (Transcoding) and adaptation of data tothe requirements of the air interface (Rate Adaptation). It lies between A and Asubinterface.

Functions:

Transcoding TC defines speech compression: compresses / decompresses theincoming speech data from 64 kbit/s to 13 kbit/s, 12.2 or 5.6 kbit/s (embedded in16 or 8 kbit/s channels).

Rate Adaptation RA filters out the useful data (0.3 – 9.6 kbit/s in Phase 1/2)coming from the MSC (64 kbit/s) signal and forms a 16 kbit/s signal toward theBSC

The user data are sub-multiplexed into 16 kbit/s subslots on the Asub interface

Remarks:

TC and RA are implemented as algorithms in the same hardware unit as theTRAU (Siemens solution).

The TRAU is logically allocated to the BSC. Consequently, it belongs to the BaseStation Subsystem (BSS), but is generally installed at the MSC node in order tokeep line costs to a minimum.

In contrast to user information signaling information passes the TRAUtransparently.

The users information (data / speech) is embedded into 16 kbit/s channels. Theadditional space is filled with proprietary inband-signaling (i.e. information, whichare directly exchanged between BTS and TRAU)

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TRAU

64 kbit/s64 6464

64 kbit/s64 6464

64 kbit/s64 6464

64 kbit/s64 6464

16

16

16

16

B

S

C

M

S

C64 kbit/s64 6464

16161616

submultiplexer

• speech compression: 64kbit/s 13 or 5.6 kbit/s + inband signaling

• data transmission: "64 kbit/s" 0.3 - 9.6 kbit/s + inband signaling

• signaling: transparent

• speech compression: 64kbit/s 13 or 5.6 kbit/s + inband signaling

• data transmission: "64 kbit/s" 0.3 - 9.6 kbit/s + inband signaling

• signaling: transparent

TRAUTranscoding & Rate Adaptation Unit

Asub

A

Fig. 10

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The Network Switching Subsystem NSS

The NSS comprises the following functional elements:

MSC: Mobile services Switching Center

VLR: Visitor Location Register

HLR: Home Location Register

AC: Authentication Center

EIR: Equipment Identity Register

Mobile services Switching Center MSC

The MSC is concerned with the central tasks of the NSS and covers the serviceareas of several BSSs. These tasks can be compared to those of an exchange in afixed network. These tasks are supplemented by mobile specific tasks of the sub-scriber administration. The MSC handles connection tasks in the PLMN, i.e. set-up ofcircuit connections to the BSS, between each other and other networks (e.g. PSTN).The MSC visited by a customer is described as a VMSC (Visited MSC). A MSC,which represents an interface to other networks, is called GMSC (Gateway MSC).

MSCs connect the other networks with the Base Station Subsystem BSS, as well asthe other NSS units with the BSS via the signaling highways.

The MSC is a stored program controlled switching system for national andinternational GSM-PLMN applications. The MSC is a switching center that carries outall switching for the mobile stations which are actually located in the MSC area.

Other functional units of the NSS (e.g. HLR, VLR, AC,...) can be associated to theMSC.

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other

networksMSC

Mobile services

Switching Center

other

MSC/VLRs

VLRVisitor Location

Register

EIREquipment Identity

Register

HLRHome Location

Register

ACAuthentication Center

NSSNetwork &

Switching

Subsystem

Fig. 11

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Overview of call processing functions

The MSC follows the functions of a fixed network exchange as regards itsfunctionality. Consequently, varied proven call handling functions form the basis formobile specific supplementary services.

Switching of user connections

Routing functionality (path selection)

Signaling with other MSCs and external network exchanges

Evaluation of available signaling information for destination routing:

Digit translation

Legal interception

Coping with abnormal signaling conditions, e.g. loss of signaling information

Supplementary Service support

Processing of transmission path attributes, e.g. echo compensation

Call supervision

Overload protection

Control of priority calls, e.g. emergency call

Charging

Traffic measurement and traffic observation

Support of maintenance and administration functions, e.g. connection cut off, trunktest and measurement

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MSC

Mobile services

Switching Center

call processing functions(similar to fixed network exchange)

mobile communication -

specific functions

• NSS “heart & center”

• Nodes between NSS registers, BSS,

other MSCs and external networks

• Serves several BSS (BSC)

• Set-up & switching of user traffic & signaling

• Always associated with VLR

• Association with HLR/AC and EIR possible

• Gateway MSC: Gateway to external networks

• Visited MSC: MSC serving certain MS

• NSS “heart & center”

• Nodes between NSS registers, BSS,

other MSCs and external networks

• Serves several BSS (BSC)

• Set-up & switching of user traffic & signaling

• Always associated with VLR

• Association with HLR/AC and EIR possible

• Gateway MSC: Gateway to external networks

• Visited MSC: MSC serving certain MS

Fig. 12

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Mobile specific functions

Additional to normal fixed network exchanges, the MSC has many mobile specificfunctions due to the users mobility.

Mobile specific functions are for example:

Signaling with BSC, MS & NSS databases (EIR, HLR, VLR)

Processing of mobile-specific services

Mobility Management, e.g. Paging, Inter-MSC Handover, Location Update,...

Overload handling, e.g. OACSU

Interworking Function for data services

Mobile specific Announcements

...

MSC

Mobile services

Switching Center

call processing functions(similar to fixed network exchange)

• Set-up of signaling / user connections

• Signaling evaluation destination determination• Connection path selection

• Processing of abnormal signaling information• Supplementary Service support

• Call monitoring

• Traffic monitoring & measurement

• Overload protection

• Billing

• Priority control (e.g. emergency call)

• Support of O&M functions

mobile specific functions

• Signaling with BSC, MS & NSS databases

• Processing of mobile-specific services

• Mobility Management, e.g. Paging, Inter-MSC Handover, Location Update,...

• Overload handling, e.g. OACSU

• Interworking Function for data services

• Mobile specific announcements

Fig. 13

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Visitor Location Register VLR

The Visitor Location Register VLR is responsible to aid the MSC with information onthe subscriber, which are temporarily in the MSC service area. Therefor, in praxis it isalways associated with an MSC.

The VLR request the subscriber data of user with activated MS on the MSC servicearea from the HLR and stores them temporarily. Temporarily means as long as thesubscriber is not registered in a new MSC/VLR, even if he deactivated the MS.

Additional to the semipermanent subscriber data received from the HLR the VLRstores temporary data, e.g. information on the subscribers current location (theLocation Area), the state of activation (Attached / Detached),...

Furthermore, the VLR is responsible for the initiation of security functions, e.g. theAuthentication procedure, the start of ciphering and the TMSI re-allocation.

Examples of subscriber data in the VLR:

MSISDN: Mobile Subscriber ISDN number


TMSI: Temporary Mobile Subscriber Identity

HON: Handover Number

LMSI: Local Mobile Subscriber Identity

MSRN: Mobile Station Roaming Number

Triples (Authorization parameters )

....

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MSC

Mobile services

Switching Center

VLRVisitor Location Register

Tasks:

• Subscriber management in MSC area

• Associated with MSC

• Authentication co-ordination

• commands start of ciphering

Subscriber data:• Subscriber data from HLR (MSISDN, IMSI, services (BS, TS, SS), service restrictions,..) • Temporary subscriber information (LMSI, TMSI, LAI,

IMSI attach/detach, MSRN, HON, triples,...)

Entries valid until re-registration in another VLR!

Fig. 14

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Home Location Register HLR

The Home Location Register HLR is the main data base of the mobile subscriber.The subscription of a user / his subscription data is stored in one HLR only. Theremay be one or more HLRs in a GSM PLMN.

The HLR is always associated with an Authentication Center AC.

The HLR performs the following important tasks:

It sends all necessary data to the VLR.

It supports the call setup in case of Mobile Terminating Calls MTC by sendingrouting information to the Gateway MSC (Interrogation).

It transmits the Triples from AC to VLR on request

An HLR contains different semi-permanent mobile subscriber data, e.g.:


MSISDN: Mobile Station International ISDN number

Bearer Services BS

Tele Services TS

Supplementary Services SS

Restrictions

An HLR contains different temporary information of the mobile subscriber, e.g.:

VLR address

Local Mobile Subscriber Identity LMSI

Mobile Station Roaming Number MSRN

SMS flags

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HLRHome Location Register

Tasks:

• Central storage/management of subscriber data

• Delivery of data to VLR

• Routing information at MTC

• Associated with AC

Subscriber data:

• Semipermanent data: MSISDN, IMSI,

services (BS, TS & SS), service restrictions,...

• Temporary subscriber information: VLR address,

LMSI, MSRN, SMS flags,...

ACAuthentication Center

Tasks:

• Security data storage (Ki)

• Generation of triples (VLR request)

• Associated with HLR

Data / algorithms:

• Ki, IMSI, A3, A8

Fig. 15

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Authentication Center AC

An Authentication Center AC contains all necessary means, keys and algorithms forthe creation of security related authorization parameters, the so-called Triples. TheTriples are created on VLR request and delivered via HLR to the VLR. An AC isalways associated with an HLR.

Central information contained in the AC are:


Ki: Individual Key (top secret mobile subscriber identity)

Algorithms for authentication and encryption: A3, A8.

Equipment Identity Register EIR

The Equipment Identity Register EIR contains the Mobile Equipment identity: theInternational Mobile Equipment Identity IMEI. An IMEI clearly identifies a uniqueMobile Equipment ME and contains information about the place of manufacture,device type and the serial number of the equipment.

EIR are an optional feature in GSM. They have been defined by ETSI to enable theftprophylaxis. They carry out equipment identification functions: monitoring of stolen ornot allowed MEs.

There are three validity lists in EIRs: "white", "gray" and "black" lists for valid, to beobserved and to be blocked equipment.

A Common EIR (CEIR) in Dublin (Ireland) enables the world-wide identification ofstolen mobile equipment.

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EIREquipment Identity Register

Tasks:

• Storage of ME data (IMEI)

• Monitoring of IMEI: "white", "gray", "black" list

ME data:

• IMEI = International Mobile Equipment Identity

site: Dublin

CEIRCommon EIR

Tasks:

• Central, worldwide ME register • Worldwide ME theft prevention

= Type Approval Code TAC + Final Assembly Code FAC (manufacture site) + Serial Number SNR (device serial number)

+ Software Version Number SVR

Fig. 16

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GSM Phase 2+: GPRS

For the introduction of GPRS the GSM PLMN has to be enhanced by:

Gateway GPRS Support Node GGSN

Serving GPRS Support Node SGSN

Packet Control Unit PCU

Channel Codec Unit CCU

HLR Extension

GPRS MS

Serving GPRS Support Node tasks:

serves all GPRS-MS in SGSN area

Routing / Traffic-Management

Mobility Management functions,

e.g. Location Update, Attach, Paging,..

storing Location information

Security & Access Control

collecting charging data

signaling with HLR, EIR, GGSN, MSC

Gateway GPRS Support Node GGSN tasks:

Gateway to PDNs

Protocol conversion

Routing / Traffic Management

Screening / Filtering

Packet Control Unit PCU tasks:

protocol conversion

radio resource management

The Channel Codec Unit CCU enables to transmit using the new Coding SchemesCS-1, CS-2, CS-3 and CS-4.

The HLR has to be extended to include the new type of GPRS subscriber data.

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GGSN:• Gateway to PDNs

• Protocol conversion

Routing / Traffic Management

Screening / Filtering

X.25

Mobile

DTE

SGSNServing GPRSSupport Node

PSTN

InternetIntranet

GGSNGateway GPRSSupport Node

VMSC /

VLRGMSC

HLRextension

GSM Phase 2+: GPRS

ISDN

PCU

BSS

Channel Codec Unit CCU:

BTS-SW upgrade for new

Coding Schemes CS-1... CS-4

HLR Extension::GPRS subscriber data

(GPRS Register GR)

Packet Control Unit PCU:

• protocol conversion

• radio resource management

CCU

SGSN:• serves all GPRS-MS in SGSN area

• Routing / Traffic-Management

• Mobility Management functions, e.g. Location Update, Attach, Paging,..• storing Location information

• Security & Access Control

• collecting charging data

• signaling with HLR, EIR, GGSN, MSC,..

GiG

n

Gb G

r

For simplicity not all GPRS interfaces are shown

Fig. 17

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The Operation SubSystem OSS undertakes operation and maintenance tasks. Thefunctions of the network/ network elements may be centrally monitored and (remote)controlled by the OSS. The control/operation & maintenance locally at each NetworkElement NE (hardware implementation of functional elements) as local operation andmaintenance is distinguished by the central, remote-controlled functionality of theOSS.

The functions of the OSS are performed by so-called Operation & MaintenanceCenters OMC. Depending on the manufacturer, there is sometimes spatial separationbetween the operation & maintenance of BSS and NSS (Siemens: OMC-B and OMC-S).

Important functions of the OSS:

Management and commercial operation (subscriber, mobile equipment, billing)

Sampling of information on network loads (statistical survey) for networkreorganization / optimization

Security management

Network configuration

Remote operation of network elements

Quality checks

Preparation of maintenance work

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MSC/VLR

MSC/VLR

HLR/AC

EIR

NSS

BSC

BTS

BSS

TRAU

WS

• Subscriber and equipment data

management

e.g. clearing services, bills

• Network operation, configuration

& management

• Collecting network load information

& compiling statistics

• Error detection & correction

• Security management

• Performance control

OMCOperation & Maintenance Center


Fig. 18

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Telecommunication Management Network TMN

CCITT guidelines for Telecommunication Management Network TMN (CCITT M.30)designate the OSS as a telecommunication management system.

Seen from TMN level, the GMS-PLMN consists of a certain number of NetworkElements NE.

The TMN configuration of PLMN is ordered hierarchically into three levels:

the lowest level is displayed by a large number of network elements NE of thePLMN

the middle level is realized by a certain number of regional Operation &Maintenance Centers OMC

the highest level is represented by operation systems, such as networkmanagement system, administration management, charging system, nationalOMC, etc.

OSS Telecommunication

Management System

according to

TMN

Network Elements NEs

regional

OMCs

TMN: Telecommunication Management Network

national

OMCs,administration, billing,

network management system,..

Operating systems

Fig. 19

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Procedures

Procedures

Contents

2 Codes & Identities1 8 GSM Security Features2

19 Location Update 3 24 Call Setup / Call Handling4

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1 Codes & Identities

Procedures

Codes & Identities

CGI MCC MNC LAC CI

LAI

CC NDC SNMSISDN

MCC MNC MSINIMSI

X1

X2

X3

X4

X5

X6

X7

X8HLR-ID

Fig. 1

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GSM Service Areas & Codes

The GSM system is hierarchically ordered into service areas. To identify and addressa certain service areas codes are used.

International GSM Service Area

The international GSM service area is the sum of areas being served by GSMnetworks. A GSM subscriber may use all these GSM networks if his HPLMN hasRoaming Agreements with the VPLMN and his ME supports the correspondingfrequency range (GSM900 / GSM1800 / GSM1900).

National GSM Service Area

A national GSM service area contains one or more GSM-PLMN. The PLMN ofdifferent operators may supplement one another or overlap each other.

The following codes are important to identify a national GSM service area:

Mobile Country Code MCC: The MCC consists of 3 digits; it is used e.g. for theInternational Mobile Subscriber Identity IMSI ,Location Area Identity LAI and CellGlobal Identity CGI.

Country Code CC: The CC is the dialing code of the country in which the mobilesubscriber is registered. The CC consists of 2 or 3 digits and is used e.g. in theMobile Subscriber International ISDN number.

PLMN Service Area

A PLMN service area is administered by an operator. Several PLMN service areascan overlap within a country. Thus the individual PLMNs must have a clearidentification:

Mobile Network Code MNC: The MNC is the mobile specific PLMN identification;it consists of 2 digits. The MNC is used in IMSI, LAI, CGI.

National Destination Code NDC: NDCs identify the dialing code of a PLMN; itconsist of 3 digits. The NDC is used in MSISDN.

Network Color Code NCC: The NCC is a PLMN discrimination code that is notunambiguous. It is used as short identity (length: 3 bits) of a particular PLMN inoverlapping PLMN areas or in border regions; it is used e.g. in the Base StationIdentity Code BSIC.

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International

Service

Area Codes

National

MCC: Mobile Country Codee.g.: Aus 505, D 262, Lux 270

CC: Country Codee.g.: F 33, D 49, Lux 352

1 OperatorPLMN

MNC: Mobile Network Codee.g.: D1 01, D2 02, Eplus 03

NDC: National Destination Codee.g.: D1 171, D2 172, Eplus 177

MSC / SGSN „Switch“

Location Area LALA1

LA2

LAC: Location Area Code

LAI: Location Area Identity

Cell CI: Cell Identity

CGI: Cell Global Identity

MSC-Identity

Hierarchy

of GSM

Service Areas

/ Codes

MCC:

CC:

MNC:

NDC:

NCC:

LAC:

LAI:

CI:

CGI:

Mobile Country Code

Country Code

Mobile Network Code

National Destination Code

Network Colour Code

Location Area Code

Location Area Identity

Cell Identity

Cell Global Identity

Fig. 2

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MSC/VLR Service Area

GSM-PLMN are subdivided into one or more MSC/VLR service areas. An attachedmobile subscriber is registered in the VLR, which is associated to his Visited MSC.The MSC/VLR Id. is stored in the HLR, so that an MTC is possible.

Location Area LA

The LA is (in classical GSM) is stored as the most precise information of the(attached) subscribers current location. This information is stored in the VLRassociated to the VMSC. If the MS turns from one LA to another, a Location UpdateProcedure is necessary. The size of a LA is configured by the operator according tothe traffic or population density and the behavior of the mobile subscriber. A LocationArea can encompass one or more radio cells that are controlled by one or more BSC,but never belong to different MSC areas. Location Area identities are:

Location Area Code LAC: The LAC serves to identify a LA within a GSM-PLMN.The LAC length is 2 bytes.

Location Area Identity LAI = MCC + MNC + LAC; the LAI serves as anunambiguous international identification of a location area.

BTS Service Area: the Cell

The cell is the smallest unit in the GSM-PLMN. A defined quality of the receivedsignal must be guaranteed within a cell. If a MS leaves the range of a cell during aconnection, a handover to the next cell is initiated. Cell identifications are:

Cell Identity CI: The CI allows identification of a cell within a location area. The CIlength is 2 bytes.

Cell Global Identity CGI = MCC + MNC + LAC + CI = LAI + CI; the CGIrepresents an international unambiguous identification of a cell.

Base Transceiver Station Identity Code BSIC = NCC + BCC (Base Station ColorCode); The BSIC represents a non-unambiguous short identification (NCC: 3 bit;BCC: 3 bit) of a cell. The BSIC is emitted at a regular rate by the BTS. It enablesthe MS to differentiate between different surrounding cells and to identify therequested cell in a random access.

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National &

PLMN Codes Example*:

Germany

CC = 49

MCC = 262D1

Telekom

D2Mannesmann

Eplus

NDC = 171

MNC = 01

NDC = 172

MNC = 02

NDC = 177

MNC = 03

NDC = 178

MNC = 04

E2Viag Intercom

CC NDC SNSubscriber Number

MSISDNMobile Subscriber ISDN Number

MCC MNC MSINMobile Subscriber Id. No.

IMSIInternational Mobile Subscriber Identity

X1

X2

X3

X4

X5

X6

X7

X8HLR-ID

Subscriber Identities

* This figure has just an illustrative purpose and does not reflect the actual MSC areas of any German PLMN operator.

Fig. 3

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Identifier:

MSC / VLR - Identity

LAI = MCC + MNC + LAC

CGI = LAI + CIMSC / VLR

MSC / VLR

MSC / VLR

MSC / VLR

MSC / VLR

LALA

LA

LA

LA

CellCell

Principle:

MSC, Location

& Cell Area

MCC MNC LAC CI

LAI

Fig. 4

Subscriber Identities:

International Mobile Subscriber Identity IMSI = MCC + MNC + MSIN (MobileSubscriber Identification Number); IMSI length = 3 + 2 + 10 digits. The IMSI is theunique identity of a GSM subscriber. It is used for signaling and normally notknown to the subscriber. Often die first two MSIN digits are taken to specify theusers HLR in the PLMN (operator dependent).

Mobile Subscriber ISDN number MSISDN = CC + NDC + SN. MSISDN length: 2/ 3 + 3 + max. 7 digits = max. 12 digits. The MSISDN is "the users telephonenumber". A user has one IMSI (with one contract), but he can have differentMSISDN (e.g. for fax, phone,..).

Temporary Mobile Subscriber Identity TMSI: The TMSI is generated by the VLRand temporarily allocated to one MS. It is only valid in this MSC/VLR service.When changing to a new MSC area, a new TMSI has to be allocated. The TMSIconsists of a TMSI Code TIC with length 4 bytes. Often the TMSI is used togetherwith the LAI.

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2 GSM Security Features

Procedures

GSM Security Features

Security Features:

• Authentication

• Ciphering

• TMSI allocation

• IMEI check

Fig. 5

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Security Features

In GSM the security of a mobile subscriber is ensured by several features.

Authentication: protects the network operator and mobile subscriber againstunauthorized network use.

Ciphering: is used to prevent eavesdropping of radio communications.

Temporary Mobile Subscriber Identity TMSI allocation: protects thesubscribers identity in the initial access phase, where no ciphering is possible.

IMEI check: prevents the usage of stolen/non-authorized mobile equipment.

Security aspects are described in the GSM Recommendations:

02.09: “Security Aspects"

02.17: "Subscriber Identity Modules"

03.20: "Security Related Network Functions"

03.21: "Security Related Algorithm"

Prerequisites for Authentication and Ciphering

For authentication and ciphering, the Authentication Center AC and the SIM card areimportant; they store the following data:

IMSI (International Mobile Subscriber Identity)

Ki (Individual Key)

A3, A8: Algorithms for the creation of authentication and ciphering parameters

Furthermore, for ciphering the algorithm A5 is stored in the Mobile Equipment. Thisalgorithm can be found in the BTS, too.

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BSS

MSC / VLR

HLR AC

EIR

BTS

NSSSIM

IMSI

Ki

A3, A8

A5

ME

IMEI

IMEIPrerequisites

for Authentication

& Ciphering

Fig. 6

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TriplesCalculation

Random

Number

Generator

A3(Ki, RAND) = SRES A8(Ki, RAND) = Kc

RAND = RANDom numberSRES = Signed RESponseKc = Cipher Key

Data-

base

IMSI

Algorithm

A3

Algorithm

A8

RAND SRES Kc

Triple

ACAuthentication

Center

RAND

KcSRES

Ki

Fig. 7

Triples

The triples are parameters, which are necessary for authentication and ciphering.They are produced in the Authentication Center AC and consist of:

RAND (RANDom number)

SRES (Signed RESponse): the reference value for the authentication

Kc (Cipher Key): key necessary for ciphering.

The calculation of a triple in the AC occurs in the following manner:

For the subscriber with a particular IMSI the reference value of authenticationSRES is calculated by the algorithm A3 from the users individual key Ki and therandom number RAND produced by a random number generator.

The cipher key Kc is calculated by the algorithm A8 from the individual key Ki andthe random number RAND.

RAND, Kc and SRES make together a complete triple.

At the request of the VLR, several triples are generated for each mobile subscriber inthe AC and transferred to the VLR via the HLR on request.

Remark: The individual key Ki is only stored in the AC and the SIM card. Different tothe IMSI and the triples, it is never transmitted through the network. For all signalingprocedures the users IMSI is used.

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Authentication

The authentication checks the real identity of a user, i.e. his authorization to takeaccess to the network. Actually it is checked, whether the secret individual Key Kistored on the SIM card is identically to the one stored for this user in theAuthentication Center or not. The authentication procedure is or can be initiated bythe VLR in the following cases:

IMSI Attach

Location Registration

Location update with VLR change

call setup (MOC, MTC)

activation of connectionless supplementary services

Short Message Service (SMS)

Authentication Procedure

1. the VLR recognizes the need for an authentication; in the case, that no / no moreTriples are available in the VLR it requests a set of Triples from the HLR

2. the Triples are generated in the AC (see above) and sent via HLR to the VLR

3. the VLR sends the RAND to the MS; the SIM card calculates the SRES using Ki,A3 and RAND (see above)

4. the MS sends the SRES back to the VLR; the VLR compares the SRES in thetriple with the SRES calculated by the MS; if they coincide, the network accesswill be authorized and the general procedure will continue, otherwise

5. the access will be refused and the "Authentication Refused" message will be sentto the MS

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requests

triples

sends triples

sends RAND

sends SRES

MS BSS MSC VLR HLR/AC

1

2

3

4coincidence

check

4

5

sends

“Authentication

refused"

55

Authentication

Um

A B D

3

3

4

• Location Registration LR

• LUP with VLR change

• Call Setup: MOC / MTC / SMS

• Activation of connectionless supplementary services

with:

*1

*1 only if no more Triples

available in VLR

*2 only if coincidence

check negative

*2

Fig. 8

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Ciphering

Ciphering regards the security aspects of the information exchange between theMobile Station (MS) and the Base Station (BTS) on the air interface Um. Userinformation (speech/data) and signaling information are ciphered via air interface Um(UL & DL). An exception is given by the initial signaling, before the cipher commandis sent from the network side. At initial signaling data exchange ciphering is notpossible, because the users identities are necessary prerequisite for the generationof ciphering parameters. The cipher command is given after transmission of the useridentity (TMSI / IMSI) and the authentication procedure. Ciphering / Deciphering iscarried out in the BTS and in the MS.

The GSM Recommendation (02.16) of Phase 2 states that up to 8 logically differentencryption algorithms (incl. "no ciphering") should be used. The reason for this is theintention

a) to assign different algorithms to different countries and

b) to provide MS, which do not use the A5-1 algorithm, with the possibility ofroaming in different GSM-PLMN networks.

Currently 3 algorithms are defined:

A5-0: no ciphering for COCOM countries

A5-1: "strict" cipher algorithm (originally MoU algorithm) for MoU-1 countries , A5-1comes from GB; due to military origin (NATO), high security arrangements are tobe regarded

A5-2: "simplified" cipher algorithm for MoU-2 countries (without COCOMcountries);

Remark: A5-0 is implemented in every MS and every BTS to enable access of everyMS in every network. Additionally A5-1 or A5-2 can be implemented.

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Ciphering

• Prevents eavesdropping in Um

• Application in user information and signalling

• Exception: initial signalling

ciphered information

Cipher commandMS BTS

A5 A5

Rec. 02.16: max. 8 cipher algorithms

A5-0: no ciphering; COCOM countries

A5-1: "strict" ciphering; MoU-1 countries

A5-2: "simple" ciphering; MoU-2 countries (except COCOM)

Rec. 02.16: max. 8 cipher algorithms

A5-0: no ciphering; COCOM countries

A5-1: "strict" ciphering; MoU-1 countries

A5-2: "simple" ciphering; MoU-2 countries (except COCOM)

Fig. 9

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Ciphering& Authentication

Authentication:A3(Ki, RAND) = SRES

Ciphering:A8(Ki, RAND) = Kc

A5(Kc,TDMA-No.) = CStext XOR CS = ciphered text

Ciphering:

A5(Kc,TDMA-No.) = CS

text XOR CS = ciphered text

Authentication

& ciphering:generates RAND

A3(Ki, RAND) = SRES

A8(Ki, RAND) = Kc

Authentication:SRES comparison

MS

BTS:

A5

BTS

Umencoded

transmission !

VLR:

IMSI

Triples

AC:

A3, A8,

IMSI,Ki

VLR AC

Triples:

RAND,

SRES, Kc

RAND, KcRAND

ME:

A5

SIM:

A3, A8,

Ki, IMSI SRESSRES

XOR

XOR

plain text

cipher sequence

ciphered text

cipher sequence

plain text 0 1 0 0 1 0 1 1 1 0 0 1...

0 0 1 0 1 1 0 0 1 1 1 0...

0 1 1 0 0 1 1 1 0 1 1 1...

0 0 1 0 1 1 0 0 1 1 1 0...

0 1 0 0 1 0 1 1 1 0 0 1...

CS: cipher sequence

Fig. 10

Ciphering process

Transmitter/receiver must use the same cipher algorithms.

In order to handle ciphering individually for every user, the individual key Ki (stored inthe SIM card and the AC) is used.

The cipher key Kc is transmitted after ciphering from the VLR to the BTS. The MS isable to calculate Kc (after receiving RAND in the authentication procedure) byalgorithm A8 from RAND and Ki.

A 114 bit long cipher sequence is calculated using the cipher algorithm A5, the cipherkey Kc and the TDMA frame number (broadcasted cyclically by every BTS over thecell area).

The speech, data and signaling information are ciphered / deciphered in 114 bit longsequences being connected in a so-called "eXclusive OR" XOR operation.

Deciphering follows exactly the same scheme as ciphering, as the XOR operationyields the original values after double application of XOR (using the same ciphersequence).

To start ciphering, the network sends a cipher start command, which has to beacknowledged by the MS (being the first ciphered information).

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TMSI

Allocation

• Network requires subscriber Id. for call setup

• Id. necessary for triples calculation

• Start of transmission of Id. uncoded

• TMSI prevents eavesdropping of subscriber Id. (IMSI)

• New TMSI with VLR change & usually at call setup

• Network requires subscriber Id. for call setup

• Id. necessary for triples calculation

• Start of transmission of Id. uncoded

• TMSI prevents eavesdropping of subscriber Id. (IMSI)

• New TMSI with VLR change & usually at call setup

MS BSSsends

TMSI

= LAI + TIC

MSC VLR HLR/

AC

IMSI

Ki

Triples

determines

IMSI from

TMSI

TMSI TMSI IMSI

Authentication

Ciphering Triples

new

TMSI

assigns

new

TMSI

stores

new

TMSI

For LA change with MSC/VLR change:

• New VLR identifies old VLR by TMSI

• Subscriber data: query of old VLR

For LA change with MSC/VLR change:

• New VLR identifies old VLR by TMSI

• Subscriber data: query of old VLR

Fig. 11

TMSI Allocation

Ciphering protects the user from being eavesdropped. However, the ciphering withKc requires that the network is aware of the identity of the mobile subscriber withwhom it is in contact. Thus, during the initial phase of communication setup, when theidentity of the mobile subscriber is still unknown, the transmitted signaling informationcan not be ciphered. During this phase a third party may identify a subscriber and thedesired service.

In order to protect the identity of the subscriber in this phase, a temporaryidentification of the subscriber is distributed: the Temporary Mobile SubscriberIdentity TMSI.

The TMSI is used instead of the real user identity, the International Mobile SubscriberIdentity IMSI. This TMSI is allocated by the VLR, which is associated to the VMSC.The MS usually identifies itself with the TMSI in the initial access phase to the VLR.The VLR uses this TMSI to re-identify the IMSI. This is only possible if the TMSI hasbeen allocated by the same VLR. If not, the VLR has to request the VLR, which hasallocated the TMSI to the MS, to deliver the IMSI. Therefore, the TMSI is in mostcases transmitted together with the old LAI, which identifies uniquely a VLR. Therequest VLR - VLR is only possible, if both VLR belong to the same PLMN.Therefore, the IMSI has to be transmitted via Um at the first registration in a newPLMN and obviously at the very first usage of the SIM card (i.e. in the case ofLocation Registrations).

A new TMSI (TMSI re-allocation) can optionally be allocated to the MS after everyauthentication & cipher start (and the optional IMEI check).

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IMEI Check

MS BSS

IDENT_RSP

EIRauthentication

ciphering

IDENT_REQ

IMEI

MSC/VLR

Initiates

authentication

Ciphering

Initiates

IMEI Request

(Identity Request)

Checking

IMEI

(white, grey

or black list)

TACType Approval Code

24 Bit

FACFinal Assembly Code

8 Bit

SNRSerial Number

24 Bit

SVNSoftware Version Number

(spare) 4 Bit

ME

identified

by

IMEI

• Recognise stolen, expired and faulty MEs• Recognise stolen, expired and faulty MEs

Fig. 12

IMEI Check

In contrast to the other security mechanism authentication, ciphering and TMSIallocation, the check of the International Mobile Equipment Identity IMEI is optional. Itdepends on the operators decision whether a EIR is implemented and IMEI checksare done.

IMEI check serves to identify stolen, expired or faulty mobile equipment. A IMEIclearly identifies a particular mobile device and contains information about the placeof manufacture, type approval code and the serial number of the equipment.

The IMEI consists of: Type Approval Code TAC, Final Assembly Code FAC, SerialNumber SNR and a Software Version Number SVN.

If a IMEI check in the PLMN is intended, the Mobile Station MS will be requested tosubmit the IMEI during call setup after authentication and cipher command. The MSsends back its IMEI. The IMEI is routed to the EIR of the PLMN. A check occurs hereto find out whether the IMEI is registered on the black or gray list, i.e. whether the MSis blocked from further use of the PLMN, or whether it has to be observed.

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3 Location Update

Procedures

Location Update

MS

BTS

request

Location Update

Fig. 13

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Location Registration / Location Update

Information of the current location of a mobile subscriber are necessary to built up aconnection to the subscriber, i.e. to start a Mobile Terminating Call MTC. To keeptrack of the users current location the Location Registration / Update procedures areused. Always the MS is responsible to initiate this Location Registration /Update procedures. It informs the network on its current Location Area. TheLocation Area information is stored in the currently responsible VLR. The identity ofthis VLR is stored in the users HLR.

If a MS is "new" in a PLMN a Location Registration is performed. "New" is defined asthe very first usage of a SIM card or a first access after changing the PLMN.

In case of a Location Registration the network needs the IMSI of the MS, becauseeither no TMSI has been allocated before to the MS (in case of first SIM usage) or itis impossible to regenerate the IMSI from the TMSI, because the new VLR is not ableto get into contact with the old VLR (e.g. in case of PLMN changes). After LocationRegistration, in the following Location Updates are used to update the locationinformation in the PLMN. In a Location Update only the TMSI is transmitted via Um.

There are three reasons to perform a Location Update Procedure LUP:

Location Update with "IMSI Attach": If a MS is switched on / off, the network isinformed about the change of the current MS state, i.e. whether to be reachable ornot. Therefore, when being switched on / off, the MS performs an "IMSI Attach" /"IMSI Detach" procedure. The information whether the MS is Attached / Detachedis stored in the VLR. If an "IMSI Attach is performed it is connected with a LUP.

Normal Location Update: Normally a LUP is performed after the MS hasrecognized that it has crossed the boarder between two different Location Areas.The MS is able to recognize the LA change, because it always listens around tothe broadcast information of all cells in its environment, which include the CGI(and so the LAI). If the LAI of the strongest cell changes, a LUP is performed.

Periodical Location Update: A periodic LUP is initiated by a MS at regularintervals. If the VLR does not receive the LUP after a certain time, a "MobileStation not reachable" flag is set.

The LUP is not performed during the duration of a connection. In this case, the LUPis performed after call release.

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LAI =

2620533

MS

BTS

BCCH:

CGI =

26205A64B...

Location

Registration/

Update

• Location Registration: initial MS registration in PLMN

• Location Update

• no LU during connection!

request

Location Update

3 types of Location Update:

• normal

• periodic

• with IMSI attach

Fig. 14

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requests triples

triples

requests LUP,

LR: IMSI

LUP: TMSI

requestssubscriber data;

sends VLR-Id.& LMSI


11

1

2

3

4

5

6

sends data7

sends TMSI =

LAI + TIC

Location Update LUP

77

authentication, ciphering, (IMEI check)

*1


available in VLR

Fig. 15

Location Update Procedure LUP without change of the MSC area

1. The MS recognizes that the LAI has changed. It requests a LUP, identifying itselfwith the TMSI or IMSI. The request and the identity are forwarded to the VLR.

2. The VLR re-identifies the IMSI from the TMSI. If no / no more Triples areavailable in the VLR, it requests triples from the AC via the HLR.

3. The AC generates a set of Triples and delivers them via HLR to the VLR.

4. The VLR stores the Triples and initiates the Authentication, then gives the cipherstart command and initiates an IMEI check (optional).

5. If the Authentication, cipher start and IMEI check are successful, the VLR needsfor call setups the subscriber data. In case of a LR, they are have not beenstored before in the VLR and so they have to be requested from the HLR.Together with this request, the VLR delivers its identity and the information,where this subscriber is stored in the VLR, i.e. the Local Mobile SubscriberIdentity, to the VLR.

6. The HLR stores the VLR identity and LMSI and transmits the requestedsubscriber data to the VLR.

7. The VLR stores the subscriber data and assigns a TMSI (LR: mandatory) or anew TMSI (LUP: only with MSC/VLR change) to the MS. This TMSI is transmittedtogether with the VLRs acknowledgement, that the LUP has been successful, tothe MS. There, the new TMSI and LAI are stored on the SIM card.

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old

VLR

MSC

BSS

new

VLR

MSC

BSS

HLR

AC

Um

LA change

with MSC / VLR change

41

1

16

6

6

3

2

5

7

7

7

Location Update Procedure LUPincl. MSC - VLR change

Fig. 16

Location Update Procedure LUP with VLR change

1. The MS recognizes that the LAI has changed. It requests a LUP, identifying itselfwith the TMSI. The request and the identity (TMSI in combination with the oldLAI) are forwarded to the new VLR.

2. The new VLR receives the TMSI and LAI. It recognizes from the LAI, that theTMSI has been allocated by another VLR (old VLR). Thus, the VLR is not able tore-identify the IMSI from the TMSI and has no chance to request the subscriberdata from the HLR. Therefor, the new VLR calculates the address of the old VLRfrom the LAI and transmits the TMSI to the old VLR and requests it to deliver theusers IMSI. The old VLR delivers the IMSI and the remaining Triples to the newVLR. Remark: If this step 2 is not possible (e.g. line break between old and newVLR) the new VLR commands the MS to transmit the IMSI directly.

3. The new VLR uses the IMSI to calculate the users HLR. The new VLR transmitits identity and LMSI to the HLR and requests the HLR to deliver the subscriberdata and, if necessary, a set of Triples.

4. The HLR stores the new VLRs identity and LMSI, confirms the information,supplies the subscriber data and, if necessary, the Triples.

5. The HLR informs the old VLR to erase the stored data set of this subscriber.

6. The VLR now starts authentication, ciphering and (optionally) IMEI check.

7. The VLR allocates a new TMSI to the MS.

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4 Call Setup / Call Handling

MOCMS starts network access

(PLMN, ISDN, PSTN)

MTCMS is contacted

MMCMS1 starts network access

MS2 is contacted

MICSpecial case MMC:

both MSs in same MSC area

Procedures

Call Setup

Fig. 17

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Call Setup

Different procedures are necessary depending on the initiating and terminating party:

Mobile Originating Call MOC: Call setup, which are initiated by an MS

Mobile Terminating Call MTC: Call setup, where an MS is the called party

Mobile Mobile Call MMC: Call setup between two mobile subscribers; MMC thusconsists of the execution of a MOC and a MTC one after the other.

Mobile Internal Call MIC: a special case of MMC; both MSs are in the same MSCarea, possibly even in the same cell.

Mobile Originating Call MOC

1. Channel Request: The MS requests for the allocation of a dedicated signalingchannel to perform the call setup.

2. After allocation of a signaling channel the request for MOC call setup, includedthe TMSI (IMSI) and the last LAI, is forwarded to the VLR

3. The VLR requests the AC via HLR for Triples (if necessary).

4. The VLR initiates Authentication, Cipher start, IMEI check (optional) and TMSIRe-allocation (optional).

5. If all this procedures have been successful, MS sends the Setup information(number of requested subscriber and detailed service description) to the MSC.

6. The MSC requests the VLR to check from the subscriber data whether therequested service an number can be handled (or if there are restrictions which donot allow further proceeding of the call setup)

7. If the VLR indicates that the call should be proceeded, the MSC commands theBSC to assign a Traffic Channel (i.e. resources for speech data transmission) tothe MS

8. The BSC assigns a Traffic Channel TCH to the MS

9. The MSC sets up the connection to requested number (called party).

Remark: This MOC as well as the MTC described in the following describes only theprinciples of an MOC / MTC, not the detailed signaling flow.

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requests

triples

triples

Setup (Phone No.,..)

Channel Request sends

subscriber Id.

TMSI (IMSI)


identification +

authentication

request

1 2 2

3

3

4

5

requests call

information

6

6

sends info78

9

Setup connection to B-subscriber

Traffic Channel

assignment

commands

channel assignment


authentication + start ciphering + IMEI check + new TMSI

*1


available in VLR

Fig. 18

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Mobile Terminating Call MTC

In the case of a MTC the mobile subscriber is the called party. The MTC call flowdiffers in dependence on the initiating party. In this example the initiating party issubscriber on an external network.

1. After analysis of the MSISDN (CC and NDC) a request to set up a call istransmitted from an external exchange to the GMSC.

2. The GMSC identifies the users HLR from the MSISDN. It starts a so-calledInterrogation to the HLR to get information of the subscribers current location.

3. The HLR identifies the subscribers IMSI from the MSISDN and checks thesubscribers current location, i.e. the VLR address. The HLR informs the VLRabout the call and requests a Mobile Station Roaming Number MSRN (includingthe VMSC address) from the VLR. The request to the VLR includes the LMSI,which enables the fast access to the users data in the VLR.

4. The VLR transmits the MSRN to the HLR, which forwards this number and theIMSI to the GMSC. If the VLR has information, that the MS is Detached currently,the call is rejected / forwarded to the Mailbox.

5. The GMSC uses the MSRN (including the VMSC address) and IMSI to get intocontact with the VMSC.

6. The VMSC requests information (LAI, TMSI) for call setup from its VLR

7. The VLR sends these data.

8. The VMSC uses the LAI to start the Paging procedure. Paging means to searchto MS in the total Location Area (the precise cell is not known).

9. The MS responses the Paging, i.e. from now on its cell is known.

10. This topic includes: Authentication, cipher start, IMEI check and TMSI Re-allocation.

11. The MSC transmits the Setup information to the MS, commands the BSC toallocate a Traffic Channel to the MS and switches through the connection.

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call

requestInterrogation:

MSRN request

sends data

requests data

(LAI, IMSI)

MS

BTS

sends IMSI

requests MSRN

1

10

23

4

5

6

Paging

7

9

8


BTS

BTS

4

sends MSRN5 5

Paging

8

Paging Response9

10 10

connection request

authentication + ciphering + IMEI check + new TMSIcall through switching11 11 11 11

Assignment of Traffic Channel

VLR HLR GMSCVMSC

Fig. 19

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EIR

HLR AC

VLR

VMSC

VLR

VMSC

trafficchannel

BSC

BSC

NSS Network Switching Subsystem RSS Radio Subsystem

Mobile Mobile Call MMC

Mobile Internal Call MIC

BTS

BTS

EIR

HLR AC

VLR

VMSC

BSC

BSC

NSS Network Switching Subsystem RSS Radio Subsystem

BTS

BTS

trafficchannel

Fig. 20

Mobile - Mobile Call MMC / Mobile Internal Call MIC

MMC and MIC are only special cases / combinations of the MOC and MTC.

Mobile Mobile Call MMC

The MMC is a call setup initiated by a MS and terminating at a MS. Thus, MOC andMTC are executed one after the other.

For the call setup of a MMC the same procedures are valid as in the case of MOCand MTC for the call setup between a mobile subscriber and a fixed subscriber. Inthe case of PLMN internal MMC, instead of inquiring the GMSC the VMSC of thecalling party queries the HLR of the called party.

Mobile Internal Call MIC

A special case of the MMC is represented by the MIC. Here, both mobile subscribersare in the same MSC area or even in the same cell. No shortening of the proceduretakes place here. MOC and MTC procedures are executed after each other, the onlydifference is that the MSC involved is VMSC for both, the calling and called party.

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OACSUOff Air Call Set Up

BTS

call setup:

signalingB- subscriber

A- subscriber

MS B-subscriber

answers

B-subscriber

answers

traffic channel

assignment

Not for:

• International calls

• Data connection

• Emergency calls

• Delayed call setup

• No traffic channel assignment until

B-subscriber answers / timer expires

Fig. 21

Off Air Call Set Up OACSU

The OACSU is used in case of overload on the radio interface (a lack of TrafficChannels). It is helpful to overcome short term bottleneck situations without rejectingcall requests.

If there is currently a lack of Traffic Channels OACSU enables to delay the TCHallocation until there is an answer of the called participant. In most cases this willneed several 10 s. There is a high probability that during this time another call isfinished and this TCH is then reserved for the delayed TCH allocation.

OACSU can theoretically be used for MOC and MTC.

In the case of OACSU so-called partial connections are set up. After the TCH is as-signed, the partial connection is completed. The delay of the TCH assignment ismonitored by a timer. When the time frame has run out, a TCH is assigned. TheOACSU can lead to an announcement for the called party, if he/she picks up thephone before the delayed assignment of the TCH.

Restraints for OACSU:

not for international calls

not for data connection

not for emergency calls

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Handover HO: Handover Types

Handover HO are a change of the physical channel during a current connection.There are various types of handover:

Intra-Cell Handover: In the case of Intra-Cell Handover, a physical channel withina cell is changed. A reason for this may be an interference in the frequencycurrently being used. Frequency and/or Time Slot can be changed. Therefore itdiffers from the feature "frequency hopping", in which the frequency is changedafter a certain algorithm, but the time slot is never changed. Frequency hoppingand Intra-Cell Handover exclude each other. The intra-cell handover is realizedinternally in the BSS, i.e. the BSC decides without MSC involvement. Only themessage "handover performed" is sent to the MSC after the handover.

Intra-BSS Handover: An Intra-BSS Handover is carried out between two cells ofthe same BSS. The procedure is decided and performed by the BSC (no MSCinvolvement). The MSC is informed only after the handover ("handoverperformed").

Intra-MSC Handover: An Intra-MSC handover is a handover between two BSSsof the same MSC. The MSC decides about this Handover and switches betweenthe two BSCs.

Inter-MSC Handover: A Inter-MSC Handover include at least two MSCs. TheMSC has to decide and to switch. Inter-MSC handovers are one of the mostcomplicated GSM procedures, in particular in the case of MSCs made by differentmanufacturers. One has to distinguish between "Basic Inter-MSC Handover" and"Subsequent Inter-MSC Handover".

Basic Inter-MSC Handover: If a MS changes for the first time from the area of anMSC (A) to the area of a MSC (B), this is described as Basic Handover.

Subsequent Handover: If the MS also leaves the MSC (B) area and moves into thearea of a further MSC (C) or returns to the area of the old MSC (A), this follow-onhandover is called Subsequent Inter-MSC Handover. The handover is controlledby the initial MSC, which is called MSC (A) = Anchor MSC. In a Subsequent Inter-MSC Handover with MSC (C) for a short time three MSCs are connected for onecall. The connection MSC (A) - MSC (B) is released after successful set up ofconnection between MSC (A)and MSC (C).

The Anchor MSC is responsible for billing. This is the reason, why Inter-PLMNHandover, i.e. Handover between different PLMNs are normally not performed.

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Handover Types

Intra-cell

BSCBTS

f 1, TS 1

f 2, TS 2

Intra-BSS

BSC

BTS

BTS

MSC

Handover

performed

Intra-MSC

MSC

BSS

BSS

Inter-MSC

MSC - BMSC - A

MSC - C

basic

subsequent

MSC

Handover

performed

Fig. 22

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Handover Decision

The handover algorithm is based on periodically measurements of MS and BTSconcerning the strength and quality of the received signals. The MS measures qualityand strength of the connection and the strength of the serving BTS and that of thesurrounding BTSs. The BTS measures quality and strength of the connection as wellas the distance MS - BTS (Timing Advance TA).

The result of the MS measurements is transmitted to the BTS. The BTS adds its ownmeasurements and transmits the data as "Measurement Report" to the BSC.

The BSC has to decide, whether a handover is necessary or not. The decision isdetermined by the comparison between the current measured values and thethreshold values. If no threshold values are exceeded, the BSC analyses whether another BTS as the current one would enable a better air interface quality. Differentother aspects have to be taken into account, e.g. the current load of the cells.Furthermore, so-called "Ping-Pong Handover" should be prevented.

If an Inter-cell handover is initiated, the criterion of availability of surrounding cells isused to set up a list of suitable handover destinations in a declining order of priority.This list forms the basis for the final handover decision that is carried out by the BSC(in case of Intra-BSS Handover) or by the MSC (in case of Inter-BSC / -MSCHandover).

Handover criteria are e.g.:

Strength of the received signal (UL and DL)

Quality of the received signal (UL and DL)

Distance MS - BTS (Timing Advance, UL)

Signal strength of suitable surrounding cells (UL, BCCH)

Interference that decrease the signal quality (UL and DL)

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Measurement:

connection quality & strength:

strength of serving BTS &

surrounding BTSs

HandoverDecision

MS

Measurement:

connection quality & strength,

distance measurement (TA)BTS

Measurement report

Timing Advance,Power control

BSC

HO

decision

Measurement value processing

(averaging, limit values,..)

Evaluation list

(suitable BTSs for HO...)

Initiation of HO type

HandoverBSC/

MSC

Measurementreport

Fig. 23

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BTS

BTS

BTS

BTS

BTS

BTS

BTS

MSC (A)

VLR

Handoverexample

MSC (B)

VLR

BSC

BSC

BTS

Level:cell Acell B

cell C

BTS

BSC to MSC (A):

HO please!

cell B

MSC (B)

A

B

C

1. BSC: HO necessary

2. Parallel connection setup

3. MS changes phys. channel

4. Original connection released

Fig. 24

Handover Example: (Basic) Inter-MSC Handover

1. During an existing connection, the MS permanently measures the quality andpower level of the received information and measures the strength of its own andthe surrounding BTS. Furthermore, the BTS measures the quality and strength ofthe connection and the Timing Advance. The results are as measurement reportto the BSC. The BSC analyses the need for Handover. If an Handover isnecessary, the BSC creates a list of preferable cells to which the Handovershould be performed. If an Handover to a cell of another BSC / MSC isnecessary, the information is forwarded to the MSC (A). In this example, aHandover from Cell A to Cell B is preferable. On basis of the BSC information,the MSC (A) decides to initiate a Basic Inter-MSC Handover to MSC (B),because Cell B is in the service area of MSC (B).

2. MSC (B) requests the BSC, which is responsible for Cell B to allocate resourcesfor this connection and prepare network transmission capacities for the call. Asecond connection is built up parallel to the existing connection. The DLinformation is split and delivered to both BTS.

3. MSC (A) gives command to the MS (via BSC) to change the physical channel.Changing the physical channel, the MS immediately is connected to Cell B.

4. The initial connection is released, the resources are set free for otherconnections. The users data are still transmitted via MSC (A); it is the Anchor-MSC.

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Emergency

Call

call setup:

Emergency Call

Center

MS

without:

• Authentification

• Ciphering

• IMEI check

• TMSI-Reallocation

Emergency call:• Priority treatment

• no security features

• fast call setup

• usually always possible,

even without valid SIM card

MSC

• Direct connection

• Supplies location info

S O S

Fig. 25

Emergency Call

The connection set up for the Tele Service "Emergency Call" is similar the that of theMobile Originating Call MOC.

The mobile subscriber starts this service either by pressing a SOS key or by dialingan emergency service number (often: 112).

The setup follows the MOC signaling flow. Differences are:

no Authentication is necessary

no Ciphering will be used

no IMEI check is performed

no TMSI Re-allocation is performed

A short call setup is resulting in this lack of security features. Furthermore, theEmergency Call should always be possible with any MS, even without a valid SIMCard.

Emergency calls are treated with precedence. This may also lead to the release ofother existing connections.

The BSS always delivers the location of the emergency call to the MSC. Dependingon this origin, the emergency connection is then transmitted from the MSC to theregionally responsible Emergency Call Center. The available location information canbe delivered to the Emergency Call Center, too (operator dependent).

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s

Short Message Service SMS transmission (MT-SMS)

MS attached (i.e. reachable):

A Short Message Service Center SM-SC (out of the scope of the GSM Rec.) triesto transmit the SMS to the requested MS via GMSC.

The GMSC performs an Interrogation to the HLR to get knowledge about thecurrent VMSC.

The HLR requests the VLR for an MSRN and forwards this to the GMSC.

The GMSC gets into contact with the VMSC and the SMS is delivered to the MS.Different to the MOC, no Traffic Channel allocation is necessary in case of SMStransmission. The SMS can be transmitted via Signaling Channel.

MS Detached (not reachable):

The SM-SC tries to transmit the SMS to the requested MS via GMSC.

The GMSC performs an Interrogation to the HLR to get knowledge about thecurrent VMSC.

The HLR requests the VLR for an MSRN. This is not possible, because thesubscriber is Detached and the VLR stores this information.

In the following, a SMS flag is set in the VLR and in the HLR. Furthermore, theHLR stores the address of the SMS-SC.

The HLR informs the GMSC that the SMS can not be delivered and the GMSCrejects the request of the SM-SC. The SMS is still stored in the SM-SC.

If the MS is switched on again, an IMSI Attach procedure is performed to the VLR.

Due to the SMS flags, the VLR informs the HLR, that the MS is reachable again.

The HLR requests via GMSC the SM-SC to start the SMS transmission again.

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SMS-

GMSCSM-SC

SMS Service CenterVMSC

VLRHLR

MS

GSM-PLMN

SMS /SMS-SC

HLR-flag+ SM-SC Id(s)

MS Detached • no SMS delivery possible• SMS stored in SM-SC

• flag in VLR & HLR

IMSI Attach • VLR informs HLR

• HLR requests SM-SC via SMS-GMSC to retransmit SMS

MS Detached • no SMS delivery possible• SMS stored in SM-SC• flag in VLR & HLR

IMSI Attach • VLR informs HLR

• HLR requests SM-SC via SMS-GMSC to retransmit SMS

VLR-flag

Fig. 26

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Radio Interface

Radio Interface

Contents

2 Physics of Layer 11 14 Logic of L12 25 MOC / MTC 3

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Radio Interface Siemens

1 Physics of Layer 1

Physics of Layer 1

TDMAframe

4.615ms

time

TS4

TS5

TS6

TS7

TS0

TS1

TS2

TS3

Frequency[MHz]

••• •••

Duplex distance: 45 MHz

200 kHz

Example:GSM900

890 915 935 960

UL DL

TS577s

Physical channel (Um)Physical channel (Um)

Radio Interface (Layer 1)

Fig. 1

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The Radio Interface: Physics of Layer 1

The Layer 1 of Um is described in GSM Rec. 04.04. In the following, L1 is separatedfor didactical reasons in the “Physics of L1” transmission and the “Logic of L1”transmission.

For the transmission of user data / signaling physical channels are allocated to theusers. A physical channel in GSM is defined by a frequency pair for UL/DL and aTime Slot TS of the TDMA frame. The frequency bandwidth in GSM is 200 kHz. ATime Slot TS has a duration of 0.577 ms. 8 TS form a TDMA frame; the duration of aTDMA frame is 4.615 ms.

The Burst

In GSM, using FDMA & TDMA for multiple access, the transmission of data is notcontinuously. In every Time Slot TS the HF has to be switched on, the data aretransmitted briefly and then the HF transmission is switched off again. This type ofHF transmission is called “pulse” or “bursty” operation. Therefore, the content of a TSis called “Burst”.

The transmitter is only allowed to transmit the HF Burst within the duration of the TS.If the HF transmission exceeds the duration of the TS, the transmission mightinterfere with the transmission of the succeeding user. In this case, strongdisturbances of both connections follow. For this reason, the transmission must betimed exactly. Furthermore, it is not possible to switch on / off immediately. Toprevent interference between neighboring TS, the GSM Rec. define a duration duringwhich the switching process must be closed. The BS and MS must be able to switchthe HF power on / off within 0.028 ms over a wide dynamic range. This range is 70dB for BS and 36 dB for MS.

So the burst transmission can be explained as a maximum of 0.028 ms for switchingon HF to the necessary power level, 0.5428 ms for the HF transmission of the so-called “useful part” (corresponding with 147 bit) and 0.028 ms for switching off the HFpower level down to “background noise” level. Note: This “useful part” + flanksexceeds the duration of a TS (0.577 ms) and often irritate readers of GSM literature.The 0.028 ms are however only time maximum limits for the flanks. They carry novaluable information and so they are allowed to interfere with the succeeding Burstsin a negligible way.

3


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Power

Time

28 s 28 s 542,8 s

The Burst

„Useful part“„Useful part“

Fig. 2

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Burst: Content

7 0 1 2 3 4 5 6 7

TBTail Bits

3 bit

“Information”142 bit

TBTail Bits

3 bit

GPGuard Period

8.25 bit

HF transmission

TS = 576 12/13 s

= 156.25 bit

1 bit = 3.6923 s

Fig. 3

Burst: Content

A Time Slot is defined as a duration of 0.577 ms (to be precise: 0.576923 ms). Thisduration is divided per definition into 156.25 bit. This means an individual bit has aduration of 3692.3 ns.

The 156.25 bit are used / defined as follows:

142 bit for the transmission of “Information” (not only users data / signaling but alsocontrol information necessary for maintenance of the connection)

3 bit as Tail Bits TB for edge limitations of the TS. They are preventing, that usefulinformation are “falling” into the flanks of the burst. TB contain no useful information.They are modulated as content “0”.

8.25 bit as Guard Period GP. The GP is not part of the HF transmission. It is used tocompensate run-time effects in the cells. Note: There is one exception of GPs: Thefirst MS transmissions of the MS toward the network use special bursts (AccessBurst) with an extended GP of 68.25 bit.

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TB Information-Bits STraining

SequenceS TB GPInformation-Bits

156.25 Bit = 576.9 s

3 57 1 26 1 57 3 8.25

Normal Burst 5 Burst Typeswith different logical content(discussed later-on)

5 Burst Typeswith different logical content(discussed later-on)

Example:

Normal Burst

Bit

S: Stealing flagTB: Tail BitsGP: Guard Period

142 bit “Information”

Fig. 4

Example: Normal Burst NB

The Normal Burst is part of the “Logic of Layer 1” and will be explained together withthe four other Burst Types later-on in detail. It is shown here for didactical reasons toget an idea of the content of what has been determined as “Information”.

The 142 bit of “Information” (content: “0” or “1”) are realized in the middle of the burstto enable reliable transmission. The 3 TB (content: “0”) on the edge provide bufferagainst data loss at the flanks of the burst.

The Normal Burst NB contains:

2 x 3 bit as Tail Bits TB

2 x 57 bit as Information (User Data / Signaling)

2 x 1 bit as “Stealing Flags” which inform the receiving side if user data or userrelated signaling is transmitted

26 bit as Training Sequence for time synchronization and transmission qualityanalysis

Now the structure of a TS / burst is explained, the content has been described downto bit level, but the question is now:

How are the “0” and “1” physically presented on the radio interface?

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1

0

fT - f

fT + f

fT

+ 180°

+ 90°

t

- 180°

- 90°

phase

binary

signal

frequency

Minimum Shift

Keying MSK

GMSK: Gaussian MSKMSK signal x Gaussian curve

smaller band-width

Fig. 5

GSM Modulation: Gaussian Minimum Shift Keying

For transmission of the binary data “0” and “1” in GSM a frequency modulationmethod has been chosen. It is known as Gaussian Minimum Shift Keying GMSK.

Minimum Shift Keying MSK

The GMSK is based on Minimum Shift Keying MSK. MSK is a modulation principle,where the information is transmitted in the instantaneous frequency of the HF signal.

The carrier frequency fT is shifted by the frequency difference f = 67.7 kHz toindicate "1" or "0". This is achieved not by shifting the frequency directly, but by achange of the phase velocity. This results in a frequency and phase variation.

Gaussian MSK

In GMSK, the phase transitions are smoothed by filtering the data with a gaussiancurve. This enables smooth phase shifts, keeping the bandwidth comparable narrow.Thus, a bandwidth of only 200 kHz can be achieved.

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Frames

TDMA frames

A single frequency band in TDMA systems is subdivided into several Time Slots TS,which can be used by different users. In GSM 8 TS form one TDMA frame (4.615ms), i.e. 8 physical channels are using the same frequency band being cyclically(every 4.615 ms) allocated to a certain user / application.

So the TDMA frame is a repetition cycle with a duration of 4.615 ms.

The TDMA frames themselves are again part of a repetition cycle of a larger duration.Certain contains are always repeated after a certain duration. This repetition cycle iscalled: Multiframe.

Multiframes

Here a separation has to be done according to the type of information a physicalchannel is transmitting. The physical channels can be used to transmit either userdata or signaling.

Multiframes of physical channels allocated for user traffic (Traffic Channels TCH) arerepetition cycles of 26 TDMA frames.

Multiframes of physical channels allocated for signaling data (mostly on one / severalof the TS0 of the carrier of one cell) are repetition cycles of 51 TDMA frames.

Certain “logical contents” are repeated on certain TDMA frames of the 26 TDMAframes of the TCH Multiframes or on the 51 TDMA frames of the signalingMultiframe.

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TDMA

frame

Frames

RFC

3

RFC

2

RFC

1

0

1

2

3

4

5

6

70

1

2

3

4

5

20

21

22

23

24

25

43

44

45

46

47

48

49

50Time

RFC

124

Frequency

0

1

2

3

4

5

6

7Time

FDMA

User

Traffic

Signaling

cyclical repetition

of certain contents

cyclical repetition

of certain contents

• TDMA-

• Multi-

• Super-

• Hyper-

Frames

Multi-

Frames

Fig. 6

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26 TDMA frame = 120 ms

Full Rate (FR) TCH

T/t = TDMA frame for TCH

A/a = TDMA frame for SACCH/T

I = Idle

2 Half Rate (HR) TCH

Example:

TCH Multiframe

T T T T T T T T T T T T A T T T T T T T T T T T T I

T t T t T t T t T t T t A t T t T t T t T t T t T a

Signaling Multiframe

Logic of L1

User related control data

to maintain connection

TCH: Traffic Channel

SACCH: Slow Associated Control Channel

Fig. 7

Example: Traffic Channel TCH Multiframe

The TCH Multiframe consists of 26 TDMA frames with user data. Every one of this 26TDMA frames contains a certain “logical content”. So certain contents are repeatedevery 120 ms. This is necessary because the “user data” which are transmitted onthis Traffic Channel are not only the user information (Traffic Channel TCH = userspeech, fax, data) which he likes to transmit. Also user related control information(so-called Slow Associated Control Channel SACCH) which are necessary tomaintain the connection are transmitted on the same physical channel. They aretransmitted every TCH Multiframe, i.e. every 120 ms on the 13th TDMA frame (FullRate TCH), respectively at Half Rate transmission for the first user of this physicalchannel on the 13th and for the second user on the 26th TDMA frame.

In Full Rate transmission the 26th TDMA frame is empty (Idle I).

A general overview and description of the different “logical contents” which aredefined in GSM and the content of the Signaling Multiframes is given later-on in“Logic of L1”.

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Time Structure of the Radio Interface:

Bit: The shortest unit of the GSM radio interface is one bit. Its information is GMSKmodulated onto the HF. Its duration is 3692.3 ns.

Time Slot TS: The TS consists of 156.25 bit. It is the shortest possible transmissiontime in GSM with a duration of 0.57688 ms.

TDMA frames: 8 TS form 1 TDMA frame with a duration of 4.615 ms. 8 physicalchannels are using the same frequency band being cyclically (every 4.615 ms)allocated to a certain user / application.

Multiframes: The TDMA frames themselves are again part of a repetition cycle of alarger duration, the Multiframe. Certain contains are always repeated after a certainduration. Multiframes for user traffic (Traffic Channels TCH) are repetition cycles of26 TDMA frames with a duration of 120 ms. Multiframes for signaling are repetitioncycles of 51 TDMA frames with a duration of 235.4 ms.

Superframe: The TCH / Signaling Multiframes are summarized in longer repetitioncycles to Superframes. Superframes consist of 51 TCH / 26 Signaling Multiframes. ASuperframe (1326 TDMA frames) is the smallest common multiple of TCH andsignaling Multiframes with a duration of 6.12 s.

Hyperframe: The Hyperframe is the GSM numbering period. It comprises 2048Superframes and is exactly 12,533.76 s or 3 h 28 min 56.76 s long. It is a multiple ofall cycles described up to now and determines all transmission cycles / periods onthe radio interface. The Hyperframe is the shortest cycle for repetition of thefrequency hopping algorithm and for ciphering.

1 Signalling Multiframe =

51 TDMA frames 235,4 ms

1 TCH Multiframe =

26 TDMA frames = 120 ms

Time

Structure

Hyperframe =

2048 Superframes 3h 29 min

0 1 2 3 4 5 6 7

0 1 2 3 24 25 0 1 2 3 49 50

0 1 2 3 4950

0 1 2 3 24 25

1 Superframe =

51 x 26

TDMA frames

6.12 ms

Numbering Periode.g. repetition of • frequency hopping

• ciphering

Channel organisationscheme

Repetition scheme

for TCH / Signaling

BURST = TS content

1 TDMA frame

= 8 TS = 4,615 ms

1 Burst = 156,25 bit = 576,88 us

(1 bit = 3,6923 us)

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Adaptive frame alignment:preventing simultaneous

transmission / receiving

UL/DL shifted by 3 TS

Adaptive frame alignment /

Timing Advance TA

76543210

76543210 DL

UL

Timing Advance TA: compensation of propagation delays

BTS commands MS to transmit earlier:

2 x propagation time MS - BTS

Fig. 9

Adaptive Frame Alignment

In GSM the numbering of the Uplink UL and Downlink DL Time Slots TS is shifted bythree TS against each other. This prevents simultaneous transmission and receptionin GSM and enables to create simpler and cheaper Mobile Stations MS. Narrowbandfilters are not necessary. This enabled to built GSM handhelds directly fromcommercial start of GSM in the early 90th.

Timing Advance TA

The Guard Periods GP of the Normal Bursts are not able to compensate signaldelays in larger GSM cells. The MS receives synchronization signals from the BS,synchronizes their transmission based on this signals, but it cannot recognize itsdistance from the BS. The distance can be up to 35 km in a normal GSM cell. Atransmission without special compensation of this run-time delay would result ininterference with the succeeding TS.

Therefore, the BS analyses the delay of the MS transmission using the very first MSburst (which has an extended GP). The BS adjusts its transmission in the DL andinforms the MS with the Timing Advance TA information how to adjust the ULtransmission (i.e. how much earlier the transmission has to start). Over the totalconnection, the delay is analyzed by the BS and new TA values set for the MS. 64TA values (difference: plus/minus 1 bit period) can be used to compensate run-timeeffects.

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Frequency Hopping

Frequency Hopping means to change the frequency used for transmission isconsequently changed every TDMA frame following a certain frequency hoppingalgorithm. The Time Slot of the physical channel is still fixed.

The logic behind frequency hopping is to guarantee that all channels have the samehigh degree of transmission quality by dividing possible short term interference overall channels of the cell.

So a narrow-band interference does not disrupt the total transmission on one carrier,i.e. on one frequency band, because the transmission is hopping from TDMA frameto TDMA frame to other frequencies.

Nevertheless, now interference occurs for all the carrier of the cell from time to timewhen transmitting on the disturbed frequency band. But this can be compensated inGSM, because in classical GSM there is always redundancy on the transmitted data.The redundant information is delivered in the next TS of the succeeding TDMAframe, i.e. on another frequency (which is not disturbed).

Frequency hopping is optional in GSM. It is on the PLMN operators decision to usefrequency hopping or not. Frequency hopping significantly improves the quality /reliability of transmission.

The carrier transmitting the Broadcast Control Channel BCCH (carrying informationnecessary for MS synchronization to the network) does not participate in frequencyhopping.

Frequency hopping is done in the MS and BS, managed from the BSC. Thefrequency hopping algorithm can be configured from an OMC.

Frequency Hopping

frame 0 frame 1 frame 2 frame 3 frame 4 frame 5

RFC 1

RFC2

RFC 3

RFC 4

RFC 5

TCH

Compensation of

narrow-band interference

stable & reliable transmission(redundant bits on different TDMA frames)

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2 Logic of L1

Signaling

Traffic

User Data

DL

DL

UL

UL + DL

DL

UL+

BCCH

FCCH

SCH

PCH

AGCH

RACH

SDCCH

SACCH

FACCH

TCH/F

TCH/H

BCHBroadcast Channel

CCCHCommon Control

Channel

DCCHDedicated Control

Channel

Logic of L1


Fig. 11

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Logical Channels

Different signaling and user data contents determine different Logical Channels inGSM.

For user data transmission two different Logical Channels are used:

TCH/F Traffic Channels, Full rate (FR/EFR speech: 13 / 12.2 kbit/s; data: 9.6kbit/s)

TCH/H Traffic Channels, Half rate (HR speech: 5.6 kbit/s; data: 4.8/2.4/1.2/0.6/0.3kbit/s)

For signaling 3 types of Logical Channels are used: BCHs, CCCHs and DCCHs.

Broadcast Channels BCH are used DL only for MS synchronization & information:

FCCH Frequency Correction Channel: for MS frequency synchronization

SCH Synchronization Channel: for MS time synchronization; contains additionallyTDMA frame no., BSIC

BCCH Broadcast Control Channel: contains system & cell parameters, e.g. CGI(i.e. PLMN, LAI), channel combining, frequency hopping algorithm, cipher mode,cell capabilities: e.g. EFR/FR/HR, VAD/DTX, ASCI, HSCSD, GPRS, EDGE,..)

Common Control Channels CCCH are used uni-directional UL & DL for initialaccess:

PCH Paging Channel: to search the MS in the LAI in case of an MTC

RACH Random Access Channel: MS request for dedicated signaling resources

AGCH Access Grant Channel: to grant a dedicated channel to the MS

Dedicated Control Channels DCCH are used bi-directional for dedicated signaling:

SDCCH Stand-alone Dedicated Control Channel: dedicated signaling between MS& BS for Call Setup (Authentication, Cipher start, IMEI check, TMSI-Reallocation,Setup,..) LUP procedures, SMS

SACCH Slow Associated Control Channel: allocated together with SDCCH orTCH; control information to maintain connection (e.g. DL: Power Control, TimingAdvance, Comfort Noise; UL: Measurement Reports for Handover,..)

FACCH Fast Associated Control Channel: allocated instead of TCH in case ofenhanced demand for signaling resources (Handover, Call Release, IMSI-Detach,OACSU..)

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Allocation of signaling channel

Signaling MS BTSE for e.g. Call Setup (Authentication, Cipher start, IMEI check,Setup info,..) LUP, SMS,...

Signaling

Traffic

User Data

DL

DL

UL

UL + DL

DL

UL+

FCCH

SCH

PCH

AGCH

SDCCH

SACCH

FACCH

TCH/F

TCH/H

Frequency synchronization

Time synchronization + BSIC, TDMA-No.

Paging / Searching (MTC)

Measurement Report,

TA, PC, cell parameters,...

Signalling instead of TCH(e.g. for HOV, IMSI Detach, Call Release)

BCHBroadcast Channel

CCCHCommon Control

Channel

DCCHDedicated Control

Channel

User data Full Rate

Logical channels

User data Half Rate

BCCH: Broadcast Control Channel

FCCH: Frequency Correction Channel

SCH: Synchronisation Channel

PCH: Paging Channel

AGCH: Access Grant Channel

RACH: Random Access Channel

SDCCH: Stand-alone Dedicated Control Channel

SACCH: Slow Associated Control Channel

FACCH: Fast Associated Control Channel

TCH: Traffic Channel

BCCHCGI, FR/EFR/HR, VAD/DTX, HSCSD,frequency hopping, channel combinations

RACH Request for signaling channel

Fig. 12

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Burst Types

The HF transmission, which is transmitted in a Time Slot with a pre-defined bitsequence is call Burst. In GSM there are 5 different Burst types defined:

Normal Burst NB: The NB is used for most of the Logical Channels (TCH, BCCH,PCH, AGCH, SDCCH, SACCH, FACCH). It consists of the following bit sequence:

2 x 3 bit as Tail Bits TB for edge limitation of the HF burst (content: “0”),

2 x 57 bit as Data Bits (Information), which carry the users data or signalinginformation.

2 x 1 bit as Stealing Flags S, which indicate whether user data (TCH) or userrelated signaling (FACH) is transmitted in this Burst.

26 bit as Training Sequences, which are fixed bit pattern (8 different sequencesexist for NB) for synchronization of the transmitted burst & recognition oftransmission quality

8.25 bit as Guard Period GP, which is not part of the HF transmission; used asguard period between succeeding TS.

Frequency Correction Burst: It is used for the FCCH only, consisting of:

142 Fixed Bits with content “0”; it is used for MS frequency synchronization

2 x 3 bit as Tail Bits

8.25 bit Guard Period

Synchronization Burst: It is used for the SCH only, consisting of:

64 bit as Training Sequence for initial precise MS time synchronization

2 x 39 bit with Information necessary for initial MS access (BSIC, TDMA framenumber, NB training sequence used in this cell,..)

Random Access Burst: It is used for RACH only, consisting of:

36 bits Information for initial access (BSIC, MS random no., access reason)

41 bits as Synchronization Sequence

8 + 3 bits as Tail Bits

68.25 bits Guard Period GP; the extended GP prevents interference with thesucceeding TS occurring due to the run-time problem (the MS lacks of informationabout its distance to the BS before starting access)

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Dummy Burst: The Dummy Burst has NB structure; it is transmitted in special casesif nothing else (useful) is to be transmitted (e.g. at the BCCH carrier, which has to betransmitted continuously because it is the cell beacon).

TB

3bit

Information57 bit

S1bit

Training

Sequence26 bit

S1bit

TB

3bit

GP

8.25bit

Information57 bit

156.25 Bit = 576.9 s

Normal Burst TCH, BCCH, PCH, AGCH, SDCCH, SACCH, FACCH

Burst Types

TB

3bit

Fixed bits142 bit

TB

3bit

GP

8.25bit

Frequency Correction Burst: FCCH

TB

3bit

Information39 bit

Training

Sequence64 bit

TB

3bit

GP

8.25bit

Synchronization Burst: SCH

TB

8bit

Synchronization

Sequence41 bit

TB

3bit

GP

68.25bit

Information36 bit

Random Access Burst: RACH

Information39 bit

Dummy Burst: Structure Normal Burst

Fig. 13

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Multiframe: Channel Combinations

There are seven different schemes to co-ordinate the logical channels in multiframes.Three schemes are used for the co-ordination of Full rate and Half rate TrafficChannels. Four schemes are used to co-ordinate signaling, depending on therequirements of the individual cell. The network operator has do decide, whichchannel combinations are used for a cell.

Combination I – III are used for TCH Multiframe co-ordination (Full rate / Half rate).

Combination IV – VII are used for Signaling Multiframe co-ordination.

Combination I: TCH/F + FACCH/F + SACCH/F

Combination I is used to transmit Full rate user data & speech. The frames 0–11and 13-24 are used for user data, frame 12 is used for SACCH (user relatedcontrol data) and frame 25 is not used (I: Idle).

Combination II & III: TCH(0,1) + FACCH/H(0,1) + SACCH/H(0,1) respectivelyTCH/H(0) + FACCH/H(0) + SACCH/H(0) + TCH/H(1) + FACCH/H(1) + SACCH/H(1)

Combination II & III are used to transmit Half rate user data & speech. 2 TCH/Huser have to share the 26 multiframes. Data from user 1 or user 2 are filledalternately into the frames. The SACCH of user 1 is on frame 12, the SACCH ofuser 2 is on frame 25.

Combination IV: FCCH +SCH + CCCH (PCH & AGCH) + BCCH

Combination IV offers much space for the Common Control Channels CCCH.Therefore, this combination is used often for cells with many carrier. As BCCHcarrier it is the cell beacon and so it must be used exactly only on one carrier ofthe cell. It is allocated on TS 0 of this carrier and has to be transmittedcontinuously. If no useful information is to be transmitted, Dummy Bursts have tobe used. There is no Power Control used on the cells beacon. Combination IVlacks of dedicated signaling channels (SDCCH and SACCH). Therefore, it has tobe used together with combination VII.

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I) TCH/F + FACCH/F + SACCH/F

II) TCH/H(0,1) + FACCH/H(O,1) + SACCH/H(0,1)

III) TCH/H(0) + FACCH/H(0) + SACCH/H(0) +

TCH/H(1) + FACCH/H(1) + SACCH/H(1)

IV) FCCH + SCH + CCCH + BCCH

V) FCCH + SCH + CCCH + BCCH + SDCCH/4 + SACCH/4

VI) CCCH + BCCH

VII) SDCCH/8 + SACCH/8

Multiframe: Channel Combinations

F

0

S

1

BCCH

2 - 5

CCCH

6 - 9

F

10

S

11

CCCH

12 - 19

F

20

S

21

CCCH

22 - 29

F

30

S

31

CCCH

32 - 39

F

40

S

41

CCCH

42 - 49

I

50

F:FCCH

S:SCH

B: BCCH

R

0

R

1

R

10

R

11

R

20

R

21

R

30

R

31

R

40

R

41

R

50

DL

UL

Combination IV

C: CCCH (PCH, AGCH)

I: Idle

R: RACH

TCH-Combinations

shown before

TCH-Combinations

shown before

Fig. 14

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Combination V: FCCH + SCH + CCCH + BCCH + SDCCH/4 + SACCH/4

Combination V is the minimum configuration for a cell, because is contains alllogical channels necessary for signaling in a cell. It is often used for cells with onlyone or two carrier. For combination V the same is valid as for combination IV: It isthe cell beacon, it must be allocated on TS 0 of exactly one carrier of the cell. Ithas to be transmitted continuously. SDCCH/4 and SACCH/4 means that thiscombination offers the capacity for 4 simultaneous dedicated signalingconnections.

Combination VI: CCCH + BCCH

Combination VI can be used together with combination IV and VII for cells withvery much traffic and many carriers (up to 16 carriers). This means to be anincreased demand for Common Control Channels, which are offered bycombination VI. The multiframe structure of combination VI is similar as thestructure of combination IV, without FCCHs and SCHs. In combination with IV,combination IV is allocated on TS0 on the carrier and VI combinations can beallocated at TS 2 / 4 / 6 depending on the traffic volume of the cell.

Combination VII: SDCCH/8 + SACCH/8

Combination IV and VI offer no dedicated signaling channels. Therefore, they haveto be used together with combination VII. Combination VII offers up to 8simultaneous dedicated signaling channels. Combination VII can be allocated onTS 0 of other carrier than the BCCH carrier. The BCCH indicates the allocation ofcombination VII.

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Signaling Multiframe: Combination V

F S F S F S IBCCH CCCH CCCH CCCHSDCCH

0SDCCH

1 F SSDCCH

2SDCCH

3 F SSACCH

0SACCH

1

F S F S F S IBCCH CCCH CCCH CCCHSDCCH

0SDCCH

1 F SSDCCH

2SDCCH

3 F SSACCH

2SACCH

3

SDCCH

0SDCCH

1SDCCH

2

SDCCH

0SDCCH

1SDCCH

2

SACCH

2

SACCH

0

SDCCH

3

SDCCH

3

R R

R R

SACCH

3

SACCH

1

RR

RR

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R R

R

R

RR

RR

RR

RR

RR

RR

DL: BCCH + CCCH + 4 SDCCH (SDCCH/4) + 4 SACCH (SACCH/4)

UL: CCCH + SDCCH/4 + SACCH/4

ISDCCH

0SDCCH

1SACCH

4SACCH

5

ISDCCH

0SDCCH

1SACCH

0SACCH

1

SDCCH/8 + SACCH/8

UL

Combination VIIDL

SDCCH

2SDCCH

3

SDCCH

2SDCCH

3

SDCCH

4SDCCH

5

SDCCH

4SDCCH

5

SDCCH

6SDCCH

7

SDCCH

6SDCCH

7

SACCH

6SACCH

7

SACCH

2SACCH

3

I

I

I

I

SACCH

5SACCH

6

SACCH

0SACCH

1

SACCH

7

SACCH

2

I

I

I

I

I

I

SDCCH

0SDCCH

1

SDCCH

0SDCCH

1

SDCCH

2SDCCH

3

SDCCH

2SDCCH

3

SDCCH

4SDCCH

5

SDCCH

4SDCCH

5

SDCCH

6SDCCH

7

SDCCH

6SDCCH

7

SACCH

4

SACCH

0

Fig. 15

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L1 Summary: Physical Channels & GSM Data Rates

GSM uses combined TDMA and FDMA for multiple access.

GMSK has been chosen as modulation principle. The GSM channel bandwidth is 200kHz, the modulation rate 270.833 kbit/s (derived from the GSM frequency normal 13MHz: 13 MHz/48).

According to the GSM TDMA principle chosen with 8 physical channels on onecarrier the total gross data rate for 1 physical channel is 270,833 / 8 = 33,85 kbit/s.

1 physical channel consists of 1 TS (UL/DL) on 1 carrier. 1 TS consists of 156.25 bit.

In the Normal Burst, used for TCH transmission, only 114 bit of these 156.25 bit areinformation bits (user data & user related signaling). Therefore, only 24.7 kbit/s of the33.85 kbit/s are information.

In a TCH Multiframe, only 24 of the 26 frames are filled with TCH, i.e. user data. Theother frames are filled with SACCH (frame 12) or Idle (frame 25). Therefore, the realgross rate of user data in GSM is 22.8 kbit/s.

The net rate in GSM is 13 kbit/s for FR speech, 12.2 kbit/s for EFR, 9.6 kbit/s for datatransmission (+ different other rates for HSCSD and GPRS). The difference betweenthe GSM net rate of user data and the gross rate of 22.8 kbit/s is used for dataredundancy to enable a reliable transmission.

The GSM modulation rate is 270,833 kbit/s. I.e. one single bit has a duration of3692.3 ns.

156.25 bit form one Time Slot TS, i.e. the duration of one TS is 0.5769 ms.

8 TS form one TDMA frame, i.e. the duration of one TDMA frame is 4.615 ms; itcontains 1250 bit.

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L1 Summary: Physical Channel / GSM Data Rates

TB3

Information57

S1

Training Seq.26

S1

TB3

GP8.25

Information57

0 1 2 3 4 5 6 7

RFC1

RFC2

RFC3

RFCi

RFC123

RFC124

••• •••

UL: 890 MHz 915 MHz

FDMA

GMSK

Modulation200 kHz

270.833kbit/s

TDMA

1 TDMA Frame: 4.615 ms / 1250 bit

1 TS: 33.85 kbit/s

1 Normal Burst: 576.9 s / 156.25 bit

1 Bit = 3.6923 s

24.7 kbit/s = 22.8 kbit/s TCH data (incl. redundancy)

+ 0.95 kbit/s SACCH + 0.95 kbit/s “Idle”

Fig. 16

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3 MOC / MTC

MOC / MTC


RACH: Channel Request

AGCH: Immediate Assign

SDCCH: CM Service Request

SDCCH: Authentication Request

SDCCH: Authentication Response

SDCCH: Cipher Mode Command

SDCCH: Cipher Mode Complete

SDCCH: Setup

SDCCH: Call Proceeding

SDCCH: Assign Command

•••

Fig. 17

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The MOC is defined as an MS initiated call setup. Several procedures are necessarybetween the MS and the BSS respectively the CN to set up a call. In the following theL1 messages on Um necessary for a “normal” MOC (without Off Air Call SetupOACSU; no emergency call) are shown:

Channel Request (RACH):MS requests the assignment of a dedicated signalingchannel

Immediate Assignment (AGCH): the network assigns a dedicated signalingchannel (SDCCH & SACCH). Additionally, a first TA information and PowerControl PC is included.

CM Service Request (SDCCH): the MS provides information on the requestedservice (Basic Call, Emergency Call, SMS,...) and transmits the subscribersidentity (TMSI / IMSI).

Authentication Request (SDCCH): the networks checks the real identity (Ki) of theSIM transmitting RAND.

Authentication Response (SDCCH): the MS answers with the SRES on theAuthentication Request

Cipher Mode Command (SDCCH): the network commands the MS to startciphering

Cipher Mode Complete (SDCCH): the MS acknowledges the cipher start (firstciphered message)

Setup (SDCCH): the MS transmits the Setup information including the desired TS /BS and number of the B-subscriber.

Call Proceeding (SDCCH): the network acknowledges the authorization for therequested service and confirms the call proceeding.

Assign Command (SDCCH): a TCH is allocated to the MS

Assign Complete (FACCH): the MS confirms the TCH allocation (using TCHresources)

Alerting (FACCH): the network informs the MS on successful call setup (i.e. thephone of the B subscriber rings). This starts generation of the ringing signals in theMS, too.

Connect (FACCH): the MS is informed, that the B subscriber accepted the call

Connect Acknowledge (FACCH): the MS confirms the Connect message

TCH: now network switch over to data transfer; the communication is able to start

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MOCMobile

Originating

Call








SDCCH: Setup

SDCCH: Call Proceeding


FACCH: Assign Complete

FACCH: Alerting

FACCH: Connect

FACCH: Connection Ackn.

TCH

MS requests for signaling channel

Signalling channel allocation [SDCCH x, TA]

Request MOC (SMS, Emergency Call,..)[TMSI/IMSI]

Request Authentication [RAND]

Authentication Response [SRES]

Start Ciphering [A5-X]

Acknowledgement; 1st ciphered message

Setup Message [Called No.]

Requested Service possible(after subscriber profile check in VLR)

TCH-Allocation [frequency, TS]

Acknowledgement on TCH resource

“Ringing at B-Subscriber”, start ringing signal in MS

“B-Subscriber accept call”

AcknowledgementStart of user data transmission & charging

Fig. 18

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The basic MOC includes at least 14 messages. As a rule, this signaling requires lessthan 2 s.

Optional further messages are:

IMEI Request, IMEI Response to check the equipment identity

TMSI Reallocation: to allocate a new TMSI to the MS

IMEI check and TMSI reallocation are proceeded after start of ciphering

OACSU:

In case of (TCH) overload on Um OACSU can be used. In this case, the AssignCommand / Assign Complete messages are sent after the Alert message, wasting noTCH resources during this time (only SDCCH resources).

Emergency Call

In case of an Emergency Call, Authentication and Cipher are skipped. Call setup isfaster and allows usage of every Mobile Equipment (even without valid SIM card;IMEI on black list).

MOC Part I & Part II

The two slides MOC Part I & Part II are optional for the TM2100 “GSM Introduction”course. They show the full message flow for a Basic MOC between MS and BSS /NSS, including IMEI check and TMSI reallocation as well as the Call Release.

The SS7 message flow using L4 protocols MAP & BSSAP and L3 Radio Interfacemessages of RR, MM and CM can be used for self-study.

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MOCPart I

Channel Request CHAN_REQ

MS BSS MSC VLR

ISDN

Immediate Assign IMM_ASS_CMD)

CM Service Request CM_SERV_REQCM_SERV_REQ

Authentication Request AUTH_REQAUTH_REQ

Authentication Response AUTH_RSPAUTH_RSP

Cipher Mode Command CIPH_CMDCIPH_CMD

Cipher Mode Complete CIPH_MOD_COMCIPH_MOD_COM

Check IMEI

TMSI Re-allocation TMSI_REAL_COMTMSI-REAL-CMD

TMSI_REAL_COMTMSI_REAL_COM

SETUP

SETUP

Process Access Request

PROC_ACCESS_REQ

AUTH_RSP

Set Cipher Mode

SET_CIPH_MODE

Forward New TMSI

FORW_NEW_TMSI

TSMI Acknowledged

TMSI_ACK

SEND INFO

EIR

Fig. 19

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MS BSS MSC VLR

ISDN

Call Proceeding CALL_PROCCALL_PROC

ALERTALERT

MOCPart II

Assign Command ASS_CMDAssign Request ASS_REQ

ASS_COMAssign Complete ASS_COM

Connect CONCON

CON_ACK

Connect Acknowledged CON_ACK

Initial Address Message IAM

Address Complete Message ACM

Answer Message ANM

User data

Release RELREL

DISCDisconnect DISC

Clear Command CLR_CMDRelease phys. Channel CHAN_REL

REL_COMRelease Command REL_COM

Clear Complete CLR_CMPDisconnect DISC

Release REL

Release Complete RLC

Complete Call CALL_CMP

Fig. 20

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MS requests for signaling channel

Signalling channel allocation [SDCCH x, TA]

Request MOC (SMS, Emergency Call,..)[TMSI/IMSI]

Request Authentication [RAND]

Authentication Response [SRES]

Start Ciphering

Acknowledgement; 1st ciphered message

Setup Message[Bearer Service, Calling No.]

Requested Service possible in MS

TCH-Allocation [frequency, TS]

Acknowledgement (on TCH resource)

“Ringing signal started in MS”

“Mobile subscriber accept call”

AcknowledgementStart of user data transmission & charging

MTCMobile

Terminating

Call








SDCCH: Setup

SDCCH: Call Confirmed


FACCH: Assign Complete

FACCH: Alerting

FACCH: Connect

FACCH: Connection Ackn.

TCH

PCH: Paging Request Searching MS in Location Area

Fig. 21


The MTC is initiated by the network if there is a call for the subscriber. The MTCmessage flow is very similar to the MOC message flow. The most importantdifference on Um is the start. The MS has to paged in all cells of a Location Area LA,using the Paging message.

Paging (PCH): The MS is paged in all LA cells using the TMSI / IMSI.

Setup (SDCCH): Another difference between MTC and MOC is the Setup message.In an MTC it is transmitted from the network to the MS, giving information on therequested service (TS, BS) and the ISDN / MSISDN number of the calling party.

Call Confirmed (SDCCH): After checking its capabilities to support the requestedservice, the MS acknowledges the Setup message with Call Confirmed.

Alerting (FACCH): Different to the MOC, in the MTC the Alerting message istransmitted from the MS to the network, to indicate the start of ringing in the MS.

Connect (FACCH) & Connection Acknowledge: Different to the MOC, in the MTCboth messages have opposite direction, too.

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Appendix

Appendix

Contents

2References1 3 Abbreviations2

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Appendix Siemens

1 References

M. Mouly, M.B. Pautet, "The GSM System for Mobile Communications", Cell & Sys(1992), ISBN 2-9507190-0-7

S. Redl, M. Weber, K. Oliphant, "An introduction to GSM", Artech House Inc.(1995), ISBN 0-89006-785-6

A. Mehrotra, "GSM System Engineering", Artech House Inc. (1997), ISBN 0-89006-860-7

G. Heine, "GSM-Signalisierung", Funkschau: Funktechnik, Franzis-Verlag GmbH(1998), ISBN 3-528-15302

G. Heine, "GSM Networks: Protocols, Terminology and Implementation", ArtechHouse Inc. (1999), ISBN 0-89006-471-7

G. Heine, "GPRS from A – Z", Artech House Inc. (2000), ISBN 1-58053-181-4

G. Heine, "GPRS, EDGE, HSCSD and the Path to 3G", Artech House Inc. (2001),CD-ROM, ISBN 1-58053-275-6

"System Description D900/D1800 - GSM PLMN" A50016-D1109-V11-2-7618

"Technical Description D900/D1800 - Switching Subsystem (SSS)" A50016-D1109-V2-1-7618

"Base Station System (TED-BSS)" A30808-X3247-H10-1-7618

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Appendix Siemens

2 Abbreviations

AB access burst

AC authentication center

ACCH associated control channel

ACE antenna coupling equipment

ACE-Rx ACE receive side

ACE-Tx ACE transmit side

ACG auxiliary clock generator

ACM address complete message

ACU antenna combining unit

ADC analog to digital converter

AEF additional elementary function

AF audio frequency

AFC automatic frequency control

AGC automatic gain control

AGCH access grant channel

AMA automatic message accounting

AMPC ATM bridge Processor C

ANT-COMB antenna combiner

AoC advice of charge

AP application part

APS application program system

ARFCN absolute radio frequency number

ARQ automatic repeat request

ASN ATM Switching Network

ATB all trunks busy

ATE automatic test equipment

AUC authentication center

AUT(H) authentication

BA BCCH allocation

BAIC barring of all incoming calls

BAOC barring of all outgoing calls

BAP base processor (CP113)

BCC base transceiver station color code

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Appendix Siemens

BCCH broadcast control channel

BCH broadcast channel

BER bit error rate

BHCA busy hour call attempts

BIC- Roam barring of incoming calls when roaming outside the HPLMN country

BNHO barring all outgoing calls except those to HPLMN

BOIC barring of outgoing international calls

BOIC-exHC barring of outgoing international calls except those directed to theHPLMN

BS base station

BSC base station controller

BSCU base station control unit

BSIC base transceiver station identity code

BSS base station system

BSSAP base station system application part

BSSMAP base station system management application part

BSSOMAP base station system operation and maintenance application part

BSU base station system switch unit

BTS base transceiver station

CA cell allocation

CAS channel associated signaling

CAP call processor (CP113)

CBCH cell broadcast channel

CBS cell broadcast service

CC call control

CC channel coding

CC country code

CCBS completion of calls to busy subscribers

CCC common channel control

CCG central clock generator

CCH control channel

CCITT Comité Consulatif International Téléphonique et Télégraphique

CCNC common channel signaling network control

CCNP common channel signaling network processor

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Siemens Appendix

CCS7 common channel signaling system No. 7

CCS common channel signaling

CCU channel coding unit

CdPA called party address

CF call forwarding

CFB call forwarding on mobile subscriber busy

CFNRc call forwarding on mobile subscriber not reachable

CFNRy call forwarding on no reply

CFU call forwarding unconditional

CGI cell global identity

CgPA calling party address

CHA component handling

CI cell identity

CIC circuit Identification code

CKSN cipher key sequence number)

CLIP calling line identification

CLIR calling line identification restriction

CMD command

CMY common memory

CNI comfort noise insertion

COLI calling line identification

CoLP connected line identification presentation

CoLR connected line identification restriction

CP call processing

CP coordination processor

CPU central processing unit

CR code receiver

CRC cyclic redundancy check

CT call transfer

CT Craft Terminal

CTC continuity check

CUG closed user group

CW call waiting

DAS digital announcement system

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DB dummy burts

DBMS data base management system

DCCH dedicated control channel

DCN data communication network

DCP data communication processor

DCS1800 digital communication system

DE digital exchange

DEC digital echo compensator

DEMUX demultiplexer

DHA dialogue handling

DIU digital interface unit

Dm control/data channel

DL down link

DPC destination point code)

DPPC data post processing computer

DPPS data post processing system

DRX discontinuous reception

DSMX digital signal multiplexer

DTAP direct transfer application part

DTMF dual tone multi frequency

DTX discontinuous transmission

EIR equipment identification register

EMML extended man machine language

ERP effective radiated power

EWSD Digitales Elektronisches Wählsystem

FAC final assembly code

FACCH fast associated control channel

FACCH/F full rate FACCH

FACCH/H half rate FACCH

FB frequency correction burst

FC filter coupler

FCCH frequency correction channel

FDMA frequency division multiple access

FEC forward error correction)

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Siemens Appendix

FHE frequency hopping equipment

FN frame number

FPLMTS future public land mobile telecommunication system (CCITT)

GCR Group Call Register

GMSC gateway MSC

GMSK gaussian minimum shift keying

GOS grade of service

GP guard period

GP group processor

GSM Global System for Mobile communications

GSM PLMN GSM public land mobile network

HANDO handover

HC hard copy

HF history file

HLR home location register

HLR-ID home location register identity

HMSC home MSC

HO HANDO

HOLD call hold

HPA high power amplifier

HPLMN home PLMN

HSN hopping sequence number

IAM initial address message

ICB incoming calls barred

ID identification

ID identity

IMEI international mobile equipment identity

IMN installation manual

IMSI international mobile subscriber identity

IMT-2000 International Mobile Telecommunications

IN intelligent network

IOC input/output controller

IOP input/output processor

IOP: AUC input/output processor for the authentication center

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ISC international switching center

ISDN integrated services digital network

ISUP OSDN user part

IWE interworking equipment

IWF interworking function

IWUP interworking user part

Kc cipher key (ciphering key)

Ki individual subscriber authentication key

LA location area

LAC Location area code

LAI location area identity

LAN local area network

LAPDm link access protocol on the Dm channel

LE local exchange

LIC Line Interface Circuit

Lm TCH with capacity lower than Bm

LMSI local mobile station identity

LMT local maintenance terminal

LR location register

LTG line/trunk group

MA mobile allocation

MAP mobile application part

MAH mobile access hunting

MB message buffer

MBG message buffer group

MBU message buffer unit

MCC mobile country code)

MCI malicious call identification

ME mobile equipment

MFC multifrequency code

MGT mobile global title

MIB management information base

MM mobility management

MMI man machine interface

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MMI man machine interpreter

MMN maintenance manual

MML man machine language

MNAP management network access point

MNC mobile network code

MOC mobile originating call

MP Main Processor

MPTY multi party service

MPU Main Processor Unit

MS mobile station

MS mobile subscriber

MSC mobile services switching center

MSIN mobile subscriber identification number

MSISDN mobile station international ISDN number

MSRN mobile station roaming number

MT mobile termination

MTC mobile termination call

MTE mobile termination equipment

MTP message transfer part

NB normal burst

NCC network color code (PLMN color code)

NDC national destination code

NE network entity, network element

NEF network element function

NF network function

NI national Indicator)

NM network management

NMC network management center

NMSI national mobile station identification

O&M operation and maintenance

OACSU off air call set up

OCB outgoing calls barred

ODAGEN office date area generator

OMAP operation & maintenance application part

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OMC operation & maintenance center

OMC- B operation & maintenance center for BSS

OMC- S operation & maintenance center for SSS

OMP operation & maintenance processor

OMP- B operation & maintenance processor for BSS

OMP- S operation & maintenance processor for SSS

OMS operation & maintenance subsystem

OMT operation & maintenance terminal

OMT- B operation & maintenance terminal for BSS

OMT- S operation & maintenance terminal for SSS

OPC originating point code

PA power Amplifier

PCH paging channel

PCM pulse code modulation

PCM- INT PCM interface

PCS personalization center for SIM

PDN public data network

PIN personal identification number

PLMN public land mobile network

PM performance management

PSPDN packet switched public data network

PSTN public switched telephone network

PSU power supply unit

QA Q (interface adapter)

QOS quality of service

RA rate adaptation

RAB random access burst

RACH random access channel

RAE recorded announcement equipment

RAND random number

REC recommendation

REQ request

RES response

RF radio frequency

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Siemens Appendix

RFC radio frequency channel

RFCH radio frequency channel

RFCN radio frequency channel number

RFM radio frequency management

RFN reduced TDMA frame number

RLP radio link protocol

RMA regional maintenance area

RMC regional maintenance center

ROI remote operation interface

ROSE remote operation service element

RPE- LTP regular pulse excited long term prediction

RR radio resource management

RSE radio system entity

RSS radio subsystem

RT radio terminal

RX or Rx receiver

RXLEV received signal level

RxMC receiver multicoupler

RXQUAL received signal quality

SACCH Slow Associated Control Channel

SACCH/T slow, TCH- associated control channel

SACCH/TF slow, TCH/FS- associated control channel

SACCH/TH slow, TCH/HS associated control channel

SAP service access point

SAPI service access point indicator

SB synchronization burst

SC Switch Commander

SCCP signaling connection control part

SCF Signaling Control Function

SCP Signaling Control Point

SCH synchronization channel

SCN sub- channel number

SCP service control point (IN)

SDCCH standalone dedicated control channel

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SFH slow frequency hopping

SG safeguarding

SGC switch group control

SGL service guidelines

SI service indicator

SIM subscriber identity module

SM security management

SMC submultiplex channel

SMG Special Mobile Group

SMS service management system

SN subscriber number

SN switching network

SNR serial number

SP signaling point

SPC signaling point code

SPC stored program control

SRES signed response

SSF Signaling Switching Function

SSG space stage group

SSM space stage module

SSNC Signaling System Network Control

SSP Service Switching Point

SSS switching subsystem

STP signaling transfer point

SW software

SYP system panel

SYPC system panel control

SYPD system panel display

TA Terminal Adaptation

TAC Type Approval Code

TAC technical assistance center

TB tail bit

TC transaction capability

TCAP transaction capability Part

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Siemens Appendix

TCB transcoder board

TCG transcoder group

TCGQ transcoder group quartet

TCH traffic channel

TCH/F full rate traffic channel

TCH/FS TCH full rate speech

TCH/H half rate traffic channel

TCH/HS TCH half rate speech

TDMA time division multiple access

TE terminal equipment

TETRA Terrestrial Trunked Radio Access

THA transaction handling

TMN telecommunication management network

TMRP tower mounted receiver preamplifier

TMS telecommunication management system

TMS test mobile station

TMSI temporary mobile subscriber identity

TN telecommunication network

TN timeslot number

TRAU transcoding and rate adaptation unit

TRX transceiver

TS tele service

TS timeslot

TSM time stage module

TSG time stage group

TUP telephony user part

TX or Tx transmitter

UL uplink

UMTS universal mobile telecommunication system

UP user part

UUS user to user signaling

VAD Voice Activity Detection

VBR Variable Bit Rate

VE exchange equipment

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VBS Voice Broadcast Service

VGCS Voice Group Call Services

VHE Virtual Home Environment

VLR Visitor Location Register

VMSC Visited MSC

VoIP Voice over IP

VPLMN Visited PLMN

WAN Wide Area Network

WAP Wireless Application Protocol

WARC World Administrative Radio Conference

WLL Wireless Local Loop

WS Work Station

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Part 6

CDMA Overview

1 The way to CDMA Technology Pages (1-39)

2 Basic Concept of Spread Spectrum Technology

Pages (1-16)

3 CDMA codes and its usage Pages (1-20)

4 CDMA Air Interface Overview Pages (1-18)

5 CDMA System Aspects Pages (1-15)


7 Reference Pages (1-3)

8 Glossary Pages (1-6)

Sub ‐ Sections CDMA Overview


Chapter 1 The way to CDMA Technology

Aim of study This chapter introduces introduction to cellular technology, cellular system

architecture and components.

Contents Pages

1 Introduction to Cellular Technology 2

2 Advantages of Digital Communications 8

3 Cellular System Architecture 11

4 Cellular System Components 15

5 Wireless Digital Transmission Problems 17

6 Solutions against Air transmission Problems 21

7 Transmission Principles 24

8 Data Transmission 38

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1.1 Progress in Radio Communications

The quest to know the unknown and see the unseen is inherent in human nature.

It is this restlessness that has propelled mankind to ever-higher pinnacles and ever-deeper depths. This insatiable desire led to the discovery of light as being electromagnetic, paving the way to discovery of the radio.

The origin of radio can be traced back to the year 1680 to Newton theory of composition of white light of various colors. This theory brought the importance as light as an area of study to the attention of many scientists, especially those in Europe, who began to pursue experiments with light which lead to importantdiscoveries connected to the eventual development of the radio.

These discoveries are the foundation of today’s wireless cimmunicaton systems. Experiments with light are still being carried out today in many universities, and industries. One of the outcomes of light experiments in the 1970s is the optical fiber, which is currently being used for long – haul voice and data transmission. It is believed that the use of optical fiber technology will increase dramatically the introduction of wideband networks for voice, data, and video transmission, which is based on the Asynchronous Transfer Mode (ATM) switch.

Radio connections were first used for Wireless Communications in the late 19th century; information was sent via "ether" as follows:

1 Introduction to Cellular Technology

The Way To CDMA Technology

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Progress in Radio communications

1873 Electromagnetic wave theory by J.C. Maxwell

1887 Experimental proof of the existence of electromagnetic waves by H. Hertz

1895 First receiver with antenna for weather reports by A. Popow

1895 First wireless transmission using spark inductor generated by G. M. Marconi

1897 Marconi Wireless Telegraphy Company founded

1901 First transatlantic transmission by Marconi

1909 First radio broadcast at New York, Caruso

1917 First mobile transmission, BS - train

1952 Usage of Analogue Mobile Systems in USA and Europe

1978 CEPT reserved 2x25MHz for GSM

1992 Commercial use of GSM

Fig.1


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1.2 The Growth in Cellular Market & its demands

The cellular telephone industry has enjoyed phenomenal growth since its inception in 1983. In just one more example of the impossibility of projecting the adoption of new technologies, a widely accepted 1985 prediction held that the total number of cellular subscribers might reach as many as 900,000 by the year 2000. In fact, by the end of 1994 there were well over 20 million subscribers in the United States alone, and approximately 50 million worldwide. Recent annual subscriber growth rates have been as high as 40%, and it is believed that this growth rate could continue through the rest of the 1990s.

In order to meet this increasing demand for service, new digital cellular telephone systems have been introduced during the first half of the 1990s. As today's cellular operators move to adopt these new technologies in their systems, they demand:

l Increased capacity within their existing spectrum allocation and easy

deployment of any technology it takes to get them that capacity increase. l Higher capacities and lower system design costs (plus lower infrastructure

costs) which will lead to a lower cost per subscriber. l A lower cost per subscriber combined with new subscriber features, which

will help the operators to increase their market penetration. l An increased market penetration, which will lead to an increase in number of

subscribers and a system, which offers support for that, increased capacity. l High quality calls must be maintained during the change to or migration to any

new digital technology.


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Avdantages of cellular communications

Fig.2

• lower cost per subscriber • Increased market penetration • Higher capacities • lower system design costs


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1.3 Why is it called cellular? Everyone is familiar with the usage of the term “cellular” in describing mobile radio

systems. You probably know that it is called cellular because the network is composed of a number of cells. Mobile radio systems work on the basis of cells for two reasons.

The first reason is that radio signals at the frequencies used for cellular travel only a few kilometers (kms) from the point at which they are transmitted.

They travel more or less equal distances in all directions; hence, if one transmitter is viewed in isolation, the area around it where a radio signal can be received is typically approximately circular. If the network designer wants to cover a large area, then he must have a number of transmitters positioned so that when one gets to the edge of the first cell there is a second cell overlapping slightly, providing radio signal. Hence the construction of the network is a series of approximately circular cells.

The second reason has to do with the availability of something called radio spectrum. Simply, radio spectrum is what radio signals use to travel through space.

Using a mobile radio system, it consumes a certain amount of radio spectrum for the duration of the call. An analogy here is car parks. When you park your car in a car park it takes up a parking space. When you leave the car park, the space becomes free for someone else to use. The number of spaces in the car park is strictly limited and when there are as many cars as there are spaces nobody else can use the car park until someone leaves.

Radio spectrum in any particular cell is rather like this. However, there is an important difference. Once you move far enough away from the first cell, the radio signal will have become much weaker and so the same bit of radio spectrum can be reused in another cell without the two interfering with each other. By this means, the same bit of radio spectrum can be reused several times around the country. So splitting the network into a number of small cells increases the number of users who can make telephone calls around the country.

So, in summary, cellular radio systems are often called “cellular” because the network is composed of a number of cells, each with radius of a few kilometers, spread across the country. This is necessary because the radio signal does not travel long distances from the transmitter, but it is also desirable because it allows the radio frequency to be reused, thus increasing the capacity of the network.


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Fig. 3


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2.1 Digital Communication

First of all we can say that a digital communication system is one where the voice signal has been digitized prior to wireless transmission.

Digitizing is aprocess where the voice signal is sampled and discrete, numiric representation of the signal are transmitted ,rather than the original signal itself.

This is much different from analog systems where the original,continuous voice signal is transmitted using a standard form of FM modulation.

As the term „Digital“ implies, the voice signal is digitized for transmission within the cellular networks.Once digitized, Advanced coding , transmission,and error correction techniques are employed. These additional techniques make it possible to detect and correct transmission errors at the receiving end.

Another advantage of digital wireless communications is that digital provides more traffic capacity per given RF spectrum. This is made possible by using the channel bandwidth more efficiently .

In digital systems, multible users occupy the same frequency, and they are separated by time or codes. This is more efficient than assigning each user to a separate frequency , which is efficient than assigning each user to a separate frequency, which is common in analog systems.

Digital systems also use techniques to reduce, or compress the amount of information to be transmitted over the air from each user.

These compression techniques can take advantage of the probability that not every user needs maximum bandwidth at exactly the same moment.

Another advantage of digital communication system is that they have ah inherent level of security . Unothorized listeners must have complex receivers, they must decode the digital information, and then they must convert the digital signal into analog signal.

Digital has better built-in support for non-voice services and user data traffic.

By bypassing the voice signal compression process, user data can be processed directly in their digital formats.

With digital systems, there is no need to convert the signal. The data is simply passed through as digital information. This digital information can usually be processed through the system at higher speeds.

Lastly , Analog sytems, on the other hand, use much simpler transmission techniques, which require a receiver no more complex than an inexpensive FM radio.

2 Advantages of Digital Communications


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Avdantages of digital communications

Fig. 4

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• Security • Higher capacities • Easily Maintainance • Minaturization an friendleness • High Quality with low cost • Worlwide Availability • New Service Implementation • High Fidility

Distance to BS

Signal Quality Digital Signal

AnalogueSignal

Transmission Quality:

“Easy to regenerate”


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2.2 Digital Mobile System

As demand for mobile telephone service has increased, service providers found that basic engineering assumptions borrowed from wireline (landline) networks did not hold true in mobile systems and the early analogue systems quickly became saturated, and the quality of service decreased rapidly.

The components of a typical digital cellular system is shown in fig.. The advantages of digital cellular technologies over analog cellular networks

include increased capacity and security. Technology options such as TDMA and CDMA offer more channels in the same analog cellular bandwidth and encrypted voice and data.

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3.1 System Architecture Increases in demand and the poor quality of old service led mobile service

providers to research ways to improve the quality of service and to support more users in their systems. Because the amount of frequency spectrum available for mobile cellular use was limited, efficient use of the required frequencies was needed for mobile cellular coverage. In modern cellular telephony, rural and urban regions are divided into areas according to specific provisioning guidelines.

Deployment parameters, such as amount of cell-splitting and cell sizes, are determined by engineers experienced in cellular system architecture.

Provisioning for each region is planned according to an engineering plan that includes cells, clusters, frequency reuse, and handovers.

• Cells and Cell Splitting

A cell is the basic geographic unit of a cellular system.The term cellular comes from the honeycomb shape of the areas into which a coverage region is divided. Cells are base stations transmitting over small geographic areas that are represented as hexagons. Each cell size varies depending on the landscape. Because of constraints imposed by natural terrain and man-made structures, the true shape of cells is not a perfect hexagon.

Unfortunately, economic considerations made the concept of creating full systems with many small areas impractical. To overcome this difficulty, system operators developed the idea of splitting cells into sectors to form sector cells.

• Clusters

A cluster is a group of cells in which all available frequencies have been used once. No channels are reused within a cluster.

• Frequency Reuse

The concept of frequency reuse is based on assigning to each cell a group of radio channels used within a small geographic area. Cells are assigned a group of channels that is completely different from neighboring cells. The coverage area of cells are called the footprint. This footprint is limited by a boundary so that the same group of channels can be used in different cells that are far enough away from each other so that their frequencies do not interfere.Cells with the same number have the same set of frequencies.

3 Cellular System Architecture


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Cluster

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3.2 Types of cells

Different types of cells are used due to the density variation of population.

• Macrocells The macrocells are large cells for remote and sparsely populated areas.

• Microcells These cells are used for densely populated areas. By splitting the existing areas

into smaller cells, the number of channels available is increased as well as the capacity of the cells. The power level of the transmitters used in these cells is then decreased, reducing the possibility of interference between neighboring cells.

• Selective cells It is not always useful to define a cell with a full coverage of 360 degrees. In some

cases, cells with a particular shape and coverage are needed. These cells are called selective cells.

A typical example of selective cells is the cells that may be located at the entrances of tunnels where coverage of 360 degrees is not needed. In this case, a selective cell with coverage of 120 degrees is used.

• Umbrella cells A freeway crossing very small cells produces an important number of handovers

among the different small neighboring cells. In order to solve this problem, the concept of umbrella cells is introduced. An umbrella cell covers several microcells. The power level inside an umbrella cell is increased comparing to the power levels used in the microcells that form the umbrella cell. When the speed of the mobile is too high, the mobile is handed off to the umbrella cell. The mobile will then stay longer in the same cell (in this case the umbrella cell). This will reduce the number of handovers and the work of the network .A too important number of handover demands and the propagation characteristics of a mobile can help to detect its high speed.

• Handoff The final obstacle in the development of the cellular network involved the problem

created when a mobile subscriber traveled from one cell to another during a call. As adjacent areas do not use the same radio channels, a call must either be dropped or transferred from one radio channel to another when a user crosses the line between adjacent cells. Because dropping the call is unacceptable, the process of handoff was created. Handoff occurs when the mobile telephone network automatically transfers a call from radio channel to radio channel as a mobile crosses adjacent cells.

During a call, When the mobile unit moves out of the coverage area of a given cell site, the reception becomes weak. At this point, the cell site in use requests a handoff. The system switches the call to a stronger frequency channel in a new site and the call continues as long as the user is talking, and the user does not notice the handoff at all.


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4.1 Cellular System Components

The cellular system offers mobile and portable telephone stations the same service provided fixed stations over conventional wired loops. It has the capacity to serve tens of thousands of subscribers in a major metropolitan area. The cellular communications system consists of the following four major components that work together to provide mobile service to subscribers: 1. Mobile telephone switching office (MTSO)

2. Cell site with antenna system 3. Mobile Station (MS)

• Mobile Telephone Switching Office (MTSO) The MTSO is the central office for mobile switching. It houses the mobile switching

center (MSC), field monitoring and relay stations for switching calls from cell sites to wireline central offices (PSTN).

• The Cell Site

The term cell site is used to refer to the physical location of radio equipments that provide coverage within a cell. A list of hardware located at a cell site includes power sources, interface equipment, radio frequency transmitters and receivers, and antenna systems.

• Mobile Station (MS)

The mobile subscriber unit consists of a control unit and a transceiver that transmits and receives radio transmissions to and from a cell site. Three types of MSUs are available: 1. The mobile telephone (typical transmit power is 4.0 watts) 2. The portable (typical transmit power is 0.6 watts)

3. The transportable (typical transmit power is 1.6 watts) The mobile telephone is installed in the trunk of a car, and the handset is installed

in a convenient location to the driver. Portable and transportable telephones are hand held and can be used anywhere. The use of portable and transportable telephones is limited to the charge life of the internal battery.

4 Cellular System Components


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5.1 Reasons leading to Wireless Digital Transmission Problems Wireless communication channels suffer from severe attenuation and signal

fluctuations and this is mainly due to three important reasons which are:

1. Velocity of Mobile Station within the area of the Base Tranciever Station. 2. Distance between Mobile Station and the Base Tranciever Station. 3. Obstacles between the Mobile Station and the Base Tranciever Station. Large attenuation is due to the user’s mobility through the propagation

environment that causes almost no direct signal from the transmitter can reach the receiver. Even if so, the line-of-sight signal may be superimposed by its reflected or scattered duplicates that reach the receiver at different time instant causing signal fluctuations. When a mobile station moves from one location to another, all propagation scenario may change completely and the received signal changes accordingly. Three different models that are commonly used to characterise a wireless channel are:

• Propagation path loss (near-far attenuation) .

• Shadowing (variation on the average power) .

• Multipath fading (fast signal fluctuation).

• Propagation path loss It occurs when the received signal becomes weaker and weaker due to

increasing distance between MS and BTS . Path loss is proportional to the square of the distance and the square of the transmitted frequency .

• Shadowing

It is due to obstacles being between the MS and the BTS , like buildings, hills etc. When the MS moves around , the signal fluctuates normally around a mean value depending on the obstacles.

• Multipath fading It occures when there is more than one transmission path to the MS or BTS ,

and therefore more than one signal is arriving at the receiver .This may be due to buildings or mountains , either close to or far from the reciving device,Rayleigh fading and time dispersion are forms of multipath fading.

1. Rayleigh fading It occures when the signal takes more than one path between the MS and BTS. Rayleigh fading occurs when the obstacles are near to the receiving antenna

2. Time dispersion

It contrasts to Rayleigh fading , the reflected signal comes from an object far away from the receiving antenna .Since the bit rate on the air is 270 kbit/sec,one bit corresponds to 3.7 µ sec or 1.1 km . If an obstacle is further than 500 m away, then the reflected bit will interfere with the next transmitted bit (ISI).

5 Wireless Digital Transmission Problems


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5.2 Result of Wireless Digital Transmission Problems • Bit Error Rate

Sometimes, when you are using a mobile phone, you will notice that the speech quality “breaks up” or disappears completely for short periods of time. By moving toward a window you can sometimes improve the situation. This loss of speech quality is caused by errors. That means, the transmitter might send 1011, but because of propagation problems, such as fast fading, the receiver gets 1001.The third bit is said to be in error. This is a little like spelling something over the phone.You might say “S” but the person at the other end might respond “was that F?” An error was made because the line was not of sufficient quality. Mobile phones contain advanced systems for correcting errors but However, these systems are not always able to remove all the errors. Without error correction, the speech quality would always be so terrible that you would never be able to understand the other person.

Interference, fading, and random noise cause errors to be received, the level of which will depend on the severity of the interference. The presence of errors can cause problems. For speech coders such as ADPCM (Adaptive Defrential PCM), if the bit error rate (BER) rises above 10-3 (that is, 1 bit in every 1000 is in error, or the error rate is 0.1%) then the speech quality becomes unacceptable.

For near-perfect voice quality, error rates of the order of 10-6 are required. For data transfers, users expect much better error rates, for example on computer files, error rates higher than 10-9 are normally unacceptable.

If the only source of error on the channel was random noise, then it would be possible, and generally efficient, to simply ensure that the received signal power was sufficient to achieve the required error performance without any need for error correction. However, where fast fading is present, fades can be momentarily as deep as 40 dB. To increase the received power by 40 dB to overcome such fades would be highly inefficient, resulting in a significantly reduced range and increased interference to other cells. Instead, error correction coding accepts that bits will be received in error during fades but attempts to correct these using extra bits (“redundant” bits) added to the signal.


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6.1 Solutions for Wireless Digital Transmission Problems

• Antenna Diversity

It increases the received signal strength by taking advantage of the nature properties of radio waves , there are two diversity methods, they are :-

1. Space diversity .

2. Polarization diversity .

♦ Space diversity can be achieved by mounting two receivers instead of one . If the two receivers

are physically separated , the probability that both of them are affected by a deep fading dip at the same time is low .

♦ Polarization diversity With this technique the two space diversity receivers are replased by one dual

polarized antenna , the antenna contains two differently polarized antenna arrays.

• Time Advance

Time Advance is introduced to overcome the effect of time alignment. When the MS is moving far away from the BTS , this BTS tells the MS how much time ahead of the synchronization time it must transmit the burst .

6 Solutions of Air transmission problems


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6.2 Solutions for Bit Error Rate • Channel Coding

Error correction is widely deployed in mobile radio, where fast fading is almost universally present. Error correction systems all work by adding redundancy to the transmitted signal. The receiver checks that the redundant information is as it would have expected and, if not, can make error correction decisions. In an error detection scheme, the receiver requests that the block that was detected to be in error is retransmitted. Such schemes are called automatic request repeat (ARQ).Some of the more advanced coding systems can perform error correction and also detect if there were too many errors for it to be possible to correct them all and hence request retransmission in this case.

• Interleaving

Signals traveling through a mobile communication channel are susceptible to fading. The error-correcting codes are designed to combat errors resulting from fades and, at the same time, keep the signal power at a reasonable level. Most error-correcting codes perform well in correcting random errors. However, during periods of deep fades, long streams of successive or burst errors may render the error-correcting function useless. Interleaving is a technique for randomizing the bits in a message stream so that burst errors introduced by the channel can be converted to random errors.

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7.1 Duplex Transmission

• FDD and TDD

Two duplex methods are used for coordinating the uplink (UL) and downlink (DL) components of a transmission between a base station and a mobile station, namely Frequency Division Duplex (FDD) and Time Division Duplex (TDD).

UL and DL are implemented for FDD in different frequency bands. The gap between the two frequency bands for UL and DL is known as the duplex distance. It is constant for all mobile stations in a standard. Generally the DL frequency band is positioned at the higher frequency than the UL band.

In the case of TDD, UL and DL are implemented in the same frequency band, Uplink (UL) and Downlink (DL) takes place at different times. There is fast switching between UL and DL transmission, so that the user has the impression of simultaneous transmission and reception.

As a result, only a fraction of the time needed for analog transmission is required for digital transmission of subscriber data.

7 Transmission Principles


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7.2 Multiple Access Techniques

Wireless telecommunications has drastic increase in popularity, resulting in the need for technologies that allow multiple users to share the same spectrum, called Multiple Access techniques.

FDMA, TDMA and CDMA are the three major technologies available, along with variations of each.

All three technologies have one goal in common that is the most important concept to any cellular telephone systems which is “Multiple Access”, meaning that multiple, simultaneous users can be supported. In other words, a large number of users share a common pool of radio channels. The technologies differ significantly in the manner by which they accomplish this sharing.

7.2.1 Frequency Division Multiple Access

FDMA is used for standard analog cellular. Each user is assigned a discrete band of the RF spectrum.The voice signal of each user is modulated on a separate channel frequency, which is assigned 100% of the time to that user.

For example:

AMPS systems use 30 kHz "slices" of spectrum for each channel. Narrowband AMPS (NAMPS) requires only 10 kHz per channel. TACS channels are 25 kHz wide. With FDMA, only one subscriber at a time is assigned to a channel. No other conversations can access this channel until the subscriber's call is finished, or until that original call is handed off to a different channel by the system. In order to overcome this inefficiency, digital access technologies were introduced.

FDMA requires NO system timing. FDMA requires NO timing accuracy.

FDMA –based Analog system generally considered as a low capacity system.


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7.2.1.1 The Advanced Mobile Phone Service (AMPS)

AMPS was released in 1983 using the 800-MHz to 900-MHz frequency band and the 30 kHz bandwidth for each channel as a fully automated mobile telephone service. It was the first standardized cellular service in the world and is currently the most widely used standard for cellular communications. Designed for use in cities, AMPS later expanded to rural areas. It maximized the cellular concept of frequency reuse by reducing radio power output. The AMPS telephones (or handsets) have the familiar telephone-style user interface and are compatible with any AMPS base station. This makes mobility between service providers (roaming) simpler for subscribers. Limitations associated with AMPS include:

1. Low calling capacity 2. Limited spectrum

3. No room for spectrum growth 4. Poor data communications 5. Minimal privacy

6. Inadequate fraud protection

AMPS is used throughout the world and is particularly popular in the United States, South America, China, and Australia. AMPS uses frequency modulation (FM) for radio transmission. In the United States, transmissions from mobile to cell site use separate frequencies from the base station to the mobile subscriber.


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7.2.2 Time Division Multiple Access

In TDMA users are still assigned a discrete slice of RF spectrum, but multiple users now share that RF channel on a time slot basis. Each of the users alternate their use of the RF channel . Frequency Division is still used, but these carriers are now further subdivided into some number of time slots ber carrier.

A user is assigned a particular time slot in a carrier and can only send or receive information at those times. This is true wether or not the other time slots are being used. Information flow is not continuous for any user, but rather is sent and received in „bursts“ . The bursets are re-assembled at the receiving end , and appear to provide continuous sound because the process is very fast.

TDMA digital standards include North American Digital Cellular (known by its standard number IS-54), Global System for Mobile Communications (GSM), and Personal Digital Cellular (PDC).

For example, IS-54 based TDMA system, a 30 kHz channel is divided into 6 time slots each with 30 kHz band modulated signal. Although there are 6 time slots, each user needs 2 time slots, so there are a total of 3 users per 30 kHz channel. This is three times more efficient than AMPS

PDC divides 25 kHz slices of spectrum into three channels. GSM system uses both FDMA and TDMA operates with a 200 Khz bandwidth,

divided into 8 timeslots, where each user is assigned a single timeslot, thus allowing 8 users per channel frequency.

TDMA requires timing synchronization TDMA requires millisecond accuracy. GSM and TDMA are about 3 times more spectral efficient than analog.


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7.2.2.1 The GSM network

The GSM technical specifications define the different entities that form the GSM network by defining their functions and interface requirements. The GSM network can be divided into four main parts: The Mobile Station (MS).

The Base Station Subsystem (BSS). The Network and Switching Subsystem (NSS).

The Operation and Support Subsystem (OSS).

• Mobile Station MS

A Mobile Station consists of two main elements: 1. The Mobile Equipment Terminal.

2. The Subscriber Identity Module (SIM). There are different types of terminals distinguished principally by their power and

application: The `fixed' terminals are the ones installed in cars. Their maximum allowed output power is 20 W.The GSM portable terminals can also be installed in vehicles. Their maximum allowed output power is 8W.

The handhels terminals have experienced the biggest success thanks to their weight and volume, which are continuously decreasing. These terminals can emit up to 2 W. The evolution of technologies allows decreasing the maximum allowed power to 0.8 W.

• The SIM (Subscriber Identity Module) The SIM is a smart card that identifies the terminal. By inserting the SIM card into

the terminal, the user can have access to all the subscribed services. Without the SIM card, the terminal is not operational. The SIM card is protected by a four-digit Personal Identification Number (PIN). In order to identify the subscriber to the system, the SIM card contains some parameters of the user such as its International Mobile Subscriber Identity (IMSI).

Another advantage of the SIM card is the mobility of the users. In fact, the only element that personalizes a terminal is the SIM card. Therefore, the user can have access to its subscribed services in any terminal using its SIM card.


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• The Base Station Subsystem

The BSS connects the Mobile Station and the NSS. It is in charge of the transmission and reception. The BSS can be divided into two parts:

• The Base Transceiver Station (BTS).

• The Base Station Controller (BSC).

1. The Base Transceiver Station: The BTS corresponds to the transceivers and antennas used in each cell of the

network. A BTS is usually placed in the center of a cell. Its transmitting power defines the size of a cell. Each BTS has between one and sixteen transceivers depending on the density of users in the cell.

2. The Base Station Controller:

The BSC controls a group of BTS and manages their radio ressources. A BSC is principally in charge of handovers, frequency hopping, exchange functions and control of the radio frequency power levels of the BTSs.

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• The Network and Switching Subsystem Its main role is to manage the communications between the mobile users and

other users, such as mobile users, ISDN users, fixed telephony users, etc. It also includes data bases needed in order to store information about the subscribers and to manage their mobility. The different components of the NSS are described below.

1. The Mobile services Switching Center (MSC) It is the central component of the NSS. The MSC performs the switching functions

of the network. It also provides connection to other networks. 2. Home Location Register (HLR)

The HLR is considered as a very important database that stores information of the suscribers belonging to the covering area of a MSC. It also stores the current location of these subscribers and the services to which they have access. The location of the subscriber corresponds to the SS7 address of the Visitor Location Register (VLR) associated to the terminal.

3. Visitor Location Register (VLR)

The VLR contains information from a subscriber's HLR necessary in order to provide the subscribed services to visiting users. When a subscriber enters the covering area of a new MSC, the VLR associated to this MSC will request information about the new subscriber to its corresponding HLR. The VLR will then have enough information in order to assure the subscribed services without needing to ask the HLR each time a communication is established. The VLR is always implemented together with a MSC; so the area under control of the MSC is also the area under control of the VLR.

4. The Authentication Center (AuC) The AuC register is used for security purposes. It provides the parameters needed

for authentication and encryption functions. These parameters help to verify the user's identity.

5. The Equipment Identity Register (EIR) The EIR is also used for security purposes. It is a register containing information

about the mobile equipments. More particularly, it contains a list of all valid terminals. It is identified by its International Mobile Equipment Identity (IMEI). The EIR allows then to forbid calls from stolen or unauthorized terminals (e.g, a terminal which does not respect the specifications concerning the output RF power).

6. The Operation and Support Subsystem (OSS) The OSS is connected to the different components of the NSS and to the BSC, in

order to control and monitor the GSM system. It is also in charge of controlling the traffic load of the BSS. However, the increasing number of base stations, due to the development of cellular radio networks, has provoked that some of the maintenance tasks are transfered to the BTS. This transfer decreases considerably the costs of the maintenance of the system.


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7.2.3 Code Division Multiple Access CDMA is a general category of digital wireless radio technologies that uses spread

spectrum techniques to modulate information across given bandwidth. IS-95 was the first application of CDMA, where information signals from all users

are simultaneously modulated across the entire channel band width (1.23 Mhz). Unique digital codes keep users separated on the 1.23 Mhz channel. All the three multiple Access technologies take advantage of the fact that radio

signals travel only a finite distance. The result is that frequencies can be reused with minimal interference after a minimum distance. The resulting assignment of frequencies is referred to “reuse pattern.”

CDMA doesn’t require frequency reuse pattern i.e. every code can be used in every sector of every cell.

In CDMA, timing is critical and aquired from the Global Positioning system”GPS” as accurate synchronization between cells is critical to CDMA operation.

CDMA also requires microsecond accuracy. The major advantage of CDMA when compared to the other technologies is its

efficient use of available spectrum, as bandwidth efficiecy directly to system capacity. The greater the efficiency, the more users can share the same spectrum, but it also can impact the amount of infrastructure equipment required to support a given number of users. This indirectly impacts the cost of operation.

In recent times, CDMA has gained widespread international acceptance by cellular radio system operators as an upgrade that will increase both their system capacity and the service quality.


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8.1 Data Transmission Development

One of the problems of data transmission using GSM is posed by the current comparatively user-unfriendly usage of data services in the terminals (e.g. SMS) or the complicated connection of terminal equipment via adapter.

Terminal equipment in which different functions are integrated, as well as displays optimized for each individual data transmission form provide an answer to this.

A decisive problem is posed by the comparatively low data transmission rates of GSM Phase 1 and 2. Data transmission rates of 0.3 -9.6 kbit/s compared to 64 kbit/s using ISDN are considerably too low.

To increase the data transmission rates in the Europian system new bearer services are being developed in GSM Phase 2+, which will adapt the data transmission rates to the ISDN transmission rates in various usage areas or even, be considerably above them.

1. High Speed Circuit Switched Data HSCSD

2. General Packet Radio Service GPRS 3. Enhanced Data rates for the GSM Evolution EDGE

To increase the data transmission rates in American System after deployment of

CDMA techniques IS95B was developed, which will adapt the data transmission rates to the ISDN transmission rates in various usage.

8 Data Transmission


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Chapter 2 Basic Concept of Spread Spectrum

Technology

Aim of study This chapter introduces advantage of CDMA & spread spectrum technology.

Contents Pages

1 Advantages of CDMA 2

2 Spread Spectrum Technology 6

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Basic Concept of Spread Spectrum Technology

When implemented in a cellular telephone system, CDMA technology offers many benefits to meet Mobile Radio Requirements. The following is an overview

of the advantages of CDMA.

1.1 Increased Capacity

Capacity can be increased in cellular systems in one of two ways: 1. By getting more channels per MHz of spectrum

2. By getting more channels reuse per unit of geographic area With CDMA, signals can be received in the presence of high levels of interference, All

users on a carrier share the same RF spectrum. The same CDMA RF carrier frequency is used in every cell site, and in every sector of a sector cell site.

Increasing capacity in CDMA can be done by the following techniques: -

1.1.1 Lowering Eb/No

Eb/No provides a measure of the performance of a CDMA link between the mobile and the cell. It is the ratio in dB between the energy of each information bit and the noise spectral density. The noise is a combination of background interference and the interference created by other users on the system.

CDMA describes Eb/No noise interference in terms of the Frame Erasure Rate (FER). Using an interference threshold, the CDMA system erases frames of information that contain too many errors. The FER, then, describes the number of frames that were erased due to poor quality. Therefore, as the Eb/No level increases, the FER decreases, and system voice quality is improved.

1.1.2 Voice Activity Detection

When no voice activity is detected, the vocoder will drop its encoding rate, because there is no reason to have high speed encoding of silence. The encoded rate can drop to1 kbps or less. Thus the variable rate vocoder uses up channel capacity only as needed. Since the level of "interference" created by all of the users directly determines system capacity, and voice activity detection reduces the noise level in the system, capacity can be maximized.

1.1.3 Power Control

CDMA can also increase system capacity by using POWER CONTROL, which will be discussed later.

1 Advantages of CDMA

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1.2 Improved Call Quality

Cellular telephone systems using CDMA are able to provide higher quality sound and fewer dropped calls than systems based on other technologies. Advanced error detection and error correction schemes greatly increase the likelihood that frames are interpreted correctly. Sophisticated vocoders offer high speed coding and reduce background noise.

CDMA takes advantage of various types of diversity to improve speech quality.

1.3 Simplified System Planning

All users on a CDMA carrier share the same RF spectrum.

1.4 Enhanced Privacy

CDMA is an “Anti Jamming” system. In addition, since the digitized frames of information are spread across a wide slice of spectrum, it is unlikely that a casual eavesdropper will be able to listen in on a conversation.

1.5 Improved Coverage

A CDMA cell site has a greater range than a typical analog or digital cell site. Therefore fewer CDMA cell sites are required to cover the same area. Depending on system loading and interference, the reduction in cells could be as much as 50% when compared to GSM!

CDMA's greater range is due to the fact that CDMA uses a more sensitive receiver than other technologies.

1.6 Increased Portable Talk Time

Because of precise power control and other system characteristics, CDMA subscriber units normally transmit at only a fraction of the power of analog and TDMA phones. This will enable portables to have longer talk and standby time. (This direct comparison assumes, of course, similar cell sizes between the CDMA and analog or TDMA systems.)

1.7 Bandwidth on Demand

A wideband CDMA channel provides a common resource that all mobiles in a system utilize based on their own specific needs. At any given time, the portion of this "bandwidth pool" that is not used by a given mobile is available for use by any other mobile. This provides a tremendous amount of flexibility - a flexibility that can be exploited to provide powerful features, such as higher data rate services. In addition, because mobiles utilize the "bandwidth pool" independently, these features can easily coexist on the same CDMA

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The major concern in Wireless is digital communication is efficient use of

Bandwidth and power. But there are scenarios where it is necessary to sacrifice the efficient use for design considerations. One such scenario is secure communication in hostile environment. This design objective is met using a modulation technique called as Spread Spectrum (SS).

Defining Spread Spectrum

A complete definition to Spread Spectrum is in two parts 1. Spread Spectrum is a means of transmission in which the data sequences occupy a

bandwidth in excess of the minimum bandwidth necessary to send it.

2. Spread Spectrum is accomplished before transmission through the use of a code that is independent of data sequences .The same code is used at the receiver to despread the received signal so that the original data sequence may be recovered.

In CDMA each user is assigned a unique code sequence it uses to encode its information-bearing signal. The receiver, knowing the code sequences of the user, decodes a received signal after reception and recovers the original data. This is possible since the crosscorrelations between the code of the desired user and the codes of the other users are small. Since the bandwidth of the code signal is chosen to be much larger than the bandwidth of the information-bearing signal, the encoding process enlarges (spreads) the spectrum of the signal and is therefore also known as spread-spectrum modulation. The resulting signal is also called a spread-spectrum signal, and CDMA is often denoted as spread-spectrum multiple access (SSMA) the spectral spreading of the transmitted signal gives to CDMA its multiple access capability. It is therefore important to know the techniques necessary to generate spread-spectrum signals and the properties of these signals. A spread-spectrum modulation technique must be fulfill two criteria:

The transmission bandwidth must be much larger than the information bandwidth. The resulting radio-frequency bandwidth is determined by a function other than the

information being sent (so the bandwidth is statistically independent of the information signal).

The ratio of transmitted bandwidth to information bandwidth is called the processing gain, Gp, of the spread-spectrum system; the receiver correlates the received signal with a synchronously generated replica of the spreading code to recover the original information-bearing signal. This implies that the receiver must know the code used to modulate the data.

Because of the coding and the resulting enlarged bandwidth, SS signals have a number of properties that differ from the properties of narrowband signals. The most interesting ones, from the communication systems point of view, are discussed below.

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2 Spread Spectrum Technology

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2.1 Properties of SS signals

• Multiple Access Capability

If multiple users transmit a spread-spectrum signal at the same time, the receiver will still be able to distinguish between the users provided each user has a unique code that has a sufficiently low cross-correlation with the other codes. Correlating the received signal with a code signal from a certain user will then only despread the signal of this user, while the other spread-spectrum signals will remain spread over a large bandwidth. Thus, within the information bandwidth the power of the desired user will be larger than the interfering power provided there are not too many interferers, and the desired signal can be extracted.

• Protection Against Multipath Interference In a radio channel there is not just one path between a transmitter and receiver. Due to reflections (and refractions) a signal will be received from a number of different

paths. The signals of the different paths are all copies of the same transmitted signal but with different amplitudes, phases, delays, and arrival angles. Adding these signals at the receiver will be constructive at some of the frequencies and destructive at others. In the time domain, this results in a dispersed signal. Spread-spectrum modulation can combat this multipath interference.

• Privacy & Interference Rejection The transmitted signal can only be despread and the data recovered if the receiver

knows the code. Cross-correlating the code signal with a narrowband signal will spread the power of the narrowband signal thereby reducing the interfering power in the information bandwidth.

• Anti-Jamming capability This is more or less the same as interference rejection except the interference is now

willfully inflicted on the system. It is this property, together with the next one, that makes spread-spectrum modulation attractive for military applications.

• Low Propability of Interception Because of its low power density, the spread-spectrum signal is difficult to detect and

intercept by a hostile listener.

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2.2 Spread-Spectrum Multiple Access (SS-MA)

2.2.1 Direct Sequence Spread Spectrum (DS-SS)

In DS-CDMA the modulated information bearing signal (the data signal) is directly modulated by a digital, discrete-time, discrete-valued code signal. The data signal can be either analog or digital; in most cases it is digital.

In the case of a digital signal the data modulation is often omitted and the data signal is directly multiplied by the code signal and the resulting signal modulates the wideband carrier. It is from this direct multiplication that the direct sequence CDMA gets its name.

After transmission of the signal, the receiver uses coherent demodulation to despread the SS signal, using a locally generated code sequence. To be able to perform the dispreading operation, the receiver must not only know the code sequence used to spread the signal, but the codes of the received signal and the locally generated code must also be synchronized. This synchronization must be accomplished at the beginning of the reception and maintained until the whole signal has been received. The code synchronization/tracking block performs this operation. After despreading a data modulated signal results, and after demodulation the original data can be recovered.

2.2.2 Advantages of DS-SS:

The generation of the coded signal is easy. It can be performed by a simple multiplication.

Since only one carrier frequency has to be generated, the frequency synthesizer (carrier generator) is simple.

Coherent demodulation of the DS signal is possible.

No synchronization among the users is necessary.

2.2.3 Disdvantages of DS-SS:

It is difficult to acquire and maintain the synchronization of the locally generated code signal and the received signal. Synchronization has to be kept within a fraction of the chip time.

For correct reception the synchronization error of locally generated code sequence and the received code sequence must be very small, a fraction of the chip time.

The power received from users close to the base station is much higher than that received from users further away. Since a user continuously transmits over the whole bandwidth, a user close to the base will constantly create a lot of interference for users far from the base station, making their reception impossible. This near-far effect can be solved by applying a power control algorithm so that all users are received by the base station with the same average power. However this control proves to be quite difficult.

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2.3 FREQUENCY HOPPING Spread Spectrum (FH-SS)

In frequency hopping CDMA, the carrier frequency of the modulated information signal is not constant but changes periodically. During time intervals T the carrier frequency remains the same, but after each time interval the carrier hops to another (or possibly the same) frequency. The hopping pattern is decided by the code signal.

If the hopping rate is (much) greater than the symbol rate, one speaks of a fast frequency hopping (F-FH). In this case the carrier frequency changes a number of times during the transmission of one symbol, so that one bit is transmitted in different frequencies. If the hopping rate is (much) smaller than the symbol rate, one speaks of slow frequency hopping (S-FH).

2.3.1 Advantages of FH-SS:

Synchronization is much easier with FH-CDMA than with DS-CDMA. With FH CDMA synchronization has to be within a fraction of the hop time. Since spectral spreading is not obtained by using a very high hopping frequency but by using a large hop-set, the hop time will be much longer than the chip time of a DS-CDMA system. Thus, an FH-CDMA system allows a larger synchronization error.

The different frequency bands that an FH signal can occupy do not have to be contiguous because we can make the frequency synthesizer easily skip over certain parts of the spectrum. Combined with the easier synchronization, this allows much higher spread-spectrum bandwidths.

The probability of multiple users transmitting in the same frequency band at the same time is small. A user transmitting far from the base station will be received by it even if users close to the base station are transmitting, since those users will probably be transmitting at different frequencies. Thus, the near-far performance is much better than that of DS.

Because of the larger possible bandwidth a FH system can employ, it offers a higher possible reduction of narrowband interference than a DS system.

2.3.2 Disdvantages of FH-SS:

A highly sophisticated frequency synthesizer is necessary. An abrupt change of the signal when changing frequency

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2.4 TIME HOPPING Spread Spectrum (TH-SS)

In time hopping CDMA the data signal is transmitted in rapid bursts at time intervals determined by the code assigned to the user. The time axis is divided into frames, and each frame is divided into M time slots. During each frame the user will transmit in one of the M time slots. Which of the M time slots is transmitted depends on the code signal assigned to the user. Since a user transmits all of its data in one, instead of M time slots, the frequency it needs for its transmission has increased by a factor M.

2.4.1 Advantages of TH-SS:

Implementation is simpler than that of FH-CDMA and the near-far problem is much less of a problem since TH-CDMA is an avoidance system, so most of the time a terminal far from the base station transmits alone, and is not hindered by transmissions from stations close by.

The multiple access capability of THSS signals is acquired in the same manner as that of the FH-SS signals; namely, by making the probability of users’ transmissions in the same frequency band at the same time small. In the case of time hopping all transmissions are in the same frequency band, so the probability of more than one transmission at the same time must be small. This is again achieved by assigning different codes to different users. If multiple transmissions do occur, error-correcting codes ensure that the desired signal can still be recovered. If there is synchronization among the users, and the assigned codes are such that no more than one user transmits at a particular slot, then the THCDMA reduces to a TDMA scheme where the slot in which a user transmits is not fixed but changes from frame to frame.

2.4.2 Disdvantages of TH-SS:

In the time hopping CDMA, a signal is transmitted in reduced time. The signaling rate, therefore, increases and dispersion of the signal will now lead to overlap of adjacent bits. Therefore, no advantage is to be gained with respect to multipath interference rejection.

It takes a long time before the code is synchronized, and the time in which the receiver has to perform the synchronization is short.

If multiple transmissions occur, a large number of data bits are lost, so a good error-correcting code and data interleaving are necessary.

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2.5 HYBRID SYSTEMS

The hybrid CDMA systems include all CDMA systems that employ a combination of two or more of the above-mentioned spread-spectrum modulation techniques or a combination of CDMA with some other multiple access technique. By combining the basic spread-spectrum modulation techniques, we have four possible hybrid systems:

DS/FH, DS/TH, FH/TH, and DS/FH/TH; and by combining CDMA with TDMA or multicarrier modulation we get two more:

CDMA/TDMA and MC-CDMA. The idea of the hybrid system is to combine the specific advantages of each of the modulation techniques.

2.5.1 Advantages of H-SS:

If we take, for example, the combined DS/FH system we have the advantage of the anti-multipath property of the DS system combined with the favorable near-far operation of the FH system.

2.5.2 Disdvantages of H-SS:

Of course, the disadvantage lies in the increased complexity of the transmitter and receiver.

Coherent demodulation is difficult because of the problems in maintaining phase relationships during hopping.

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Chapter 3 CDMA codes and its usage

Aim of study This chapter introduces Iterium Standard-95 System & Pseduo Random Noise

Sequence.

Contents Pages

1 Iterium Standard-95 System 2

2 Pseduo Random Noise Sequence 4

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Rectangle

1.1 IS-95 Interim Standard 95 (IS-95) is a U.S. digital cellular system based on CDMA that

allows each user within a cell and in adjacent cells to use the same radio channel. Each IS-95 channel occupies 1.23MHz of spectrum in each one-way link; the user

data is spread to a channel chip rate of 1.2288MHz. IS-95 uses a different modulation and spreading technique for the forward and reverse links. On the forward link, the base station simultaneously transmits the user data for all mobiles in the cell by using different spreading sequence for each mobile. The user data is encoded, interleaved, and spread by one of sixty-four orthogonal spreading sequences (Walsh functions).

To avoid interference, all signals in a particular cell are scrambled using a pseudorandom sequence of length 215-1 chips.

CDMA base stations transmit information in four logical channel formats:

Pilot channels, sync channels, paging channels, and traffic channels.

On the reverse link, all mobiles respond in an asynchronous fashion. The user data is encoded, interleaved, and then blocks of 6 bits are mapped to one of the 64 orthogonal Walsh functions. Finally, the data is spread by a user specific code of 42 bits (channel identifier) and the base station pseudorandom sequence of length 215 chips. The reverse channel is organized in:

Access channels and traffic channels. At both the base station and the terminal, Rake receivers are used to resolve and

combine multipath components, in order to improve the link quality. In IS-95, a three-finger Rake receiver is used at the base station.

CDMA codes and its usage

1 Iterium Standard-95 System

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2.1 PN Sequences • What are PN sequences?

A Pseudo-random Noise (PN) sequence is a sequence of binary numbers, e.g. ±1, which appears to be random; but is in fact perfectly deterministic. The sequence appears to be random in the sense that the binary values and groups or runs of the same binary value occur in the sequence in the same proportion they would if the sequence were being generated based on a fair "coin tossing" experiment. In the experiment, each head could result in one binary value and a tail the other value. The PN sequence appears to have been generated from such an experiment. A software or hardware device designed to produce a PN sequence is called a PN generator.

Pseudo-random noise sequences or PN sequences are known sequences that exhibit the properties or characteristics of random sequences. They can be used to logically isolate users on the same frequency channel. They can also be used to perform scrambling as well as spreading and despreading functions. The reason we need to use PN sequences is that if the code sequences were deterministic, then everybody could access the channel. If the code sequences were truly random on the other hand, then nobody, including the intended receiver, would be able to access the channel. Thus, using a pseudo-random sequence makes the signal look like random noise to everybody except to the transmitter and the intended receiver.

• Why PN sequence is chosen as a noise like waveform? To know that we have to understand what is called “white Noise”. The adjective “white” is used in the sense that white light contains equal amounts of all

frequencies within the visible band of electromagnetic radiation. It has power spectral density independent of the operating frequency. We express the

power spectral density of white noise by Sw (f) = No/2 ... No = KT0 watts /Hz. where K is Boltzman constant & T0 is the equivalent noise temperature.

• Equivalent noise temperature of a system “T0”: - It is the temperature at which a noisy resistor has to be maintained such that, by

connecting the resistor to the input of a noiseless version of the system, it produces the same available noise in the actual system it depends only on the parameters of the system

Since the auto correlation function is the inverse Fourier of the power spectral density it follows that for white noise, the auto correlation function of white noise consists of a delta function weighted by the factor No/2 and occurring at τ = 0.

Accordingly, any two different samples of white noise, no matter how closely together in time, they are taken, are uncorrelated. So we have to search for a code sequence has a noise like wave or almost has autocorrelation function near that of white noise.


2 Pseduo Random Noise Sequence

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2.1.1 PN Sequence Generation

These sequences are easily generated by using an M-bit linear feedback shift register with the appropriate feedback taps, e.g. as shown in Fig. For M = 5. With the appropriate taps, the length (N) of the serial bit stream at the output will be a maximum (Lmax): N = Lmax = 2M - 1

The meaning of bit-stream length in this context is the maximum length of the bit sequence before it starts repeating itself. PN sequences of maximum length are called maximal linear code sequences, but because non-maximal PN sequences are rarely used in SS systems, “PN sequences” will be used to denote maximal linear code sequences for this document. Also “PN codes” or “PN code sequences” will be used synonymously with “PN sequences”. The feedback taps are added modulo-2 (exclusive OR’ed) and fed to the input of the initial shift register. Only particular tap connections will yield a maximum length for a given shift register length. These maximal length PN codes have the following properties:

1. Code balance: The number of ones and the number of zeros differ by only 1, i.e., there is 1 more one

than the number of zeros. This particularly useful when the channel is AC coupled (no DC transmission).

2. Autocorrelation: Using signaling values of ±1, the autocorrelation of a PN sequence has a value of –1 or

all phase shifts of more than one bit time. For no has shift (perfect alignment with itself), the autocorrelation has a value of N, the sequence length.

3. Modulo-2 addition: Modulo-2 addition of a PN sequence with a shifted version of itself results in a differently

shifted version of itself. 4. Shift Register States:

The binary number represented by the M bits in the shift register randomly cycle through all 2M values, except for 0, in successive 2M-1 clocks.

If the value of 0 (all shift register bits are 0) is ever present in the shift register, it will stay in that state until reloaded with a nonzero value.


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2.1.2 A PN Generator Example

A PN generator is typically made of N cascaded flip-flop circuits and a specially selected feedback arrangement. The flip-flop circuits when used in this way are called a shift register since each clock pulse applied to the flip-flops causes the contents of each flip-flop to be shifted to the right. The feedback connections provide the input to the left-most flip-flop. With N binary stages, the largest number of different patterns the shift register can have is 2N. The all-binary-zero state, however, is not allowed because it would cause all remaining states of the shift register and its outputs to be binary zero. The all-binary-ones state does not cause a similar problem of repeated binary ones provided the number of flip-flops input to the modulo-2 adder is even. The period of the PN sequence is therefore 2N -1. For example, starting with the register in state 001, the next 7 states are 100, 010,101, 110, 111, 011, and then 001 again and the states continue to repeat. The output taken from the right-most flip-flop is 1001011 and then repeats. With the three-stage shift register, the period is 23-1 or 7.


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2.2 Types of PN Sequences in CDMA There are two different types of PN codes and one output of Hadamard Matrix

used in IS-95 CDMA Technology:

1. Short PN code 2. Long PN code 3. Walsh codes

IS-95 uses the two types of maximum-length PN generators to spread the signal power uniformly over the physical bandwidth of about 1.25 MHz. The PN spreading on the reverse link also provides near orthogonality of and hence, minimal interference between signals from each mobile. This allows reuse of the band of frequencies available, which is a major advantage of CDMA.

2.2.1 Short Code: A 15-stage linear shift register generates the short PN code. Therefore, the maximum

length of the Short PN Code is L = 2N-1 = 215-1 = 32,768-1 chips.

By implementation, an extra chip is inserted at the end of the sequence, yielding a sequence of length L=32,768 chips. The short PN code runs at a speed of 1,228,800 chips per second. This yields a repetition cycle of 32,768/1,228,800=26.67 ms.

The short PN code consist of two PN Sequences I and Q each 32,768 chips long generated in similar but differently tapped 15 bit shift register, the two sequences scramble the information on the I and Q phase channels. § These codes are used for cell identification in a reused cell. § The chip rate of the short PN code is 1.2288 Mcps.

2.2.2 Long Code: The PN chips from the long code are used to provide several randomizing functions in

the IS-95 system. These include providing chips for message-scrambling on the forward and reverse links, for identifying individual mobiles and access channels on the reverse links by using unique offsets for each entity and for randomizing the location of the power control bits on the forward traffic channels. A 42-stage linear shift register generates the long PN code. Therefore, the maximum length of the long PN code is

L = 2N-1 = 242-1 = 4.4 x 1012 = 4.4 trillion chips. The Long PN Code also runs at a speed of 1,228,800 chips per second. This yields a

repetition cycle of 4.4 x 1012/1,228,800 = 41-42 days. The long PN code is generated in a 42-stage linear shift register generator with the

output of the 42nd stage input into the first stage and modulo-2 added with the outputs of stages 1, 2, 3, 5, 6, 7, 10, 16, 17, 18, 19, 21, 22, 25, 26, 27, 31, 33, and 35. The output of the long code generator is taken after the output of each flip-flop in the generator has been added with a corresponding bit in a 42-bit mask, which is unique to each user, access, and paging channel. § Base band data scrambling in the forward link § Base band data spreading in the reverse link


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2.2.3 Walsh code: In 1923, J.L. Walsh introduced a complete set of orthogonal codes, based on rearranging the Rademacher code. These codes are also binary valued codes. The Walsh code, also known as the Hadamard code, is a set of 64 orthogonal codes, there purpose is to provide:

1. Forward channel spreading over the 1.2288MHz band; 2. Unique identification to a mobile.

The chip rate (code rate) of a Walsh code is 1.2288 Mchips per second (Mcps). The four different types of forward channels are designated as follows:

1. Pilot channel: W0 (Walsh code 0); 2. Paging channel: W1 to W7 (unused paging codes can be used for traffic); 3. Sync channel: W32; 4. Traffic channel: W8 to W31 and W33 to W63.


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2.3 Correlation between PN sequences The correlation of two random variables x(t) and y(t), is a time-shift comparison which

expresses the degree of similarity or the degree of likeness between the two variables. The Auto-Correlation function R, provides the degree of similarity between a random variable x(t) and a time-shifted version of x(t).

Likewise, the cross-correlation function provides the degree of similarity, or the degree of likeness between a random variable x(t) and time-shifted version of another random variable y(t). To get the average value of the auto-correlation or cross-correlation, a normalization by the sequence length L is required. Consider Ci(t) and the time-shifted version of itself, say Ci(t-1)

Ci(t) = 1 0 0 1 1 1 0 Ci(t-1) = 0 0 1 1 1 0 1

When corresponding bits from the two sequences have the same parity (or match each other), we call the match an agreement "A". Likewise, when corresponding bits from the two sequences do not have the same Parity (do not match each other), we call the mismatch a disagreement "D" .By counting all the agreements and all the disagreements over the full length L of the sequence, a measure of correlation can be estimated as:

Correlation = Total number of "A" - Total number of "D" Now, consider the reference PN code C i(t) and its time-shifted versions as shown. Now let us compute the correlation of C i(t) and Ci(t-t), for all suitable values of t (here

from 0 to 7). In general, it can be shown that the full-length auto-correlation function (R) of PN codes

or PN sequences is characterized by a large positive number equal to the length of the PN sequence (R=2n-1) when time shift=0, and -1 for all time-shifts equal or greater than the duration of one chip. So when normalized by the length, the auto-correlation function is equal to 1 at time-shift zero and is very small (-1/L) for all values of time shifts equal or greater than one chip.

In summary, the auto-correlation function of PN codes is a two-value function. Its maximum value occurs when the time-shift parameter is zero. For all other values equal to or greater than one chip, the correlation function is -1.

• Orthogonality of PN sequences Consider the reference PN Code Cj(t) and the time-shifted versions of another code Ci(t)

as shown. Let us compute the cross-correlation of Cj(t) and Ci(t-t) for all suitable values of t (0 to 7).

Two PN sequences Ci(t) and Cj(t) are said to be orthogonal if and only if their respective normalized correlation function is equal to 1 at a time-shift of zero and their cross correlation function is equal to zero for all time-shift values. As shown above, averaged over the code length, the cross-correlation function of PN sequences is not zero. As a result, PN sequences are not perfectly orthogonal.


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2.4 Process Gain and its Benefits The primary benefit of processing gain is its contribution towards jamming resistance to

the DSSS signal. The PN code spreads the transmitted signal in bandwidth and it makes it less susceptible to narrowband interference within the spread BW. The receiver of a DSSS system can be viewed as unspreading the intended signal and at the same time spreading the interfering waveform. This operation is best illustrated on Figure, which, depicts the power spectral density (psd) functions of the signals at the receiver input, the despread signal, the band pass filter power transfer function, and the band pass filter output. The figure graphically describes the effect of the processing gain on a jammer. The jammer is narrow, and has a highly peaked psd, while the psd of the DSSS is wide and low. The despreading operation spreads the jammer power psd and lowers its peak, and the BPF output shows the effect on the signal to jammer ratio.

If for example, BPSK modulation is used and an Eb/No of lets say 14dB is required to achieve a certain BER performance, when this waveform is spread with a processing gain of 10dB then the receiver can still achieve its required performance with the signal having a 4dB power advantage over the interference. This is derived from the 14dB required minus the 10dB of PG.

The higher the processing gain of the DS-SS waveform the more the resistance to interference of the DSSS signal. If a code with a length of 16 bits is to be used then the processing gain is equivalent to 10 Log[16] dB or 12.04dB.


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We can define GP as:

Where SNRo and SNRi are the output and input SNR of the correlator, respectively. Where BWD and BWSS are the bandwidth of the data before and after SS modulation. Fig.11


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2.5 Spreading Code Acquisition and Tracking No matter which form of spread spectrum technique we employ, we need to have the

timing information of the transmitted signal in order to despread the received signal and demodulate the despread signal. For a DS-SS system, we see that if we are off even by a single chip duration, we will be unable to despread the received spread spectrum signal, since the spread sequence is designed to have a small out-of-phase autocorrelation magnitude. Therefore, the process of acquiring the timing information of the transmitted spread spectrum signal is essential to the implementation of any form of spread spectrum technique. Usually the problem of timing acquisition is solved via a two-step approach:

• Initial code acquisition (coarse acquisition or coarse synchronization), which synchronizes the transmitter and receiver.

• Code tracking, which performs and maintains fine synchronization between the transmitter and receiver.

Given the initial acquisition, code tracking is a relatively easy task and is usually accomplished by a delay lock loop (DLL). The tracking loop keeps on operating during the whole communication period. If the channel changes abruptly, the delay lock loop will lose track of the correct timing and initial acquisition will be reperformed. Sometimes, we perform initial code acquisition periodically no matter whether the tracking loop loses track or not.

Compared to code tracking, initial code acquisition in a spread spectrum system is usually very difficult. First, the timing uncertainty, which is basically determined by the transmission time of the transmitter and the propagation delay, can be much longer than a chip duration. As initial acquisition is usually achieved by a search through all possible phases (delays) of the sequence, a larger timing uncertainty means a larger search area. Beside timing uncertainty, we may also encounter frequency uncertainty that is due to Doppler shift and mismatch between the transmitter and receiver oscillators. Thus this necessitates a two-dimensional search in time and frequency. Moreover, in many cases, initial code acquisition must be accomplished in low signal-to-noise-ratio environments and in the presence of jammers. The possibility of channel fading and the existence of multiple access interference in CDMA environments can make initial acquisition even harder to accomplish.

The problem of achieving synchronization in various fading channels and CDMA environments is difficult and is currently under active investigation. In many practical systems, side information such as the time of the day and an additional control channel, is needed to help achieve synchronization.


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2.5.1 Initial Code Acquisition

As mentioned before, the objective of initial code acquisition is to achieve a coarse synchronization between the receiver and the transmitted signal. In a DS-SS system, this is the same as matching the phase of the reference-spreading signal in the despreader to the spreading sequence in the received signal. We are going to introduce several acquisition techniques, which perform the phase matching just described.

• Acquisition strategies

Serial search

The first acquisition strategy we consider is serial search. In this method, the acquisition circuit attempts to cycle through and test all possible phases one by one (serially) as shown in Figure.

The circuit complexity for serial search is low. However, penalty time associated with a miss is large.

Therefore we need to select a larger integration (dwell) time to reduce the miss probability. This, together with the serial searching nature, gives a large overall acquisition time (i.e., slow acquisition).


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Parallel search

Unlike serial search, we test all the possible phases simultaneously in the parallel search strategy as shown in figure. Obviously, the circuit complexity of the parallel search is high. The overall acquisition time is much smaller than that of the serial search.


Fig.14

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CDMA Air Interface Overview

2.5.2 Code Tracking

The purpose of code tracking is to perform and maintain fine synchronization. A code-tracking loop starts its operation only after initial acquisition has been achieved. Hence, we can assume that we are off by small amounts in both frequency and code phase. A common fine synchronization strategy is to design a code tracking circuitry, which can track the code phase in the presence of a small frequency error. After the correct code phase is acquired by the code tracking circuitry, a standard phase lock

Loop (PLL) can be employed to track the carrier frequency and phase. In this section, we give a brief introduction to a common technique for code tracking, namely, the early-late gate delay-lock loop (DLL).

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Chapter 4 CDMA Air Interface Overview

Aim of study This chapter introduces CDMA air links and channels.

Contents Pages

1 CDMA Air Links and Channels 2

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1.1 CDMA Air- Links The IS-95 CDMA system is unique in that its forward and reverse links have different link

structure. This is necessary to accommodate the requirements of a land-mobile communication system. The forward link consists of four types of logical channels: pilot, sync, paging, and traffic channels. There is one pilot channel, one sync channel, up to seven paging channels, and several traffic channels. Each of these forward-link channels is first spread orthogonally by its Walsh function, and then a quadrature pair of short PN sequences spreads it.

All channels are added together to form the composite SS signal to be transmitted on the forward link.

The reverse link consists of two types of logical channels: access and traffic channels. Each of these reverse-link channels is spread orthogonally by a unique long PN sequence; hence, each channel is identified using the distinct long PN code. The reason that a pilot channel is not used on the reverse link is that it is impractical for each mobile to broadcast its own pilot sequence.

Forward Link

We defined the structure of a Hadamard matrix and described how Walsh codes are generated using such a matrix. The IS-95 CDMA system uses a 64 by 64 Hadamard matrix to generate 64 Walsh functions that are orthogonal to each other, and each of the logic channels on the forward link is identified by its assigned Walsh function.

Reverse Link The reverse link supports two types of logical channels: Access channels and Traffic

channels.

Because of the noncoherent nature of the reverse link, Walsh functions are not used for channelization. Instead, Long PN sequences are used to distinguish the users from one another.

1 CDMA Air- Links and Channels

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Fig.1

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1.2 Forward Link Channels

1.2.1 Pilot Channel The pilot channel is is used by the base station to provide a reference for all mobile

stations. It provides a phase reference for coherent demodulation at the mobile receiver to enable coherent detection. It is assigned the Walsh code W0.

The pilot signal level for all base stations is kept about 4 to 6 dB higher than the traffic channel with a constant signal power. The pilot is used for comparisons of signal strength between different base stations to decide when to perform handoff. The pilot signals from all base stations use the same PN sequences, but each base station is identified by a unique time offset. These offsets are in increments of 64 chips to provide 512 unique offsets.

Each terminal segregates the set of PN Offset values (and implicitly the set of base stations) in a system into four categories:

• The active list contains base stations currently used for traffic channel transmissions. In a soft handoff condition, there is more than one base station in this list.

• The candidate list consists of base stations classified by the terminal, on the basis of measured signal quality, as available for traffic channel transmissions.

• The neighbor list is a set of nearby base stations that could soon be available for handoff.

• The remaining list contains the base stations that are not in any of the other categories.

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1.2.2 Sync Channel

Unlike the pilot channel, the sync channel carries baseband information. The information is contained in the sync channel message that notifies the mobile of important information about system synchronization and parameters.

The baseband information is error protected and interleaved, it is then spread by Walsh function 32 and further spread by the PN sequence that is identified with the serving sector. The baseband information is at a rate of 1.2 Kbps.

The Sync Channel is used with the pilot channel to acquire initial time synchronization. The Sync channel message parameters are:

• System Identification (SID)

• Network Identification (NID)

• Pilot short PN sequence offset index

• Long-code state

• System time

• Offset of local time

• Daylight saving time indicator

• Paging Channel data rate (4.8 or 9.6kbps).

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1.2.3 Paging Channel

Similar to the sync channel, the paging channel also carries baseband information. But unlike the sync channel, the paging channel transmits at higher rates; it can transmit

at either 4.8 or 9.6 Kbps. As shown in Figure, the baseband information is first error protected, and then if the data rate is at 4.8 Kbps, the bits are repeated once. Otherwise, they are not repeated. Following interleaving, the data is first scrambled by a decimated long PN sequence, then it is spread by a specific Walsh function assigned to that paging channel and further spread by the short PN sequence assigned to the serving sector. Also note from Figure that the long PN code undergoes a decimation ratio of 64:1 (i.e., from 1.2288 Mcps to 19.2 Ksps). The long-code generator itself is masked with a mask specific to each unique paging channel number (i.e., 1 through 7). Therefore, the longcode mask used for paging channel 1 (spread by Walsh function 1) is different from that used for paging channel 3 (spread by Walsh function 3).

Some of the messagescarried by the paging channel include:

• System Parameter message: such as base station identifier, the number of paging channels, and the page channel number.

• Access Parameters message: parameters required by the mobile to transmit on an access channel.

• Neighbor List Message: information about neighbor base station parameters, such as the PN Offset.

• CDMA Channel List message: provides a list of CDMA carriers. • Page message: provides a page to the mobile station. • Channel Assignment message: to inform the mobile station to tune to a new

frequency. • Data Burst message: data message sent by the base station to the mobile. • Authentication Challenge: allows the base station to validate the mobile identity.

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1.2.4 Rate Set1 Traffic Channel

The forward traffic channel is used to transmit user data and voice; signaling messages are also sent over the traffic channel.

For Rate Set 1, the vocoder is capable of varying its output data rate in response to speech activities. Four different data rates are supported: 9.6, 4.8, 2.4, and 1.2 Kbps. For example, during quiet periods of speech, the vocoder may elect to code the speech at the lowest rate of 1.2 Kbps.

The baseband data from the vocoder is convolutionally encoded for error protection. For Rate Set 1, a rate 1/2 convolutional encoder is used. The encoding effectively doubles the data rate. After convolutional encoding, the data undergoes symbol repetition, which repeats the symbols when lower rate data are produced by the vocoder. The following is the repetition scheme:

• When the data rate is 9.6 Kbps, the code symbol rate (at the output of the convolutional encoder) is 19.2 Ksps. In this case, no repetition is performed.

• When the data rate is 4.8 Kbps, the code symbol rate is 9.6 Ksps; each symbol is repeated once, yielding a final modulation symbol rate of 19.2 Ksps.

• When the data rate is 2.4 Kbps, the code symbol rate is 4.8 Ksps; each symbol is repeated three times, yielding a final modulation symbol rate of 19.2 Ksps.

• When the data rate is 1.2 Kbps, the code symbol rate is 2.4 Ksps; each symbol is repeated seven times, yielding a final modulation symbol rate of 19.2 Ksps.

The reason for repeating symbols is to reduce overall interference power at a given time when lower rate data are transmitted.

In a real CDMA system, when the vocoder is transmitting at 4.8 Kbps, the energy per symbol transmitted is one-half that of 9.6 Kbps. When the vocoder is transmitting at 2.4 Kbps, the energy per symbol transmitted is oneforth that of 9.6 Kbps, and when the vocoder is transmitting at 1.2 Kbps, the energy per symbol transmitted is one-eighth that of 9.6 Kbps.

After symbol repetition, the data is interleaved to combat fading (see Figure), and then the interleaved data is scrambled by a decimated long PN sequence. A long PN code generator generates the long PN sequence. The generator outputs a long PN sequence at 1.2288 Mcps. Because the data rate at the interleaver output is 19.2 Ksps, the PN sequence is decimated by a ratio of 64:1 to also achieve a rate of 19.2 Kcps; the decimated long PN sequence at 19.2 Kcps is then multiplied with the 19.2-Ksps data stream. Note that the long-code generator produces the long PN sequence using a mask that is specific to the mobile.

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1.2.5 Rate Set 2 Traffic Channel

The forward traffic channel structure is similar for Rate Set 2. The Rate Set 2 vocoder codes speech at higher rates, and it delivers a better voice quality than that of Rate Set 1. The Rate Set 2 vocoder supports four variable rates: 14.4, 7.2, 3.6, and 1.8 Kbps.

Note that in order to maintain the output of the block interleaver at 19.2 Ksps, the

rate of the convolutional encoder is increased to R = 3/4.

Fig.7

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1.3 Reverse Link channels The reverse link supports two types of logical channels: Access channels and Traffic

channels.

Because of the noncoherent nature of the reverse link, Walsh functions are not used for channelization. Instead, Long PN sequences are used to distinguish the users from one another.

1.3.1 Access Channel

The mobile communicates with the base station when it doesn’t have a traffic channel assigned using the access channel. The mobile uses this channel to make call originations and respond to pages and orders. The baseband data rate of the access channel is fixed at 4.8 Kbps.

The baseband information is first error protected by an R = 1/3 convolutional encoder. The lower encoding rate makes error protection more robust on the reverse link, which is often the weaker of the two links. The symbol repetition function repeats the symbol once, yielding a code symbol rate of 28.8 Ksps. The data is then interleaved to combat fading. Following interleaving, the data is coded by a 64-ary orthogonal modulator.

The set of 64 Walsh functions is used, but here the Walsh functions are used to modulate, or represent, groups of six symbols. The reason for orthogonal modulation of the symbols is again due to the noncoherent nature of reverse link. When a user’s transmission is not coherent, the receiver (at the base station) still has to detect each symbol correctly. Making a decision of whether or not a symbol is +1 or -1 may be difficult during one symbol period.

However, if a group of six symbols is represented by a unique Walsh function, then the base station can easily detect six symbols at a time by deciding which Walsh function is sent during that period. The receiver can easily decide which Walsh function is sent by correlating the received sequence with the set of 64 known Walsh functions. Note that on the forward link, Walsh functions are used to distinguish among the different channels. On the reverse link, Walsh functions are used to distinguish among the different symbols (or among groups of six symbols). The orthogonally modulated data at 4.8 Ksps (modulation symbols) or at 307.2 Ksps (code symbols) are then spread by the long PN sequence. The long PN sequence is running at 1.2288 Mcps, and the bandwidth of the data after spreading is 1.2288 Mcps. Remember that the long PN sequence is used to distinguish the access channel from all other channels that occupy the reverse link. The data is further scrambled in the I and the Q paths by the short PN sequences (also running at 1.2288 Mcps) defined in the IS-95 standard.

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Fig.8

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1.3.1 Access Channel (Cont.) Is used by a terminal without a call in progress to send messages to the base station for

three principal purposes: to originate a call, to respond to a paging message, and to register its location. Each base station operates with up to 32 access channels. The messages carried by the access channel include:

• Registration Message: sends to the base station information necessary to page the mobile, such as: location, status, and identification.

• Order message: to transmit information such as base station challenge, mobile station acknowledgement, local control response, and mobile station reject.

• Data Burst message: user-generated data message sent by the mobile station to the base station.

• Origination message: allows the mobile station to place a call’ sending dialed digits. • Page Response message: used to respond to a page. • Authentication Challenge Response message: contains necessary information to

validate the mobile station’s identity.

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1.3.2 Traffic Channel

The reverse traffic channel is used to transmit user data and voice; signaling messages are also sent over the traffic channel. The structure of the reverse traffic channel is similar to that of the access channel. The major difference is that the reverse traffic channel contains a data burst randomizer.

The orthogonally modulated data is fed into the data burst randomizer. The function of the data burst randomizer is to take advantage of the voice activity factor on the reverse link. Recall that the forward link uses a different scheme to take advantage of the voice activity factor, when the vocoder is operating at a lower rate, the forward link transmits the repeated symbols at a reduced energy per symbol and thereby reduces the forward-link power during any given period.

The approach taken to reduce reverse-link power during quieter periods of speech is to pseudorandomly mask out redundant symbols produced by symbol repetition.

This is accomplished by the data burst randomizer. The data burst randomizer generates a masking pattern of 0s and 1s that randomly masks out redundant data. The masking pattern is partially determined by the vocoder rate. If the vocoder is operating at 9.6 Kbps, then no data is masked. If the vocoder is operating at 1.2 Kbps, then the symbols are repeated seven times, and the data burst randomizer masks out, on average, seven out of eight groups of symbols.

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1.3.2 Traffic Channel (Cont.)

This channel can multiplex primary (voice) and secondary (data) or signaling traffic. Some of the typical messages that the reverse traffic channel carries are:

• Order messages: include base station challenge, parameter update confirmation, mobile station acknowledgement, service option request and response, release, connect, DTMF tone, etc.

• Authentication Challenge Response message: information to validate the mobile station.

• Data Burst message: a user-generated data message sent by the mobile to the base station.

• Pilot Strength Measurement message: information about the strength of other pilot signals that are not associated with the serving base station.

• Power Measurement Report message: sends FER statistics to the base station. • Handoff Completion message: is the mobile response to a Handoff Direction

message. • Parameter Response message: is the mobile response to the base station to a

Retrieve Parameters message.

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1.4 How calls from a base station are encoded and transmitted to a cellphone

At the base station, each voice conversation is converted into digital code and compressed with a vocoder. The vocoder output is doubled by a convolutional encoder that adds redundancy for error checking. Each bit from the encoder is replicated 64 times and exclusive OR'd with a Walsh code that is used to identify that call from the rest. The output of the Walsh code is exclusive OR'd with the next string of bits (PN sequence) from a pseudo-random number generator, which is used to identify all the calls in a particular cell's sector. At this point, there is 128 times as many bits as there were from the vocoder's output. All the calls are combined and modulated onto a carrier frequency in the 800 MHz range.

At the receiving side, the received signals are quantized (turned into bits) and run through the Walsh code and PN sequence correlation receiver to recover the transmitted bits of the original signal. When 20ms of voice data is received, a Viterbi decoder corrects the errors using the convolutional code, and that all goes to the vocoder that turns the bits back into waveforms (sound).

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1.4 How calls from a base station are encoded and transmitted to a cellphone

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Fig.12

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Chapter 5 CDMA System Aspects

Aim of study This chapter introduces power control in CDMA & handoff versus handover.

Contents Pages

1 Power Control in CDMA 2

2 Rake Receiver 9

3 Handoff Versus Handover 11

4 Multiuser Detection 14

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CDMA System Aspects

1.1 Introduction

Power control is one of the most important system requirement, and it is analyzed for cellular networks based on FDMA and TDMA, and for DS-CDMA cellular networks, In most modern systems, both base stations and mobiles have the capability of real-time (dynamic) adjustment of their transmit powers.

1.1.1 Effect of NO Power Control:

In case of no power control, if a mobile station signal is received at the base station with a too low level of received power [MS is far from the cell site, or in an unusual high attenuation channel], High level of interference is experienced by this mobile and its performance (BER) will be degraded.

On the other hand, if the received power level is too high, the performance of this mobile is acceptable, but increases interference to all other mobile stations that are using the same channel.

Fast power control greatly optimizes the system capacity,but since many subscribers transmit in the same frequency band and the same frequency can be used in each cell (re-use = 1), each user can cause interference for the others.

The power control is used to solve the called “NEAR-FAR” problem.

1.1.2 The (NEAR – FAR) Problem

In ideal cases, the power received at the BTS is identical for all UE served by the BTS (assuming the transfer rates are identical). This ideal situation also represents the maximum capacity of the cell.

Genuine fast power control is necessary because of the mobility of the UE. This mobility causes rapid variation in the attenuation of the power of the UE. Let us consider the shown example:

If the mobiles are permitted to transmit the same power from two different distances, the ratio of the received signals at the base station will be not equal. Therefore, the objective of the mobile power control is to produce a nominal received power from all mobiles in a given cell or a sector. Because of that, well-defined power control is essential for proper functioning of the DS-CDMA system. In the absence of power control the capacity of the DS-CDMA mobile system is very low, even lower than that of mobile systems based on FDMA. One of the reasons for the use of power control both in FDMA/TDMA and in DS-CDMA networks is to prolong battery life by using a minimum of transmitter power to achieve the required transmission quality. According to the above-mentioned facts, for proper operation of a modern high-capacity cellular radio system, power control is an essential feature.

1 Power Control in CDMA

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CDMA System Aspects

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CDMA System Aspects

1.2 Classification of Power Control Techniques According to what is measured to determine power control command, power

control techniques can be classified into: 1. Strength-based. 2. SIR-based.

3. BER-based. 1. Strength-based.

The strength of a signal arriving at the base station from a mobile is measured to determine whether it is higher or lower than the desired strength and then it is adjusted so it is considered the easiest method.

2. SIR-based. The measured quantity is the Signal to Interference Ratio where interference consists of

channel noise and multi-user interference. SIR-based power control reflects better system performance such as QoS and capacity. A serious problem associated with SIR-based power control is the potential to get positive feedback to endanger the stability of the system.

3. BER-based.

Bit Error Rate is defined as an average number of erroneous bits compared to the original sequence of bits. If the signal and interference powers are constant, the BER will be a function of the SIR, and in this case the QoS is equivalent. However, in reality the SIR is time-variant and thus the average SIR will not correspond to the average BER. In this case the BER is a better quality measure.

1.2.1 According to update strategies, power control

algorithms can be classified as follows: 1. Fixed step size algorithm 2. Adaptive step size to the channel variation 1. Fixed step size algorithm

Power control command in fixed step size algorithms is a simple 1-bit command. It has been shown that the inverse algorithm is superior to the fixed step size algorithm. However, the fixed step size algorithm is easier to implement because the inverse algorithm needs additional bandwidth on the return channel to carry the power control step size instead of the1-bit control command as in fixed step size algorithm. A compromise would be to use an adaptive delta-modulation algorithm.

2. Adaptive step size to the channel variation

A specific example of the adaptive step size approach is the inverse update algorithm, which increases or decreases the mobile users' transmit power by the actual difference between the received signal power and the desired received signal power.

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CDMA System Aspects

Fig.3

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CDMA System Aspects

1.2.2 According to direction of transmission , power control can be classified into:

1. Forward link (from mobiles to base stations). 2. Reverse link (from base stations to mobiles). 1. Forward link power control:

Forward link (base station to mobile) power control is a one step process .The base station controls its transmitting power so that a given mobile receives extra power to overcome fading, interference, BER, etc. In this mechanism, the cell site reduces its transmitting power while the mobile computes the frame error rate (FER). Once the mobile detects 1% FER, it sends a request to stop the power reduction.

2. Reverse Link Power Control Power control for the reverse link is a combined technique consisting of closed-loop and

open-loop power controls. Also, it is a fixed step size algorithm and strength-based distributed algorithm. The goal of open-loop power control is the estimation of a path loss and a loss due to shadowing between the base and the mobile station. According to this process, the mobiles transmit the initial power control signal.

However, multipath fading in a reverse and a forward DS-CDMA link is an independent process since the frequency separation of these links is 45MHz and it greatly exceeds the coherent bandwidth of the channel. Thus, closed-loop power control is used. Every cell site demodulator measures the received signal-to-noise ratio (SNR) from each mobile station. The measured SNR is compared to the desired SNR for that mobile station and a power adjustment command is sent to the mobile station. This power adjustment command is combined with the mobile station open-loop estimate to obtain the final value of the mobile station transmit power. The base station measures the signal quality (BER) and based on that determines the desired SNR for specific mobile station. In previously described power control technique, the subscribers are power controlled by the base station of their own cell. However, the interference level from subscribers in other cells varies not only according to the attenuation in the path to the subscriber's cell site, but also inversely to the attenuation from the interfering user to his own cell site, which through power control by that cell site may increase or decrease the interference to the desired cell site. It has been shown that the maximal number of subscribers in the cell is the highest when there are no subscribers in the neighboring cells. As the number of subscribers in the neighboring cells increases the maximal number of subscribers in the cell decreases.

Power control for DS-CDMA reverse link is the single most important system requirement because of the Near/ Far effect. In this case, it is necessary to have a dynamic range for control. For the forward link, no power control is required in a single cell system, since all signals are transmitted together and hence vary together. However in multiple cell systems, interference from neighboring cell sites fades independently from the given cell site and thereby degrades performance. Thus it is necessary to apply power control in this case also, to reduce intercell interference.

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CDMA System Aspects

1.2.3 According to techniques, power control

can be classified as follows: 1. Open-loop power control. 2. Closed-loop power control. 3. Outer loop power control 1. Open-loop power control

Reverse link (mobile to base station) open loop power control is accomplished by adjusting the mobile transmit power so that the received signal at the base station is constant irrespective of the mobile distance; where each mobile computes the relative path loss and compensates the loss by adjusting its transmitting power. The total received power at the cell site is the sum of all powers, which determines the system capacity.

2. Closed-loop power control

Closed-loop power control is accomplished by means of power up or power down command originating from the cell site. A single power control bit is inserted into the forward encoded data stream, the mobile responds by adjusting the power. In order to lower processing delay and to save bandwidth in the forward link, command bits for power control from the base to the mobile station are not coded and they are susceptible to errors.

3. Outer Loop PC

Signal to interference ratio is varied, to guarantee QoS (BER,..)

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CDMA System Aspects

2 RAKE Receiver

2.1 RAKE Receiver Theory and Structure

A spread-spectrum signal waveform is well matched to the multipath channel. In a multipath channel, the original transmitted signal reflects from obstacles such as buildings, and mountains, and the receiver receives several copies of the signal with different delays. If the signals arrive more than one chip apart from each other, the receiver can resolve them. Actually, from each multipath signal’s point of view, other multipath signals can be regarded as interference and they are suppressed by the processing gain. However, a further benefit is obtained if the resolved multipath signals are combined using RAKE receiver. Thus, the signal waveform of CDMA signals facilitates utilization of multipath diversity. Expressing the same phenomenon in the frequency domain means that the bandwidth of the transmitted signal is larger than the coherence bandwidth of the channel and the channel is frequency selective (i.e., only part of the signal is affected by the fading).

RAKE receiver consists of correlators, each receiving a multipath signal. After despreading by correlators, the signals are combined using, for example, maximal ratio combining. Since the received multipath signals are fading independently, diversity order and thus performance are improved.

After spreading and modulation the signal is transmitted and it passes through a multipath channel, which can be modeled by a tapped delay line (i.e., the reflected signals are delayed and attenuated in the channel).

It is necessary to measure the tapped delay line profile and to reallocate RAKE fingers whenever there is need. Small-scale changes, less than one chip, are taken care of by a code-tracking loop, which tracks the time delay of each multipath signal.

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3 Handoff Versus Handover

3.1 Handoff versus Handover The act of transferring support of a mobile from one base station to another is termed

handover or Handoff. It occurs when a call has to be handed over or off from one cell to another as the user moves between cells. In GSM system it is termed hard handoff or Handover where the connection to the current cell is broken, and then the connection to the new cell is made. This is known as a "break-before-make" handoff.

But in a CDMA system the same frequency band is shared between all the cells. Thus there is well-defined efficient bandwidth utilization. Though there is frequency reuse, the orthogonal nature of the waveforms serves to distinguish between the signals that occupy the same frequency band so it is called soft Handover or Handoff.

3.2 Soft Handover

In soft handover a mobile station is connected to more than one base station simultaneously. Soft handover is used in CDMA to reduce the interference into other cells and to improve performance through macro diversity.

3.2.1 The Importance Of Soft Handover

In power controlled CDMA systems soft handoff is preferred over hard handoff strategies. This is more pronounced when the IS-95 standard is considered wherein the transmitter power is adjusted dynamically during the operation. Here the power control and soft handoff are used as means of interference-reduction, which is the primary concern of such an advanced communication system. The previous and the new wideband channels occupy the same frequency band in order to make an efficient use of bandwidth, which makes the use of soft handoff very important. The primary aim is to maintain a continuous link with the strongest signal base station otherwise a positive power control feedback would result in system problems. Soft handoff ensures a continuous link to the base station from which the strongest signal is issued. Soft handoff requires less power, which reduces interference and increases capacity.

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CDMA System Aspects

Fig.7

3.3 Softer Handover

Is a soft handover between two sectors of a cell. As known that, in a cellular system there is spatial separation between cells using the same frequencies). This is called the frequency reuse concept.

Because of the processing gain, such spatial separation is not needed in CDMA, and frequency reuse factor of one can be used. Usually, a mobile station performs a handover when the signal strength of a neighboring cell exceeds the signal strength of the current cell with a given threshold. Since in a CDMA system the neighboring cell frequencies are the same as in the given cell, this type of approach would cause excessive interference into the neighboring cells and thus a capacity degradation. In order to avoid this interference, an instantaneous handover from the current cell to the new cell would be required when the signal strength of the new cell exceeds the signal strength of the current cell. This is not, however, feasible in practice. The handover mechanism should always allow the mobile station to connect into a cell, which it receives with the highest power (i.e., with the lowest pathloss).

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4 Multiuser Detection

The current CDMA receivers are based on the RAKE receiver principle, which considers other users’ signals as interference. However, in an optimum receiver all signals would be detected jointly or interference from other signals would be removed by subtracting them from the desired signal. This is possible because the correlation properties between signals are known (i.e., the interference is deterministic not random).

The capacity of a direct sequence CDMA system using RAKE receiver is interference limited. In practice this means that when a new user, or interferer, enters the network, other users’ service quality will go below the acceptable level. The more the network can resist interference the more users can be served. Multiple access interference that disturbs a base or mobile station is a sum of both intra- and inter-cell interference. Multiuser detection (MUD), also called joint detection and interference cancellation (IC), provides a means of reducing the effect of multiple access interference, and hence increases the system capacity.

In the first place MUD is considered to cancel only the intra-cell interference, meaning that in a practical system the capacity will be limited by the efficiency of the algorithm and the inter-cell interference. In addition to capacity improvement, MUD alleviates the near/far problem typical to DS-CDMA systems. A mobile station close to a base station may block the whole cell traffic by using too high a transmission power. If this user is detected first and subtracted from the input signal, the other users do not see the interference. Since optimal multiuser detection is very complex and in practice impossible to implement for any reasonable number of users, a number of suboptimum multiuser and interference cancellation receivers have been developed. The suboptimum receivers can be divided into two main categories: linear detectors and interference cancellation. Linear detectors apply a linear transform into the outputs of the matched filters that are trying to remove the multiple access interference using too high a transmission power. If this user is detected first and subtracted from the input signal, the other users do not see the interference. Since optimal multiuser detection is very complex and in practice impossible to implement for any reasonable number of users, a number of suboptimum multiuser and interference cancellation receivers have been developed. The suboptimum receivers can be divided into two main categories: linear detectors and interference cancellation. Linear detectors apply a linear transform into the outputs of the matched filters that are trying to remove the multiple access interference (i.e., the interference due to correlations between user codes). Examples of linear detectors are decorrelator and linear minimum mean square error (LMMSE) detectors. In interference cancellation multiple access interference is first estimated and then subtracted from the received signal. Parallel interference cancellation (PIC) and successive (serial) interference cancellation (SIC) are examples of interference cancellation.

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CDMA System Aspects

Fig.9

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Chapter 6: Appendix

1

Chapter 6 Appendix

Contents Pages

1 Abbreviations 2

Chapter 6: Appendix

2

Chapter 6

Appendix 1 Abbreviations A

AC Authentication Center ACCH Associated Control CHannel ACE Antenna Coupling Equipment ADC Analog to Digital Converter AGCH Access Grant Channel AMR Adaptive MultiRate speech AMX ATM MultipleXer AMPS Advanced Mobile Phone Services ANSI American National Standards Institute (USA) AP Application Part ARFCN Absolute Radio Frequency Channel Number ARIB Association of Radio Industries and Business (Japan) ARQ Automatic Repeat reQuest ASCI Advanced Speech Call Items ASN ATM Switching Network ATM Asynchronous Transfer Mode AUC Authentication Center B

BA BCCH Allocation BCC Base transceiver station Color Code BCCH Broadcast Control CHannel BCH Broadcast CHannel BER Bit Error Rate BPSK Binary Phase Shift Keying BS Base Station BSC Base Station Controller

Chapter 6: Appendix

3

BSIC Base transceiver Station Identity Code BSS Base Station System BSSAP Base Station System Application Part BSSMAP Base Station System Management Application Part BTS Base Transceiver Station C

CA Cell Allocation CAMEL Customized Applications for Mobile network Enhanced

Logic CATT China Academy of Telecommunication Technology

(China) CC Call Control CC Country Code CCH Control CHannel CCITT Comité Consulatif International Téléphonique et

Télégraphique CCS7 Common Channel signaling System No. 7 CCU Channel Coding Unit CDMA Code Division Multiple Access CEPT Conference Europèene des Postes et

Telecommunication CGI Cell Global Identity CI Cell Identity CN Core Network CP Call Processing CS Coding Scheme CUG Closed User Group CWTS Chinese Wireless Telecommunication Standardization

Institute D

D-AMPS Digital AMPS DCA Dynamic Channel Allocation DCS1800 Digital Cellular System in the 1800 MHz band DECT Digital Enhanced Cordless Telephone

Chapter 6: Appendix

4

DL Down Link DoA Direction of Arrival DRNS Drift RNS DRX Discontinuous Reception DS-CDMA Direct Sequence CDMA DSP Digital Signal Processor DTAP Direct Transfer Application Part DTX Discontinuous Transmission DwPTS Downlink Pilot Time Slot E

EDGE Enhanced Data Rates for GSM EFR Enhanced Full Rate speech EIR Equipment Identification Register ERC European Radio communication Committee ERMES European Radio MEssage System ESA European Space Agency ESCD Enhanced Circuit Switched Data ETSI European Telecommunications Standard Institute F

FAC Final Assembly Code FACCH Fast Associated Control CHannel FB Frequency correction Burst FCCH Frequency Correction CHannel FDD Frequency Division Duplex FDMA Frequency Division Multiple Access FEC Forward Error Correction FN Frame Number FPLMTS Future Public Land Mobile Telecommunication System FR Frame Relay FR Full Rate speech FRAMES Future RAdio wideband MultiplE access Systems

Chapter 6: Appendix

5

G

GEO GEostationary Orbital GGSN Gateway GPRS Support Node GMM Global Multimedia Mobility GMPCS Global Mobile Personal Communication Systems GMSC Gateway MSC GMSK Gaussian Minimum Shift Keying GP Guard Period GPRS General Packet Radio Service GPS Global Positioning System GSM Global System for Mobile communications H

HCR High Chip Rate HCS Hierarchical Cellular Structures HEO High Elliptic Orbit HLR Home Location Register HO(V) HandOver HR Half Rate speech HPLMN Home PLMN HSCSD High Speed Circuit Switched Data I

IAM Initial Address Message ICO Intermediate Circular Orbits ID IDentification ID IDentity IMEI International Mobile Equipment Identity IMSI International Mobile Subscriber Identity IMT-2000 International Mobile Telecommunications-2000 IN Intelligent Network Inmarsat INternational MARitime SATellite ITU International Telecommunication Union IP Internet Protocol

Chapter 6: Appendix

6

IP Intelligent Peripheral ISDN Integrated Services Digital Network ISP Internet Service Provider ISUP ISDN User Part IWE InterWorking Equipment IWF InterWorking Function IWUP InterWorking User Part J

JD Joint Detection JDC Japanese Digital Cellular

K

kbps Kilo Bits per second Kc cipher Key Ki individual subscriber authentication Key L

LA Location Area LAI Location Area Identity LAN Local Area Network LAPDm Link Access Protocol on the Dm channel LCR Low Chip Rate LEO Low Earth Orbital LES Land Earth Station LIC Line Interface Circuit LMT Local Maintenance Terminal LR Location Register M

MAP Mobile Application Part MAI Multiple Access Interference MARISAT MARItime SATellite MBS Mobile Broadband System MCC Mobile Country Code Mcps Mega Chips per Second

Chapter 6: Appendix

7

ME Mobile Equipment MExE Mobile station application Execution

Environment MM Mobility Management MMI Man Machine Interface MML Man Machine Language MNC Mobile Network Code MOC Mobile Originating Call MS Mobile Station MSC Mobile services Switching Center MSISDN Mobile Station international ISDN number MSP Multiple Subscriber Profile MSRN Mobile Station Roaming Number MSS Mobile Satellite Systems MT Mobile Termination MTP Message Transfer Part MTC Mobile Termination Call MTP Message Transfer Part MUD Multiuser Detection MUX MUltipleXer N

NB Normal Burst NCC Network Color Code (PLMN color code) NDC National Destination Code NE Network Element NMT Nordic Mobile Telephone NSS Network Switching Subsystem O

O&M Operation and Maintenance OACSU Off Air Call Set Up ODMA Opportunity Driven Multiple Access OFDMA Orthogonal Frequency Division Multiple

Access

Chapter 6: Appendix

8

OMC Operation & Maintenance Center OMC-B Operation & Maintenance Center for BSS OMC-S Operation & Maintenance center for SSS OSS Operation SubSystem OVSF Orthogonal Variable Spreading Factor codes P

PA Power Amplifier PACS Personal Access Communication System PC Power Control PCM Pulse Code Modulation PCU Packet Control Unit PDA Personal Data Assistant PDC Personal Digital Cellular (Japan) PDN Packet Data Network PHS Personal Handy System (Japan) PIN Personal Identification Number PLMN Public Land Mobile Network PMR Private Mobile Radio PP Point-to-Point PSTN Public Switched Telephone Network Q

QOS Quality Of Service QPSK Quaternary Phase Shift Keying R

RA Rate Adaptation

RACH Random Access CHannel

RAND RANDom number

REQ REQuest

RES RESponse

RF Radio Frequency

RFC Radio Frequency Channel

RFCH Radio Frequency CHannel

Chapter 6: Appendix

9

RFCN Radio Frequency Channel Number

RNC Radio Network Controller

RNS Radio Network Subsystem

RRM Radio Resource Management

RSS Radio SubSystem

RU Resource Unit

RX / Rx Receiver

S

SACCH Slow Associated Control CHannel SAP Service Access Point SAPI Service Access Point Indicator SB Synchronization Burst SCCP Signaling Connection Control Part SCE Service Creation Environment SCH Synchronization CHannel SDCCH Stand- alone Dedicated Control CHannel SF Spreading Factor SFH Slow Frequency Hopping SGSN Service GPRS Support Node SIM Subscriber Identity Module SM Security Management SMG Special Mobile Group SMP Service Management Point SMS Short Message Service SN Subscriber Number SN Switching Network SP Signaling Point SP Switching Point SS Supplementary Services SSF Service Switching Function SSP Service Switching Point STP Signaling Transfer Point

Chapter 6: Appendix

10

SW Software T

T1 Standards Committee T1 Telecommunications TA Terminal Adaptor TAC Type Approval Code TACS Total Access Communication System TB Tail Bit TCAP Transaction CApability Part TCH Traffic CHannel TD-CDMA Time Division CDMA TDD Time Division Duplex TDMA Time Division Multiple Access TE Terminal Equipment TETRA TErrestrial Trunked Radio Access THSS Time-Hopping Spread Spectrum TIA Telecommunication Industry Association TMN Telecommunication Management Network TMSI Temporary Mobile Subscriber Identity TRAU Transcoding and Rate Adaptation Unit TRX TRansceiver TS Tele Service TS TimeSlot TTA Telecommunications TechnologyAssociation (South

Korea) TTC Telecommunication Technology Committee (Japan) TX / Tx Transmitter U

UE User Equipment UL UpLink UMTS Universal Mobile Telecommunications System UP User Part USIM UMTS Subscriber Identity Module

Chapter 6: Appendix

11

UTRA UMTS Terrestrial Radio Access UTRAN UMTS Terrestrial Radio Access Network UWC-136 Universal Wireless Communication V

VAD Voice Activity Detection Sprachsteuerung VBR Variable Bit Rate VBS Voice Broadcast Service VHE Virtual Home Environment VLR Visited (visitor) Location Register VMSC Visited MSC VoIP Voice over Internet Protocol VPLMN Visited PLMN W

WAN Wide Area Network WAP Wireless Application Protocol WARC World Administrative Radio Conference W-CDMA Wideband CDMA WLL Wireless Local Loop

Chapter 7: References

1

Chapter 7 References

Contents Pages

1 References 2

2 Useful links 3


2

Chapter 7

References 1 References

• M. Mouly, M.B. Pautet, "The GSM System for Mobile Communications", Cell & Sys (1992), ISBN 2-9507190-0-7

• S. Redl, M. Weber, K. Oliphant, "An introduction to GSM", Artech House Inc.(1995), ISBN 0-89006-785-6

• Mehrotra, "GSM System Engineering", Artech House Inc. (1997), ISBN 0-89006-860-7

• G. Heine, "GPRS from A – Z", Artech House Inc. (2000), ISBN 1-58053-181-4V.K.G. Garg, K.F. Smolik, J.E. Wilkes, „Applications of CDMA in Wireless/Personal Communications“, Feher / Prentice Hall digital and wireless communications series (1997) ISBN 0-13-572157-1

• A.J. Viterbi: „CDMA: Principles of Spread Spectrum for third Generation Mobile Communication“ (1995), ISBN 0-201-63374-4

• T. Ojanperä, R. Prasad: „ Wideband CDMA for third Generation Mobile Communication“, (1998) ISBN 0-89006-735-X

• R. Prasad, W. Mohr, W. Konhäuser, „Third Generation Mobile Communications Systems, Artech House Publishers (04/2000)

• G. Calhoun, „Third Generation Wireless Communications: Post Shannon Architectures“, Artech House Publishers (07/2000)

• Authentication and Security in Mobile Phones by Greg Rose, Qualcomm Inc., Australia.

• Security in CDMA Wireless Systems by Frank Quick, Qualcomm Inc., February 1997

• Security Aspects of Mobile Wireless Networks, by Mullaguru Naidu, July 2002.

• CDMA RF System Engineering, by Samuel C. Yang • Understanding Cellular Radio, by WILLIAM WEBB • B. J. Wysocki and T. A. Wysocki, “Power Spectra of Signal Formats for DS-

SS CDMA Wireless LANs,” IEEE TENCON, pp. 329-332, 1996 • M.Y. Rhee, CDMA Cellular Mobile Communications Network Security.

Prentice Hall, 1998 • G. Allen and S. Raymond, “Encryption of Analog Signals - A Perspective,”

IEEE Journal on selected area in communications, vol. SAC-2, No. 3, pp. 423-425, 1984.

• James A. Davis, “Security Aspects in Mobile Phone Telephony: Focus on GSM,” White Paper, Jan. 2000.

• CDMA System Analysis II, by Timothy X Brown, Silvana Susi, Sukhjinder Singh University Of Colorado, Boulder


3

2 Useful links

• http://www.3gpp.org • http://www.itu.int/imt • http://www.etsi.fr • http://www.umts-forum.org • http://www.gsmworld.com • http://www.cdg.org

Chapter 8: Glossary

1

Chapter 8 Glossary

Contents Pages

1 Glossary 2

Chapter 8: Glossary

2

Chapter 8

Glossary 1 Glossary AMPS (Advanced Mobile Phone Service): Developed by AT&T’s Bell Laboratories in the1970’s and first used in the US in 1983. The AMPS Standard has been the foundation for the industry in the United States. CDMA (Code Division Multiple Access): Known in the US as IS-95, a spread spectrum approach to digital transmission. With CDMA, each conversation is digitized and then tagged with a code. The mobile phone is then instructed to decipher only a particular code to pluck the right conversation off the air. It has a 1.25Mhz spread spectrum air interface, uses the same frequency bands as AMPS and supports AMPS operation, employing spread-spectrum technology and a special coding scheme. It was adopted by the Telecommunications Industry Association (TIA) in 1993. DAMPS (Digital AMPS): The second generation of the AMPS standard. FDMA (Frequency Division Multiple Access): FDMA is the division of the frequency band allocated for wireless cellular communication into 30 KHz channels, each of which can carry a two way voice conversation. FDMA is the basic technology used in AMPS, the most widely installed cellular phone system in North America. With FDMA, each channel can be assigned to only one user at a time. EDGE (Enhanced Data rate for GSM Evolution): The next generation of data heading towards third generation and personal multimedia environments. It builds on GPRS and is a technique to increase the maximum data capacity of GSM radio channels. It will allow GSM operators to use existing GSM radio bands to offer wireless multimedia IP-based services and applications at theoretical maximum speeds of 384 kbps with a bit-rate of 48 kbps per timeslot and up to 69.2 kbps per timeslot in good radio conditions. GPRS (General Packet Radio Service): A GSM data transmission technique that does not set up a continuous channel from a portable terminal for the transmission and reception of data, but transmits and receives data in packets, with users only paying for the volume of data sent and received. GPS (Global Positioning System): A satellite navigation system, consisting of 24 geosynchronous satellites. Used in personal tracking, navigation and automatic vehicle location technologies.

Chapter 8: Glossary

3

GSM (Global System for Mobile communications): A digital cellular or PCS network used throughout the world. Developed by ETSI in Europe. NAMPS (Narrowband AMPS): NAMPS combines cellular voice processing with digital signaling to increase the capacity and functionality of AMPS systems. PCS (Personal Communications Services): A two-way, 1900 MHz digital voice, messaging and data service designed as the second generation of cellular. TDMA (Time Division Multiple Access): A method of digital wireless communications transmission allowing a large number of users to access (in sequence) a single radio frequency channel without interference by allocating unique time slots to each user within each channel UMTS (Universal Mobile Telecommunications System): Europe's approach to standardization for third-generation cellular systems, it will be based on W-CDMA. UMTS will offer a wide range of voice, data and multimedia services with data rates from 114 Kbps to 2 Mbps, depending on whether the user is stationary or in motion. W-CDMA (Wideband Code Division Multiple Access): The European third generation wireless standard. The wideband represents the increase of the frequency band to 5 MHz, in comparison to the 1.25 MHz band used in conventional CDMA (also known as cdmaOne). AuC (Authentication Center): The component of a GSM network that authenticates subscriber and mobile equipment identities. Baseband: The signaling of a digital or analog signal at its original frequencies, i.e. not changed by modulation. BSC (Base Station Controller): The component of a GSM system that controls a group of base stations and acts as a node for connecting base stations to the mobile switching center. BSS (Base Station Subsystem): The combination of BSC’s and base stations that together provide the radio functionality in a mobile system. Cell: The basic geographic unit of a cellular system. Also, the basis for the generic industry term "cellular". The mobile network’s geographic area is divided into smaller “cells”, each of which is equipped with a low-powered radio transmitter/receiver. The cells can vary in size depending upon terrain and capacity demands. By controlling the transmission power, the radio frequencies assigned to one cell can be limited to the boundaries of that cell.

Chapter 8: Glossary

4

Cell Site: The central radio transmitter/receiver that maintains communications with mobile phones within a give range. Also called a Base Station. Diversity: The use of multiple antennas to receive or transmit the same signal, so that if one of the antennas picks up a weak signal, another antenna should have a strong signal. Downlink: The transmission of radio signals from the Base Station to the mobile handset. EIR (Equipment Identity Register): The component of a GSM system that retains information about the identity of equipment such mobile phones. Assists network operator in discovering stolen mobile phones and blocking them from using the network. Fading: A reduction in signal strength in a radio signal. Fading is usually caused by reflected waves from the transmitter having different phases from the main signal path. GMSC (Gateway Mobile Switching Center): The component of a GSM network, which provides a point of connection between the GSM network and the PSTN. Handoff: The process of transferring a mobile phone conversation from one cell site to another as a user crosses cell areas during the conversation. HLR (Home Location Register): The component of a GSM network responsible for maintaining the location of a mobile. IMEI (International Mobile Equipment Identity): The unique serial number given to each phone, to help in tracking stolen mobile phones. IMSI (International Mobile Subscriber Identity): A unique number used in GSM systems to identify individual subscribers. MAHO (Mobile Assisted Handoff): Similar to a basic handoff, except that the mobile also helps in finding a suitable base station to handoff into by providing the network with measurements indicating which base station provides the largest signal strength.

Chapter 8: Glossary

5

Modulation: Information on a carrier signal modulated by varying one or more of the signal's basic characteristics - frequency, amplitude and phase. Different modulation carries the information as the change from the immediately preceding state rather than the absolute state. MS (Mobile Station): Another name for a cellular mobile phone. MSC (Mobile Switching Center): The switch in a GSM network, which connects calls from the GMSC to the particular base station in which the mobile phone is currently located. The MSC also manages call handovers. MTSO (Mobile Telephone Switching Office): The central computer that connects a wireless phone call to the public telephone network. The MTSO controls the entire system’s operations, including monitoring calls, billing and handoffs. POTS (Plain Old Telephone Service): Standard household phone service. PSTN (Public Switched Telephone Network): The worldwide telephone network which allows people to call anywhere in the world. The PSTN mainly consists of copper cables and switches. Roaming: Roaming allows a user to operate their mobile phone in another countries network. The user’s network makes agreements with other networks worldwide to allow this to happen. Smart antenna: An antenna system with technology that enables it to focus its beam on a desired signal to reduce interference. A wireless network would employ smart antennas at its base stations in an effort to reduce the number of dropped calls, improve call quality and improve channel capacity. Soft handoff: Procedure in which two base stations, one in the cell site where the phone is located and the other in the cell site to which the conversation is being passed, both hold onto the call until the handoff is completed. The first cell site does not cut off the conversation until it receives information that the second is maintaining the call. This reduces the probability of the call being blocked. Uplink: The transmission of radio signals from the mobile handset to the Base Station.

Chapter 8: Glossary

6

VLR (Visitor Location register): The component of a GSM network which keeps track of a mobile phone’s position to the nearest location area. Walsh codes: A family of orthogonal codes often preferred for CDMA transmission. WLL Wireless Local Loop: The use of radio to replace copper wiring as a means of connecting the home to the PSTN.

Part 7

GPRS Introduction

1 Introduction and Overview Pages (1-15)

2 GPRS ‐ General Packet Radio Services

Pages (1-35)

3 GPRS Radio Interface Pages (1-18)

4 Procedures Pages (1-10)

5 Abbreviations Pages (1-10)

Sub ‐ Sections GPRS Introduction


Introduction and Overview Siemens

Contents

1 Mobile Radio Evolution 23

1.1 Trend: from Speech to Data Transmission 34

1.2 The 3rd

Mobile Radio Generation (3G) 46

2 GSM – Current Situation, Services & Applications 69

2.1 GSM – Global System for Mobile Communication 170

2.2 GSM – Implementation in an evolutionary Concept 192

3 GSM – Phase2+ 115

3.1 GSM Phase 2+ Solutions for Meeting Current and Future Mobile

Requirements 126

3.2 Data Transmission in GSM Phase2+ 138

4 Exercise 23

5 Solution 27

Introduction and Overview

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Introduction and Overview Siemen

1 Mobile Radio Evolution

0,01

0,1

1

10

100

1000

Subscriber[M.]

1982

1984

1986

1988

1990

1992

1994

1996

1998

2000

2002

Year

Germany

World

Subscriber trends:

1982 - 2002

Fig. 1 Increase in the number of subscribers due to introduction of first and second generation of mobile communication

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1.1 Trend: from Speech to Data Transmission

1G offered mainly speech transmission based on analog transmission modes.

Due to the digital transmission mode it uses, 2G offers not only pure speech trans-

mission but also a number of supplementary services and low rate data transmission.

However, mobile radio systems of 2G are suited optimally to the needs of speech

transmission, primarily; the share of data transmitted via the radio interface will not

exceed 2% even towards the end of the 90ties.

Nevertheless, growth rates in the area of data transmission are much higher than in

the area of speech transmission due to the fact that the need for mobile data trans-

mission is becoming acute in the mobile working world of tomorrow (work outside the

office, teleworking).

Forecasts predict the following figures: in the year 2001, 10% of the total traffic vol-

ume will be allotted to data transport via the radio interface, in 2005 this will already

rise to 30%, and just two years later, in 2007, data transmission will make up 50% of

the total traffic volume and will thus range equal to speech transmission.

Note that there is an underlying rapid increase in the total amount of traffic.


0

20

40

60

80

100

tra

ffic

[%

]

1996 2001 2005 2007

year

speech

data

2 G Trends:

Speech → Data transmission

1 G:

speech transmission only

2 G:

• speech transmission

• supplementary services

• data transmission

Fig. 2 Trend in the traffic to be transported by future mobile communications systems

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1.2 The 3rd

Mobile Radio Generation (3G)

Currently there are numerous different standards for 1G and 2G mobile radio sys-

tems, each of which has specific characteristics, advantages and disadvantages, ap-

plications and users. Most of the standards are used merely on a national or regional

scale and are not compatible with each other. They cannot meet the requirements

which will be indispensable for future mobile radio systems, such as improved

speech quality, worldwide availability and particularly a fast transfer of large amounts

of data.

3G currently being standardized under the heading IMT-2000 (International Mobile

Telecommunication) designates a global system of compatible standards which in-

deed is able to meet the high demands placed on future mobile radio systems (see

above). The general aim is to enable “communication with anyone, anywhere, any-

time”.

Beside speech transmission, high data rate services and multimedia applications are

to be provided to the customer across all operator-dependent, national and geo-

graphical borders at any place and any time.

The body in charge of IMT-2000 specification is the International Telecommunication

Union ITU. Thus, IMT-2000 shall become the worldwide “guideline” to which all stan-

dards of the 3G orient themselves. In the framework of IMT-2000 guidelines ETSI is

about to standardize a follow-up GSM standard based on the experiences with and

the success of GSM: the standard is known as UMTS (Universal Mobile Telecommu-

nication Standard).

UMTS is a downward compatible to GSM; as such it shall provide worldwide multi-

media access at any point in time and cover all current mobile radio applications.

Data rates of 8 kbit/s up to a maximum of 2Mbit/s shall be supported.

Apart from UMTS the regional standardization authorities draw up further 3G based

on the IMT-2000 guidelines.


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Third Generation (3G)

2G

Digital Paging

e.g. ERMES

Digital CT

e.g. DECT, PACS, PHS

Wireless

Local Loop

WLL

digital

PMR

e.g. TETRA

digital

cellular systems

e.g GSM, D-AMPS,

IS-95, PDC

digital MSS

e.g. IRIDIUM

1G

analog

Cordless Telephone CT

e.g. CT1, 1+

Paging

Wireless booth

analoge

Private Mobile Radio

PMR

analog

cellular systems


analog MSS

e.g. INMARSAT

IMT-2000:

UMTS, MC-CDMA,

TD-SCDMA,...

3G

Multiple incompatible standards

for different

one standard (family)

for all

• applications

• countries / regions

• applications

• countries / regions

• compatibility within 3G

• downward compatibility to

2G (e.g. UMTS → GSM)

• resource efficiency

• high data rates

• Multimedia

Fig. 3 Intention of third generation as a common global standard for different applications, regions, and service areas

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2 GSM – Current Situation, Services & Appli-

cations

GSM - current situation,

services & applications

Mobile Radio

Evolution

Fig. 4

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2.1 GSM – Global System for Mobile Communication

The GSM standard was planned in the early 80ties and agreed upon in 1990 as first

2G standard. The GSM standard has been specified by ETSI as a consistent open

standard for cellular mobile radio systems. It consists of more than 100 recommenda-

tions, categorized into 12 series.

Within Rel. 99 the GSM standard is know specified by GERAN, a group of 3gpp. The

new series are now be found in series number 40-50.

Commercial operation of GSM networks started in 1992. Originally the systems were

planned for Europe only, but in the middle of 1999 there were already 340 GSM net-

works worldwide in 135 countries/regions. In 2001 there are about 45 million sub-

scriber worldwide

Beside the originally planned GSM standard in the 900 MHz range (GSM900 / E-

GSM) further GSM adaptations were specified during the 90ties in the 1800 and 1900

MHz range (GSM1800 & GSM1900) as well as one adaptation for railway communi-

cation (GSM-R).

GSM900 / E-GSM

In 1990 the first GSM standard, known as GSM900 with 2x 25 MHz developed. An

extension of this, the E-GSM (Extended GSM), provides a further 20 MHz, i.e. a total

of 2 x 35 MHz for GSM, in the event that national authorizations to operate other sys-

tems expire.

GSM1800 (DCS1800)

In 1991 the DCS1800 (Digital Cellular System) standard, a GSM adaptation, was

agreed upon as result of a British initiative in view of the opening-up of a mass-

market; in 1997 this standard was renamed GSM1800. For GSM1800 2 x 75 MHz is

available in the 1800 MHz area.

GSM1900 (PCS1900)

Since 1995 PCS1900 (Public Cellular System), renamed GSM1900 in 1997 repre-

sents the GSM adaptation for the American market. 2 x 60 MHz are available for

GSM1900 and other standards (D-AMPS, IS-95,..).

GSM-R

GSM-R (Railway) was specified as GSM Adaption for mobile radio communication. In

1995 ETSI decided to reserve 2 x 4 MHz in 900 MHz range for GSM-R. First GSM-R

systems are in operation since 1998


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876 880

890

GSM

900

915 921 925

935

960 1710 1785 1805 1850

1880

1910 1930 1990[MHz] [MHz]

GSM

900

E-GSM E-GSM

GSM

1800

GSM

1800

GSM

1900

GSM-RGSM-Adaptations

GSM

1900

Frequency Range

[MHZ]

Useable HF

channels

Application Area

GSM900

E-GSM

890 - 915 / 935 - 960

880 - 915 / 925 - 960

124

174

Worldwide except

US

GSM18001710 - 1785 / 1805 - 1880 374 Worldwide except

US

GSM19001850 - 1910 /1930 - 1990 Shares HF-channels

with other standards

US

GSM-R876 - 880 / 921 - 925 19 European

railroads

Fig. 5 Adaptations of GSM in frequency due to trend to mobile communication

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2.2 GSM – Implementation in an evolutionary Con-

cept

Originally, the GSM standard was intended as a completed, non-modifiable standard

to be used until the standardization of a 3G follow-up system. However, in 1988 al-

ready it became obvious that it was not possible to standardize all the technical de-

tails and service offers requested within the time frame set. This resulted in the im-

portant decision to leave the GSM standard incomplete and develop and work on it

permanently instead. The evolutionary GSM concept thus provides enough scope for

technical evolutions and can be quickly adapted to the rapidly changing market con-

ditions. GSM developed in various phases.

GSM Phase1

Phase 1 (agreed upon in 1990/91) includes all central prerequisites for mobile, digital

transmission of information. Speech transfer plays an important role. Data transmis-

sion was also defined with transmission rates of 0.3 to 9.6 kbit/s. GSM phase 1 in-

cludes only a few supplementary services.

GSM Phase2

Research on GSM phase 2 was concluded in 1995. Mainly supplementary services

comparable to ISDN were specified, but also technical improvements such as half

rate speech were considered. Of central importance was the agreement on down-

ward compatibility, meaning that all networks and terminal equipment of phase 2

were compatible to the networks and terminal equipment of phase 1.

GSM Phase2+

Phase2+ marks a “smooth” transition as opposed to phase2. The standard is not en-

tirely re-worked. Since 1996 annual releases take place and current themes relate to

new supplementary services relevant mainly for special groups of users, as well as to

connection and call control issues, IN applications and data services with high trans-

mission rates.


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Phase 1

Phase 2

Phase 1

Phase 2+

Phase 2

Phase 1

Capabilities

year1991 1995 1997

Speech FR,

standard services

Data: max. 9,6 kbit/s

multiple

Supplementary Services (SS)

comparable to ISDN;

decision downward compatibility

Annual Releases !

• new SS

• IN-applications

• new Bearer Services

(high data rates)

GSM: evolutionary concept

Downward compatibility

Early concept:

• closed standard

• life time: until successor standardisation (3G)

Fig. 6 Evolutionary concept of the GSM standard

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3 GSM – Phase2+

GSM - Phase 2+

Mobile Radio

Evolution

Fig. 7

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3.1 GSM Phase 2+ Solutions for Meeting Current and

Future Mobile Requirements

GSM phase 2+ develops solutions for numerous demands placed on future mobile

radio systems. Improved speech quality is realized through introduction of a new

speech code (Enhanced Full Rate Speech), worldwide availability is achieved

through multi-mode terminal equipment (satellite roaming). New features (e.g. Ad-

vanced Speech Call Items ASCI for GSM-R) and IN-integration (e.g. Customized Ap-

plications for Mobile network Enhanced Logic, CAMEL) supplement the portfolio of

applications. For the implementation of „mobile Computing“ / Internet access, bearer

services such as High Speed Circuit Switched Data HSCSD, General Packet Radio

Service GPRS are standardized allowing for the adaptation of transmission rates to

those of ISDN. Also, transmission rates can be increased up to 100 kbit/s and more.

User-friendly equipment and comfortable connection options to the mobile equipment

(Blue Tooth) round off the offer and make it suited to meet future demands.

The importance of phase 2+ lies, however, also in the creation of a platform on which

the GSM follow-up standard UMTS can be based. Numerous features of phase 2+

(especially GPRS and CAMEL) are “guidelines” for UMTS and shall prepare UMTS

features. Thus, upward compatibility of GSM with the 3rd

mobile generation is en-

sured and also downward compatibility of UMTS with GSM. To successfully introduce

UMTS this compatibility with GSM as “quasi-world standard” is indispensable, as is

the usage of a common GSM (Phase 2+)/UMTS infrastructure.


EFR

Enhanced

Full Rate

CAMEL

CustomizedApplication

for Mobile network

Enhanced Logic

ASCI

Advanced Speech

Call Items

Multi-

Band / Mode

Satellite

Roaming

GSM

Phase2+

GPRS

General Packet

Radio Service

HSCSD

High Speed Circuit

Switched Data

Multiple further

features

• GSM solutions for

demands to

mobile radio:

∗ enhanced speech quality

∗ user friendly equipment

∗ world-wide connectivity /

“home PLMN” service

∗ specific services

∗ fast transfer of large

data volumes

• platform for UMTS:

compatibility GSM ⇔ UMTS

common infrastructure

GSM

Phase 2+

Solutions

EDGE

Enhanced Data Rates

for the GSM evolution

Fig. 8 Solutions for new demands and market trends offered by GSM phase 2+

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3.2 Data Transmission in GSM Phase2+

To increase the data transmission rates, in GSM phase 2+ new bearer services with

rates comparable to or higher as ISDN are developed:

HSCSD (High Speed Circuit Switched Data)

GPRS (General Packet Radio Services)

EDGE (Enhanced Data Rates for the GSM Evolution)

High Speed Circuit Switched Data HSCSD

HSCSD (Rec. 02.34) is a circuit switched data service (only point-to-point) for ap-

plications with higher bandwidth demands and continuous data stream, e.g. motion

pictures or video telephony. The higher bandwidth is achieved by combining 1-8

physical channels for one subscriber. Additionally, the data transmission codec was

changed such that a maximum of 14.4 kbit/s instead of 9.6 kbit/s can be transmitted

per physical channel. In this way, HSCSD theoretically enables transmission rates up

to 115.2 kbit/s. In order to implement HSCSD merely the GSM-PLMN software must

be modified. More problematic is the high volume of resources needed.

General Packet Radio Services GPRS

With GPRS it is possible to combine 1-8 physical channel for one user, just as with

HSCSD. Various new coding schemes with transmission rates of up to 21.4 kbit/s per

physical channel enable theoretical transmission rates up to 171.2 kbit/s. Opposite to

HSCSD, GPRS is a packet-switched bearer service, meaning that the same physical

channel can be used for different subscribers. GPRS is resource efficient for applica-

tions with a short-term need for high data rates (e.g. surfing the Internet, E-mail, ...).

GPRS also enables point-to-multipoint transmission and volume dependent charging.

Extensions of the GSM network and protocol architecture are necessary for GPRS

implementation.


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Comparison HSCSD / GPRS

TDMA-frame ⇔ 8 Time Slots

Time Slot

HSCSD

up to

14.4 kbit/s

GPRS

up to

21.4 kbit/s

HSCSD GPRS

• circuit oriented

⇒ real time applications

(e.g. video telephone)

• bundling of channels

(up to 8 time slots)

• new coding scheme

(9.6 kbit/s → 14.4 kbit/s)

• point-to-point

• small HW modifications

• packet oriented

⇒ data applications

(e.g. internet surfing)

• bundling of channels

(up to 8 time slots)

• 4 new coding schemes

(9.6 kbit/s → 9.05 ... 21.4 kbit/s)

• point-to-multipoint

• new network elements/protocols

Fig. 9 Comparison of HSCSD and GPRS

Enhanced Data rates for the GSM Evolution EDGE

EDGE (Release`99) is able to realize up to 69.2 kbit/s per physical channel though

the change of the GSM modulation procedure (8PSK instead of GMSK). Theoreti-

cally, transmission rates of up to 553.6 kbit/s (meeting 3G requirements) would be

possible by combining up to 8 channels. A combination of GPRS and EDGE could of-

fer optimum usage of Inter- and Intranet, ensuring highest economy in frequency re-

source utilization at the same time.

The change of the modulation method will require hardware changes in the BSS (the

BTS have to be upgraded) and the MS. The mobile equipment has to be small and

cheap but on the other hand high quality linear amplifiers are needed for 8PSK. The

solution to this problem could be that in the introduction phase EDGE is only used in

the downlink.

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EDGE

(Enhanced Data Rates for GSM Evolution)

EDGE:

• uses a new modulation method:

replaces GMSK by 8PSK

⇒ three bit of information can be transported

by one symbol of modulation (instead of one bit)

⇒ BTS has to be upgraded

⇒ hardware modifications are necessary

• will possibly used only DL in the introduction phase

⇒ cheap mobile phones

⇒ asymmetric data rates in UL and DL

Fig. 10 EDGE replaces GMSK modulation method to enhance data rates

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GPRS - General Packet Radio Services Sidemen's

Contents

1 GPRS Objectives and Advantages 23

1.1 GPRS Objectives and Advantages 34

1.2 Standardization 56

2 Basic Principles 79

2.1 Management of Radio Resources/ Coding Schemes 180

2.2 GPRS Subscriber Profile 102

2.3 Quality of Service (QoS) Profiles 124

3 GPRS-Architecture 1721

3.1 GPRS Architecture 1822

3.2 GSM Phase 2+, Interfaces 1924

3.3 New Network Elements for GPRS 216

4 Logical Functions 2735

4.1 Logical Functions in the GPRS Network 2836

4.2 Allocation of Logical Functions 3544

5 Exercises 47

6 Solutions 55

GPRS - General Packet Radio Services

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1 GPRS Objectives and Advantages

Objectives & Standardization

GPRS

General Packet Radio Services

Fig. 1

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Sidemen's GPRS - General Packet Radio Services

1.1 GPRS Objectives and Advantages

The transmission of data is becoming increasingly important in the field of telecom-

munication. In the fixed network, the transmission of extensive data files and E-mail

and contacts to the Intra- and Internet is by far in excess of language transmission.

The need for mobile data transport is increasing at a similarly impressive rate, yet the

presently available mobile communication systems, even GSM, still present a num-

ber of shortcomings.

Disadvantages for the user in GSM Phase 1/2:

In GSM (phase 1/2), the data rate is limited to a peak value of 9.6 kbit/s

Links to the data networks need to be routed via PSTN/ISDN (Additional charging of

the user for using a transit network)

The user is billed for the connection duration instead of being billed for his/her actual

use of the network (data volume)

The set-up of a connection takes more time (ca. 20s if a modem is used)

The length of SMS is limited (160 alphanumerical characters)

Disadvantages for the provider in GMS Phase 1/2:

Inefficient resource management & the number of users is limited.

HSCSD (High Speed Circuit Switched Data)

In principle, transmission rates of up to 115.2 kbit/s can be achieved with HSCSD.

Combining 4 timeslots, the ISDN transmission rate can be matched. One problem of

HSCSD, however, is the circuit switched data transmission. Efficient resource man-

agement is impossible. Additional costs arise for the user. For this reason HSCSD is

essentially suited for applications involving high but constant transmission rates

(videotelephony).

GPRS (GENERAL PACKET RADIO SERVICES)

GPRS is, on the one hand, intended to provide the possibility of transmitting large

volumes of data in a very short time. On the other hand it is meant to ensure effective

management of available resources, which will increase the number of users and re-

duce the costs arising for the individual user (volume-oriented fees).

Another positive consequence of the introduction of GPRS is its direct access to the

Intra- and Internet and the possibility to use point-to-point and point-to-multipoint ser-

vices side by side. An important aspect is that GSM networks are prepared for the in-

troduction of UMTS.

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GPRS Objectives

& Advantages

PSTN

Modem

ISDN

Service provider

access point

BSS

SSS

IP

Modem

SMSC

SMS

PDN´s

Intranet

Internet

PSPDN

BS-udi

BS-

3.1 kHz

audio

GPRS: • high data rates • reducing costs (volume dependent charging)

• resource efficient • Point-to-Multipoint services for PMR market

• no SMS restrictions • direct IP/X.25 connection

• prerequisite for UMTS introduction ⇒ future proof solution

Fig. 2 Limitations of the network architecture

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1.2 Standardization

The introduction of GPRS into the GSM Recommendations is carried out in two

phases.

Phase 1 of GPRS introduction was completed by ETSI in the Annual Release 1997

(03/98) and includes all central GPRS functions.

Phase 1 supports:

Point-to-point transfer of user data

TCP/IP and X.25 bearer services

GPRS identities

GPRS safety (a new ciphering algorithm specially designed for packet data)

Support of volume-oriented billing

In Phase 2, further extensions are planned for all requirements to be met by GPRS:

Support of point-to multipoint (PTM) services

Support of special point-to-point and point-to-multipoint services for applications such

as traffic telematics and GSM-R (PTM-Group Call: PTM-Multicast)

Support of further additional services

Support of additional interworking functions (e.g. ISDN)

Phase 2 will be completed in 1998 or 1999.

GPRS Phase 1 includes the introduction of a number of new recommendations;

some of the existing recommendations have been modified to cover other GPRS

functions, too.

The following recommendations are of central importance:

Rec. 02.60 General GPRS Overview

Rec. 03.60 GPRS System and architecture description

Rec. 03.64 Radio architecture description

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GPRS-Standardisation

ETSI/GERAN

GPRS Standardisation in 2 Phases

Rec. 02.60

General GPRS Overview

Rec. 03.60

GPRS system &

architecture description

Rec. 03.64

Radio architecture descriptionVery important:

• PtP Data transmission

• TCP/IP & X.25 Bearer Services

• GPRS Identities

• GPRS Security (Ciphering)

• SMS via GPRS

• volume dependent charging

Phase 1:

(Rel.`97)

• PtM data transmission

• Broadcast & Group Call →

traffic telematic, GSM-R

• further interworking

functionality

• further services

Phase 2:

(Rel.`98/99)

Fig. 3 Standardization of GPRS in phases

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2 Basic Principles

Basics

GPRS


Fig. 4

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2.1 Management of Radio Resources/ Coding

Schemes

In a GPRS-supported cell, one or several physical channels can be allocated to

GPRS transmission. These physical channels (Packet Data Channels PDCHs) are

shared by GPRS mobile stations and are taken from the common/shared pool of all

available physical channels of the cell.

Distribution of the physical channels for various logical packet data channels is based

on blocks of 4 normal bursts each. Uplink (UL) and downlink (DL) for GPRS packet

data are assigned separately (consideration of asymmetrical traffic peaks). Allocation

of circuit switched services and GPRS is achieved dynamically, depending on what

capacities are required („capacity on demand“). PDCHs need not be allocated per-

manently; however, it is possible for the operator to permanently or temporarily re-

serve a number of physical channels for GPRS traffic.

New GPRS coding schemes (CS) - CS1 - CS4 - have been defined for the transmis-

sion of packet data traffic channel PDTCH (Rec. 03.64). Coding schemes can be as-

signed as a function of the quality of the radio interface. Normally, groups of 4 burst

blocks each are coded together.

CS-1 makes use of the same coding scheme as has been specified for SDCCH in

GSM Rec. 05.03. It consists of a half rate convolutional code for forward error correc-

tion FEC. CS-1 corresponds to a data rate of 9.05 kbit/s.

CS-4 has no redundancy in transmission (no FEC) and corresponds to a data rate of

21.4 kbit/s.

CS-2 and CS-3 represent punctured versions of the same half rate convolutional

code as CS-1.

CS-2 corresponds to a rate of 13.4 kbit/s, while CS-3 corresponds to a data rate of

15.6 kbit/s.

In principle, 1 to 8 time slots TS of a TDMA frame can be combined dynamically for a

user for the transmission of GPRS packet data. Theoretically it is thus possible to

achieve peak performances of up to 171.2 kbit/s (8x21.4 kbit/s) with GPRS.

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9,05 kbit/s

13,4 kbit/s

15,6 kbit/s

21,4 kbit/s

CS-1

CS-2

CS-3

CS-4

Coding

Schemes

different

redundancy (FEC) →

“Um transmission quality”

Radio Resource Management / Coding Schemes

CS & PS (GPRS):

“capacity on demand”

Physical channel of one cell

GPRS-MSs:

sharing physical channel

GPRS-MSs:

combining 1-8 TS

Up to

171,2 kbit/s

(theoretically)

1 - 8

channel

GPRS-MSs:

asymmetric UL / DL

Fig. 5 Management of radio resources: coding schemes, FEC, and redundancy

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2.2 GPRS Subscriber Profile

The GPRS Subscriber Profile is the description of the services a subscriber is al-

lowed to use. Essentially, it contains the description of the packet data protocol used.

A subscriber may also use different packet data protocols (PDPs), or one PDP with

different addresses. The following parameters are available for each PDP:

The packet network address is necessary to identify the subscriber in the public

data net. Either dynamically assigned (temporary) addresses or (in the future) static

addresses are used in case of IP. The problem of the dynamic addresses will be

overcome with the change from Ipv4 to IPv6. In GPRS is two layer 2 protocols are al-

lowed, X.25 or IP.

The quality of service QoS: QoS describes various parameters. The subscriber pro-

file defines the highest values of the QoS parameters that can be used by the sub-

scriber.

The screening profile: This profile depends on the PDP used and on the capacity of

the GPRS nodes. It serves to restrict acceptance during transmission/reception of

packet data. For example, a subscriber can be restricted with respect to his possible

location, or with respect to certain specific applications.

The GGSN address: The GGSN address indicates which GGSN is used by the sub-

scriber. In this way the point of access to external packet data networks PDN is de-

fined. The internal routing of the data is done by IP protocol; the GSNs will have IP

addresses. A DNS function is needed to find the destination of the data packets (ad-

dress translating: e.g. www.gsn-xxx.com → 129.64.39.123)

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GPRS Subscriber Profile

Subscription profile

used Packet Data Protocols PDP

possible: 1 Subscriber - different PDPs / 1 PDP with different addresses

PDP

Parameter

Packet

network address

static/dynamic

IP address

QoS

Quality of Service

highest QoS-

parameter values in

Subscriber Profile

Screening

Profile

limits receiving / emission

of data packets

GGSN address

Access to external PDN

Fig. 6 Part of the GPRS subscriber profile are the PDPs and their parameters

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2.3 Quality of Service (QoS) Profiles

The different applications that will make use of packet-oriented data transmission via

GPRS require different qualities of transmission. GPRS can meet these different re-

quirements because it can vary the quality of service (QoS) over a wide range of at-

tributes. The quality of service profile (Rec. 02.60, 03.60) permits selection of the fol-

lowing attributes:

Precedence class

Delay class

Reliability class

Peak throughput class

Mean throughput class.

By combining the variation possibilities of the individual attributes a large number of

QoS profiles can be achieved. Only a limited proportion of the possible QoS profiles

need PLMN-specific support.

Quality of Service QoS - Profile

Different requirements for different applications ⇒

multiple GPRS QoS profiles

precedence class

delay class

reliability class

Peak

throughput

class

mean throughput

class

PLMN must support only

limited QoS service profile

Fig. 7 Quality of service parameters

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Precedence Class

Three different classes have been defined to allow assessment of the importance of

the data packets, in case of limited resources or overload:

1. High precedence

2. Normal precedence

3. Low precedence

Delay Class

GSM Rec.02.60 defines 4 delay classes (1 to 4). However, a PLMN only needs to re-

alize part of these. The minimum requirement is the support of the so-called „best ef-

fort delay class“ (Class 4). Delay requirements (maximum delay) concern the delay of

transported data through the entire GPRS network (the first two columns refer to data

packets 128 bytes in length, while the last two columns apply to packets 1024 bytes

in length).

Delay Class mean transfer

delay (sec)

95% delay

(sec)

mean transfer

delay (sec)

95% delay

(sec)

1 < 0,5 < 1,5 < 2 < 7

2 < 5 < 25 < 15 < 75

3 < 50 < 250 < 75 < 375

4 (Best Effort) unspecified unspecified unspecified unspecified

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Precedence Class

1: high priority

2: normal priority

3: low priority

Delay Class mean transfer

delay (sec)

95% delay

(sec)

mean transfer

delay (sec)

95% delay

(sec)

1 < 0,5 < 1,5 < 2 < 7

2 < 5 < 25 < 15 < 75

3 < 50 < 250 < 75 < 375

4 (Best Effort) unspecified unspecified unspecified unspecified

Delay Class

SDU size: 128 Byte 1024 Byte

minimum

requirements

Fig. 8 QoS is an assumption of several parameters, which are defined in the recommendations

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Reliability Class

Transmission reliability is defined with respect to the probability of data loss, data de-

livery beyond/outside the sequence, twofold data delivery, and data falsification

(probabilities 10-2

to 10-9

):. 5 reliability classes (1 to 5) have been defined, 1 guaran-

teeing the highest and 5 the lowest degree of reliability. Highest reliability (Class 1) is

required for error-sensitive, non-real-time applications, which have no possibility of

compensating for data loss; lowest reliability (Class 5) is needed for real-time applica-

tions which can get over data loss.

Peak Throughput Class

The peak throughput class defines the maximum data rate to be expected (in

bytes/s). However, there is no guarantee that this data rate/throughput can be

achieved over a certain period of time. This depends on the capacity of the MS and

the availability of radio resources. 9 throughput classes have been defined, ranging

from Class 1 with 1000 bytes/s (8 kbit/s) to 256,000 bytes (2048 kbit/s). The maxi-

mum data rate doubles from one class to the next.

Mean Throughput Class

The mean throughput class represents the mean data rate /throughput to be ex-

pected for data transport via the GPRS network during an activated link. A total of 19

classes have been defined. Class 1 is „best effort“ and means that the data rate for

the MS is made available on the basis of demand and availability of resources.

Class 2 stands for 100 bytes/h (0.22 bit/s), class 3 for 200 bytes/h, class 4 for 500

bytes/h and class 5 for 1000 bytes/h, etc. till Class 19 which stands for 50000000

bytes/h (111 kbit/s).

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Reliability Class

1 - 5 (lowest):

• data loss probability

• out of sequence probability

• duplicate probability

• corrupt data probability

probabilities 10-9

- 10-2

peak throughput Class

1 - 9: > 8 kbit /s - >2048 kbit /s

maximum data rate

no guarantee for this data rates

over a longer period of time

mean throughput Class

medium, guaranteed data rate; Class 1-19

1: best effort

100 Byte/h (0,22 bit/s) / 200 / 500 / 1000 / ... /

50 Mio. Byte/h (111 kbit/s)

Fig. 9 QoS is an assumption of several parameters, which are defined in the recommendations

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3 GPRS-Architecture

Architecture

GPRS


Fig. 10

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3.1 GPRS Architecture

For introducing GPRS, the logical GSM architecture is extended by two functional

units:

The Serving GPRS Support Node SGSN is on the same hierarchic level as MSC

and has functions comparable to those of a Visited MSC (VMSC).

The Gateway GPRS Support Node GGSN has functions comparable with those of a

Gateway MSC (GMSC) and offers interworking functions for establishing contact be-

tween the GSM/GPRS-PLMN and external packet data networks PDN

A GPRS Support Node GSN includes the central functions required to support the

GPRS. One PLMN can contain one or more GSNs.

In addition to GSN, extensions of functions in other GSM functional units are neces-

sary:

In the BSS a Packet Control Unit PCU ensures the reception/adaptation of packet

data from SGSN into BSS and vice versa.

GPRS subscriber data are added to the HLR. On the following pages of this script

this extension will be termed GPRS Register GR.


in BTS

for channel coding

Mobile

DTE

SGSN

Serving GPRS

Support Node

PSTN

Internet

Intranet

X.25

GGSN

Gateway GPRS

Support Node

VMSC /

VLR

GMSC

HLR

New network entities:

• SGSN

(access to BSS)

• GGSN

(access to PDN)

GPRS - Architecture

ISDN

PCU

BSS

GPRS subscription data

(GPRS Register GR)


for

protocol conversion &

radio resource

management

Fig. 11 Outline of the GPRS architecture

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3.2 GSM Phase 2+, Interfaces

Integration of functions GGSN and SGSN (which are necessary for GPRS) into a

GSM-PLMN makes it necessary to provide names for a series of new interfaces in

addition to interfaces A-G already defined in the GSM-PLMN:

Gb - between an SGSN and a BSS; Gb allows the exchange of signaling and user

data: Unlike the A-interface, in which a user is assigned a certain physical resource

for the entire/full duration of a connection, on Gb a resource is only assigned in case

of activity (i. e. when data are being transmitted/received). A large number of sub-

scribers use the same physical resources. The same holds for interfaces Gi, Gn and

Gp.

Gc - between a GGSN and an HLR

Gd - between an SMS-GMSC / SMS-IWMSC and an SGSN

Gf - between an SGSN and an EIR

Gi - between GPRS and an external packet data network PDN

Gn - between two GPRS support nodes GSN within the same PLMN

Gp - between two GSN located in different PLMNs. The Gp interface allows the sup-

porting of GPRS services over an area of cooperating GPRS PLMNs.

Gr - between an SGSN and an HLR

Gs - between an SGSN and an MSC/VLR; serves to support an MS using both

GPRS and circuit switched services (e.g. update of location information).

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PSTN

X.25

Common GSM/GPRS/UMTS Network:

Interfaces, Network Elements

ISDN

IP

IWF/TC: Interworking Function / Transcoder

IWF/

TC

A

Gb

Iu(PS)

Gi

GMSC

GGSN

GSM Phase 2+

Core Network

MSC

SGSN

HLR/ACEIRCSE

Iu(CS)

A

Gn

T

R

A

U

B

S

C

BTS

BTS

Abis

UE

(USIM)

Uu

Um

MS

(SIM)

E

SMS-GMSC

SMS-IWMSC

EG

d

GSM BSS

Asub

Gs

Gr G

c

UMTS

Terrestrial

Radio

Access

Network

Gf

VLR

SLR

Fig. 12 Common GSM/GPRS/UMTS core network, coexistence of two radio access networks (GSM BSS/UTRAN)

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3.3 New Network Elements for GPRS

3.3.1 Serving GPRS Support Node (SGSN) Functions

SGSN realizes a large number of functions for performing GPRS services.

SGSN is on the same hierarchic level as an MSC and handles many functions com-

parable to a Visited MSC (VMSC).

SGSN

is the node serving GPRS mobile stations in a region assigned to it;

traces the location of the respective GPRS MSs (Mobility Management functions);

is responsible for the paging of MS;

performs security functions and access control (authentication/cipher setting proce-

dures,...) Procedures are based on the same algorithm, ciphers and criteria as in the

former GSM. Ciphering algorithms have been optimized for the transmission of

packet data;

has routing/traffic-management functions;

collects data connected with fees/charges;

realizes the interfaces to GGSN (Gn), PCU (Gb), other PLMNs (Gp), HLR (Gr),

VLR (Gs), SMS-GMSC (Gd), EIR (Gf).

3.3.2 Gateway GPRS Support Node (GGSN) Functions

GGSN realizes functions comparable to those of a gateway MSC.

GGSN

is the node allowing contact/interworking between a GSM PLMN and a packet

data network PDN (realization Gi-interface);

contains the routing information for GPRS subscribers available in the PLMN.

Routing information serves to contact the respective SGSN in the providing area of

which an MS is momentarily located;

has a screening function;

can inquire about location informations from the HLR via the optional Gc interface

transfers data/signaling to SGSN via Gn interface.

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GGSN

SGSN & GGSN

SGSN

Serving GPRS Support Node SGSN

• serves MSs in SGSN area

• Mobility Management functions, e.g

Update Location, Attach, Paging,..

• Security and access control:

Authentication, Cipher setting, IMEI Check...

New cipher algorithm

• Routing / Traffic-Management


• realises Interfaces: Gn, Gb, Gd, Gp, Gr, Gs, Gf

• controls subscribers in its service area (SLR)

Gateway GPRS Support Node GGSN

• Gi-,Gn-Interface: Interworking PLMN ↔ PDN

• Routing Information for attached GPRS user

• Screening / Filtering


• optional Gc interface

Fig. 13 Tasks of GGSN and SGSN

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3.3.3 Physical Realization SGSN/GGSN

SGSN and GGSN functions, respectively, can be located within the same physical

unit or at different locations in different physical units. SGSN and GGSN include the

internet protocol (IP) routing function and can be linked together/Interconnected with

IP routers (IP-based GPRS backbone network for Gn). The same holds for the Gp in-

terface (SGSN and GGSN in different PLMNs); in addition there are safety functions

for inter-PLMN communication.

HLR (GPRS Register GR)

HLR includes the GPRS subscriber information (GPRS Register GR) and routing in-

formation. Access to HLR is possible from SGSN via Gr and from GGSN via Gc inter-

face.

SGSN & GGSN:

physical location

External

IP Network

GGSN

SGSN

HLR (GR)

BSSPCU

GPRS-MS

MSC/VLR

BSSPCU

HLR:

• GPRS subscriber data

(GPRS Register GR)

• Routing information

Gb

Gb

Gi

GrGs

SGSN & GGSN

in same

physical entity

SGSN

GGSN

SGSN

GGSN

GGSN

BSSPCU

GPRS-MS

BSSPCU

External

X.25 Network

IP-based

Backbone

Network

Gn

Gp

Security functions

for Inter-PLMN

communication

other

PLMN

SGSN & GGSN

in different

physical entities /

location

External

IP Network

Fig. 14 Different physical locations of SGSN and GGSN

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3.3.4 Packet Control Unit PCU

In the BSS, the PCU serves

for the management of GPRS radio channels (Radio Channel Management func-

tions), e.g. power control, congestion control, broadcast control information

for the temporal organization of the packet data transfer for uplink and downlink

it has channel access control functions, e.g. access request and grants

it serves for converting protocols from the Gb interface to the radio interface Um.

Three options for positioning the PCU are provided in Rec. 03.60:

Option A: In the BTS

Option B: in the BSC

Option C: In spatial connection with the SGSN

The different positions may be used due to the different solutions of the vendors and

with regard to the traffic, which has to be handled by the PCU/BSS.

3.3.5 Channel Codec Unit CCU

The CCU contains the following functions:

Channel coding, including forward error correction FEC and interleaving

Radio channel measurements, including received quality and signal level, timing ad-

vance measurements

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CCU

CCU

PCU

BTS BSC site GSN site

CCU

CCU


CCU

CCU


PCU

PCU

A

B

C

optional:

PCU-locationPCU, CCU, GPRS - MS

Um Abis

Gb

MS

MS

MS


• Channel Access Control functions

• Radio Channel Management functions

(Power Control, Congestion Control,...)

• scheduling data transmission (UL/DL)

• protocol conversion (Gb ↔ Um)

Gb


• Channel Coding (FEC, Interleaving,..)

• Radio Channel Measurementfuncions

(received quality & signal level, TA,..)

Fig. 15 Positioning of the new network elements in the GSM BSS

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3.3.6 GPRS Mobile Stations MS

A GPRS MS can work in three different operational modes. The operational mode

depends on the service an MS is attached to (GPRS or GPRS and other GSM ser-

vices) and on the mobile station’s capacity of simultaneously handling GPRS and

other GSM services.

„Class A“ operational mode: The MS is attached to GPRS and other GMS services

and the MS supports the simultaneous handling of GPRS and other GSM services.

„Class B“ operational mode: The MS is attached to GPRS and other GMS services,

but the MS cannot handle them simultaneously.

„Class C“ operational mode: The MS is attached exclusively to GPRS services.

Note: Various GSM specifications use the terms GPRS Class-A MS, GPRS Class-B

MS, GPRS Class-C MS.

GPRS-Mobile Station

Class A

Simultaneously handling

of GPRS and other

GSM services

Class B

GPRS and GSM

services but not

simultaneously

Class C

Only GPRS services

Fig. 16 GPRS mobile stations

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4 Logical Functions

Logical Functions

GPRS


Fig. 17

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4.1 Logical Functions in the GPRS Network

The tasks required for the handling of processes in the GSM-/GPRS network are

structured into logical functions. These functions may contain a large number of indi-

vidual functions. Logical functions are:

Network access control functions

Packet routing and transfer functions

Mobility management functions

Logical link management functions

Network management functions.

Logical functions

in GPRS networks

Network Access

Control

Functions

Mobility

Management

Functions

Radio Resource

Management

Functions

Packet Routeing

& Transfer

Functions

Logical Link

Management

Functions

Network

Management

Functions

Fig. 18 Logical functions of the GPRS network

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4.1.1 Network Access Control Functions

Network access means the way or manner in which a subscriber gains access to a

telecommunication network to make use of the services this network provides. An

access protocol consists of a defined set of procedures, which makes access to the

network possible. Network access can be obtained both from the MS and from the

fixed network part of the GPRS network. Depending on the provider, the interface to

external data networks can support various access protocols, e.g. IP or X.25. The fol-

lowing functions have been defined for access to the GPRS network:

Registration function: Registration stands for linking the identity of the mobile radio

subscriber to his packet data protocol (or protocols), the PLMN-internal addresses

and the point of access of the user to external data Protocol (PDP) networks. This

link can be static (HLR entry), or it can be effected on demand.

Authentication and authorization function: This function stands for the identifica-

tion of the subscriber and for access legitimacy when a service is demanded. In addi-

tion, the legitimacy of the use of this particular service is controlled. The authentica-

tion function is carried out in conjunction with the mobility management functions.

Admission control function: Admission control is intended for determining the net-

work resources required for performing the desired service (QoS). It also decides

whether these resources are available, and lastly it is used for reserving resources.

Admission control is effected in conjunction with the radio resource management

functions to enable assessment of radio resources requirements in each individual

cell.

Message screening function: A "screening" function is combined with the filtering of

unauthorized or undesirable information/messages. In the introduction stage of

GPRS a network-controlled screening function is supported. Subscription-controlled

and user-controlled screening may be additionally provided at a later stage.

Packet terminal adaptation function: This function adapts data packets re-

ceived/transmitted from/to the terminal equipment TE to a form suited for transport

through the GPRS network.

Charging data collection function: This function is used for collecting data required

for billing

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Network Access Control Function

Registration:

User‘s mobile ID associated with

*user‘s PDP

*address

*access points

Authentication &

Authorisation

*user

*requested services

Admission Control

*required resources

(available resouces)

(reservation of resources)

Message Screening

Filters unsolicited and

unauthorised messages

Packet Terminal Adaption

Adaption of data packets

between

MS-TE and GPRS-network

Charging Data Collection

Subscription fees + traffic fees

Fig. 19 Network access control functions

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4.1.2 Packet Routing and Transfer Functions

A route consists of an orderly list of nodes used for the transfer of messages within

and between the PLMNs. Each route consists of the node of origin, no node, one or

several relay nodes, and the node of destination. Routing is the process of determin-

ing and using the route for the transmission of a message within or between PLMNs.

Relay function: Transferring data received by a node from another node to the next

node of the route.

*Routing function: Determining the transmission path for the next hop on the route

towards the GPRS support node (GSN) the message is intended for. Data transmis-

sion between GSNs can be effected via external data networks possessing their own

routing functions; e. g. X.25, Frame Relay or ATM networks.

Address translation and mapping function: Address translation means transforming

one address into another, different address. It can be used to transform addresses of

external network protocols into internal network addresses (for routing purposes).

Address mapping is used to copy a network address into another network address of

the same type (e.g. for the routing and transmitting of messages from one network

node to the next).

Encapsulation function: Encapsulation means supplementing address- and control in-

formation into one data unit for the routing of packets within or between PLMNs. The

opposite process is called decapsulation. Encapsulation and decapsulation is ef-

fected between the GSN of the GPRS-PLMN as well as between the SGSN and the

MS.

Tunneling Function: Tunneling means the transfer of encapsulated data units in the

PLMN. A tunnel is a two-way point-to-point path, only the endpoints of which are

identified.

Compression function: for the optimal use of radio link capacity.

Ciphering function: preventing eavesdropping

Domain name server function: Decoding logical GSN names in GSN addresses. This

function is a standard function of the internet.

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Packet Routing & Transfer Function

Relay

forward data packetsRouting

„next hop“

Address Mapping

&Translation

Encapsulation

Tunneling

Compression

Ciphering Domain Name

Server

Fig. 20 Packet routing and transfer functions in the GPRS network

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4.1.3 Mobility Management Functions

Mobility management functions are used to enable tracing the actual location of a

mobile station in either the home-PLMN or a Visited-PLMN.

4.1.4 Logical Link Management Functions

Logical link management functions concern maintenance of a communication chan-

nel between an MS and the PLMN via the radio interface Um. These functions in-

clude the coordination of link state information between the MS and the PLMN and

the monitoring of data transfer activities via the logical link.

Logical link establishment function: Building up a logical link by during GPRS at-

tach.

Logical link maintenance function: Monitoring of the state of the logical link and

state modification control.

Logical link release function: De-allocation of resources associated with the logical

link.

4.1.5 Radio Resource Management Functions

Radio resource management functions include allocation and maintenance of com-

munication channels via the radio interface. The GSM radio resources must be di-

vided /distributed between circuit switched services and GPRS.

Um management function: Managing available physical channels of cells and de-

termining the share of radio resources allocated for use in the GPRS. This share may

vary from cell to cell.

Cell selection function: Allows the MS to select the optimal cell for a communication

path. This includes measurement and evaluation of the signal quality of neighboring

cells and detection and avoidance of overload in the eligible cells.

Um-tranx function: Offers capacity for packet data transfer via Um. The function in-

cludes a. o. procedures for multiplexing packets via shared physical channels, for re-

taining packets in the MS, for error detection and correction, and for flow control.

Path management function: Management of packet data communication between

BSS and serving GSN node. Establishing and canceling these paths can be effected

either dynamically (amount of traffic data) or statically (maximum load to be expected

for each cell).

4.1.6 Network Management Functions

Network management functions provide mechanisms for the support of GPRS-

related operation & maintenance functions.

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Maintenance of communication channel,

co-ordination Link state information & supervision of

data transfer activity over the logical link MS - SGSN

• Logical Link Establishment

• Logical Link Maintenance

• Logical Link Release

Keep track of current MS-location

Mobility Management Functions

Allocation & maintenance of radio communication path

• Um Management: manage resources GPRS / non GPRS

Cell Selection:select optimal cell (by MS)

• Um-tranx: MAC via Um, user multiplexing, packet discrimination

within MS, error detection & correction, flow control procedures

• Path Management:

manages packet data communication

BSS↔SGSN

(dynamic → data traffic or static)

Radio Resource

Management Functions

mechanism to support O&M

functions related to GPRS

Network Management

Functions

Logical Link

Management Functions

Fig. 21 Mobility management, logical link, radio resource and network management functions

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4.2 Allocation of Logical Functions

The tasks described in the logical functions can be allocated to various functional

units of the GSM-/GPRS network. The mobile station MS, the base station subsys-

tem BSS (with the packet control unit PCU and channel codec unit CCU), the serving

GPRS support node SGSN and the gateway GPRS support node GGSN participate

in handling the following functions:

Function MS BSS SGSN GGSN HLR

Network Access Control:

Registration X

Authentication & Authorization X X X

Admission Control X X X

Message Screening X

Packet Terminal Adaptation X

Charging Data Collection X X

Packet Routing & Transfer:

Relay X X X X

Routing X X X X

Address Translation & Mapping X X X

Encapsulation X X X

Tunneling X X

Compression X X

Ciphering X X X

Domain Name Server X

Mobility Management X X X X

Logical Link Management:

Logical Link Establishment X X

Logical Link Maintenance X X

Logical Link Release X X

Radio Resource Management:

Um Management X X

Cell Selection X X

Um-Tranx X X

Path Management X X

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GPRS Radio Interface Siemens

Contents

1 The Radio Interface (Layer 1) 23

1.1 Layer 1 of the GSM-/GPRS-Radio Interface Um 34

1.2 Channel Bundling, Sharing of Channels 56

1.3 Radio Block 78

1.4 Coding Schemes: 190

1.5 Logical GPRS Radio Channels 134

1.6 Multiframes in GPRS 178

2 Exercises 21

3 Solutions 25

GPRS Radio Interface

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1 The Radio Interface (Layer 1)

The Radio Interface Um

(Layer 1)

GPRS:

Interfaces

Fig. 1

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1.1 Layer 1 of the GSM-/GPRS-Radio Interface Um

By introducing GPRS services into the GSM-PLMN, worldwide modifications are

necessary also in the area of physical transmission (layer1) via the air or radio inter-

face Um. The tasks of layer 1 radio interface relate to the transmission of user and

signaling data as well as to the measuring of receiver performance, cell selection, de-

termination and updating of the delayed MS transmission (timing advance TA), power

control PC and channel coding.

In the GPRS, a decisive difference to the realization of the connection-oriented ser-

vices (circuit-switched services) relates to the fact that a physical channel and a so-

called packet data channel can be used by several mobile stations at the same time.

One packet data channel is allocated per radio block, i.e. for four consecutive TDMA

frames and not for a specific time interval. This means that signaling and the packet

data traffic of several mobile stations can be statistically multiplexed into one packet

data channel. Furthermore, the packet data channel can be seized asymmetrically.

On the other hand it is also possible for a mobile station to use more than one packet

data channel at the same time, i.e. to combine several physical channels of one radio

carrier. In principle, up to 8 packet data channels can be seized simultaneously. The

number of channels that are combined for reception (DL) and transmission (UL) can

be different to achieve asymmetric data rates for certain applications (e.g. file transfer

protocol FTP, internet surfing).

The assignment of radio resources can be done dynamically or in a fixed allocation.

In case of the fixed allocation a message with a bit pattern is sent downlink to indi-

cate which channels can be used by this MS for UL transmission.

If dynamic allocation is applied the MS will be receive a temporary flow identifier (TFI)

and an uplink state flag (USF) for each of the time slots it is allowed to use. The TFI

is part of the control information in the DL packet and identifies the "owner" of the

packet. Each packet also includes an USF that indicates which of the MSs (that has

been assigned to use this time slot UL) is allowed to transmit the next radio block UL.

GPRS Radio Interface Siemen

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GSM RF:

GPRS Layer 1 (Um)

L1-

tasks

Transmission

of user &

signaling data

determinate &

actualise

Timing Advance

Cell Selection

Measure

signal strength

Power Control

functions Resource optimization:

1 physical channel to be used

by many MSs simultaneously !!

asymmetrical traffic

UL / DL possible !!

High data rate traffic

up to 171.2 kbit/s:

combining 1..8 PDCH for 1 MS !!

Allocation of physical channel

(Packet Data Channel PDCH)

dynamically: 1 or 4 Radio Blocks

(1 Radio Block = 4 Normal Burst

in 4 consecutive TDMA-frames)

⇒ User & signaling data of several MSs

statistically to be multiplexed into 1 PDCH

(also fixed allocation possible)29 multislot classes

Fig. 2 Tasks of the GSM air interface, layer 1 (GSM RF)

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1.2 Channel Bundling, Sharing of Channels

Sharing of Resources in a Cell: GSM circuit switched (CS) users will share the time

slots in a BTS with the GPRS packet switched (PS)users. A physical channel can ei-

ther be used for GSM CS or GPRS PS traffic but not for both at the same time. De-

pending on the traffic load in the cell there will be more or less channels available for

GPRS, CS connections are dealt with priority.

Sharing of Physical Channels: It is a characteristic of a CS connection that the

physical resource (the time slot) is reserved for one subscriber. Therefore the

GSM CS users cannot share their channels with others. In contrast GPRS PS sub-

scribers can share physical channels. The handling of the channels, the multiplexing

of subscribers onto the same time slots is done by software (protocol, MAC) and

hardware (PCU). Packet oriented connections are not only carried out through the

core network by usage of an appropriate hardware (ATM switches) and software

(protocols) but also on the air interface. This is an important feature of GPRS with re-

gard to an optimized usage of resources on Um, which is the limiting bottleneck in the

PLMN.

Multislot Classes: The subscribers for GPRS will have different needs (applications,

data rates) and therefore the MS will have more or less capabilities. The network

(PCU) will have to identify these different MSs by their multislot class, which indicates

how many time slots (channels) can be bundled by the MS uplink and downlink. A

cheap GPRS mobile will be a GSM mobile that is able to handle the protocols and

coding schemes of GPRS. This will be multislot class 1: one time slot UL and one

time slot downlink can be "bundled". The other extreme is multislot class 29 which

will be able to receive and to transmit in eight time slots UL and DL simultaneously. In

consequence such a MS has to have two synthesizers, and a high battery capacity

because this is more or less continuous transmission and reception. The MS will

send its multislot class and the PCU will only assign time slot combinations which can

be handled by this equipment.


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Channel Bundling, Sharing of Channels

Radio Blocks

Subsriber A

Radio Blocks

Subscriber B

Radio Blocks

Subscriber C

Radio Block

Subscriber D

TS 0 TS 1 TS 2 TS 3 TS 4 TS 5 TS 6 TS 7 TS 0 TS 1 TS 2 TS 3 TS 4 TS 5 TS 6 TS 7

UL DL

Fig. 3 Channel bundling, sharing of channels

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1.3 Radio Block

Channel coding was modified substantially for GPRS purposes (GSM Rec. 03.64).

Channel coding starts with the division of digital information into transferable blocks.

These radio blocks, i.e. the data to be transferred (prior to encoding) comprise:

a header for the Medium Access Control MAC (MAC Header)

signaling information (RLC/MAC Signaling Block) or user information (RLC Data

Block) and

a Block Check Sequence BCS.

The functional blocks (radio blocks) are protected in the framework of convolutional

coding against loss of data. Usually, this means inserting redundancy.

Furthermore, channel coding includes a process of interleaving, i.e. different ar-

rangement in time. The convolutional radio blocks are interleaved to a specific num-

ber of bursts/burst blocks. In the case of GPRS, interleaving is carried out across four

normal bursts NB in consecutive TDMA frames and, respectively, to 8 burst blocks

with 57 bit each.

Four new coding schemes were introduced for GPRS (Rec. 03.64): CS-1 to CS-4.

These can be used alternatively depending on the information to be transferred and

on the radio interface’s quality.


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Radio Block Strucure

collect

user data

signaling

Radio Block

RLC Data Block BCSMAC Header

RLC/MAC Control Block BCSMAC Header

BCS: Block Code Sequence

(for error recognition)

MAC: Medium Access Control RLC: Radio Link Control

One Radio Block = 4 normal bursts

Fig. 4 Radio block

Convolutional

coding

(not CS-4)

Radio Block

Radio Block

(Redundancy !)

rate 1/2 convolutional coding

Radio Block (456 Bits)

puncturingPuncturing

(only CS-2, CS-3)

Interleaving 57 Bit8 Burst-

blocks

57 Bit 57 Bit 57 Bit57 Bit•••

Channel Coding

4 new Coding Schemes:

CS-1, -2, -3, -4

Um: Allocation of PDCH for 1 / 4 Radio Blocks = 4 / 16 Normal Bursts

Fig. 5 Channel coding schemes

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1.4 Coding Schemes:

CS-1: CS-1 uses the same coding scheme as specified by Rec. 05.03 for the

SDCCH. It comprises a half rate convolutional code for FEC forward error correction.

CS-1 corresponds to a data rate of 9.05 kbit/s.

CS-2 and CS-3 are punctured version of the same half rate convolutional code as

CS-1. The coded bits are numbered starting from 0 and certain punctured bits are

removed.

CS-2: With CS-2 the punctured bits have numbers 4 ∗ i + 3 with i = 3,...,146 (excep-

tion: i = 9, 21, 33, 45, 57, 69, 81, 93, 105, 117, 129, 141). This means that none of

the first 12 bits is punctured. CS-2 corresponds to a data rate of 13.4 kbit/s. Remark:

For CS-2 the puncturing pattern must be adapted to the future new TRAU frame for-

mat in order to be used via the Abis interface (e.g. more bits must be punctured to

make space for RLC signaling).

CS-3: With CS-3 the punctured bits have numbers 6 ∗ i + 3 and 6 ∗ i + 5 with i =

2,...,111. CS-3 corresponds to a data rate of 15.6 kbit/s.

CS-4: CS-4 has no redundancy (no FEC) and corresponds to a data rate of 21.4

kbit/s.

By bundling up to 8 packet data channels of one carrier into one MS, transmission

rates up to 171.2 kbit/s are possible.


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9,05 kbit/s 13,4 kbit/s 15,6 kbit/s 21,4 kbit/s

CS-1 CS-2 CS-3 CS-4 Different

Redundancy

(FEC)

Quality Um→

Coding

Scheme

Code

Rate

Radio

Block*

Coded

Bits

Punctured

Bits

Data Rate

kbit/s

CS-1 1 / 2 181 456 0 9,05

CS-2 ≈ 2 / 3 268 588 132 13,4

CS-3 ≈ 3 / 4 312 676 220 15,6

CS-4 1 428 456 0 21,4

Channel Coding: Coding Schemes

* Radio Block without

Uplink State Flag USF &

Block Check Sequence BCS

Fig. 6 Coding schemes of GPRS, CS1 with high redundancy, CS4 no redundancy, radio blocks

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GPRS Channel Coding

Existing channel coding procedures have been modified with a view to introducing

the GPRS. New coding schemes CS 1-4 were specified from ETSI 4. Basically, they

make it possible to transmit 9.05 kbit/s (CS-1), 13.4 kbit/s (CS-2), 15.6 kbit/s (CS-3)

and 21.4 kbit/s (CS-4) per timeslot, respectively.

On the Abis interface, transport capacity is restricted to 16 kbit/s owing to the fact that

existing TRAU frames are used. The transmission of data for CS-3 and CS-4 would

require larger transport capacities via Abis and would thus involve serious modifica-

tions in the existing network architecture. For this reason, only coding schemes CS-1

and CS-2 are supported in GR2.0/BR5.5. Of these two, CS-1 is particularly important.

Due to the unrestricted redundancy in data transmission, CS-1 is well suited to serve

as a safe basic coding for RLC/MAC data and control blocks. With a high-quality ra-

dio interface CS-1 data transmission rates of up to 8 kbit/s are possible. Even if the

air interface quality (the C/I ratio) decreases, the rate of transmission decreases very

slowly.

Under favorable radio transmission conditions, CS-2 achieves higher transmission

rates, with a maximum at 12 kbit/s. However, the rate of transmission depends more

strongly on the C/I ratio than with CS-1.

This is even truer of coding schemes CS-3 and CS-4, respectively, whose transmis-

sion rates are considerably higher than those of CS-1 and CS-2 under good radio

transmission conditions; but they rapidly decrease if the quality of the radio transmis-

sion interface gets worse.


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CS 1 - 4: Bit Rate Comparison

18 17 16 15 14 13 12 11 10 9 8 7 6

NetThroughput(kbit/s)

0

2

4

6

8

10

12

14

16

18

20

CS1

CS2

CS3

CS4

5

Channel Coding

• Introduction: CS-1 (9,05 kbit/s & CS-2 (13,4 kbit/s)

• CS-1: basic coding for RLC/MAC data & control blocks

• no CS-3 (15,6 kbit/s), CS-4 (21,4 kbit/s)

→ Abis limitation (current TRAU frames: 16 kbit/s)

Carrier / Interference C/I (dB)

Fig. 7 Comparison of the efficiency of the four coding schemes under realistic circumstances of the air interface

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1.5 Logical GPRS Radio Channels

Use of "classical" logical channels for GSM-CS

A Logical channel is used for a special purpose/contents. For example the MSs have

to find out if this cell is a suitable one (operated by the "right" network operator),

which features are offered (e.g. HR/FR/EFR, GPRS, ...), what is the structure of Um

(channel combination), ... This is provided by the BCCH which is naturally only

transmitted in the downlink. Some resources have to be given for initial access for the

MS (RACH). For these reasons logical channels have been defined to fulfill all tasks,

which are necessary in a GSM network on the air interface (see figure 13).

The GPRS subscribers will share the air interface with the circuit switched users. On

the other hand the protocol structure of GPRS is different from "classical" GSM-CS.

Therefore the user traffic and (part of) the signaling will have to be separated. Before

this separation can take place the different MS (GPRS/non-GPRS) have to be han-

dled by signaling procedures for access (channel assignment. There are two solution

of this problem. The first one is to use (some of) the logical channels for GSM-CS:

The GPRS-MS detects the BCCH of this particular cell and looks for the system in-

formation to find out if GPRS is available. If this is a cell belonging to the same rout-

ing area the MS can choose this cell and wait for paging or for the user to use the

RACH for activating a PDP. In case that the user wants to run an PS application the

GPRS MS will use an access burst (RACH) which indicates that this is a GPRS MS

and the request will be answered by the PCU assigning resources for packet

switched traffic (time slots reserved for GPRS). Signaling (e.g. for authentication) will

then take place using these resources indicated by the message in the AGCH.

So GPRS uses some of the logical channels of GSM-CS. On one hand this can be

an advantage if the resources are sufficient. On the other hand if in the future more

and more GPRS traffic has to be handled, separate logical channels reserved for

GPRS MS will have to be given. This is the second solution. In any case the GPRS

MS will have to look for the BCCH of the cell to find out if GPRS is available. If the

second solution has been chosen the GPRS MS will also read information where a

PBCCH (Packet Broadcast Control Channel) is to be found (which time slot). This

second solution will be explained in figure 14.


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Allocation of dedicated signalling channel

Dedicated signaling MS ↔ BTSE (Call

Setup, LUP, Security, SMS, CBCH,...)

Signaling

Traffic

User Data

CGI, FR/EFR/HR, GPRS available

frequency hopping, channel combination,...)

Time synchronisation + BSIC, TDMA-No.

Traffic Channel/H

DL

DL

UL

UL + DL

DL

UL

+

BCCH

FCCH

SCH

PCH

AGCH

RACH

SDCCH

SACCH

FACCH

TCH/F

TCH/H

frequency synchronisation

Paging / Searching (MTC)

Request for access

Measurement Report,

TA, PC, cell parameters,...

Signaling instead of TCH

BCH

CCCH

DCCH

User traffic (Full Rate)

User traffic (Half Rate)

Logical Channel

(for GSM Circuit Switched)

Synchronisation Channel

Frequency Correction Channel

Access Grant Channel

Random Access Channel

Paging Channel

Broadcast Control Channel

Stand Alone Dedicated

Control Channel

Broadcast Channel

Slow Associated

Control Channel

Fast Associated

Control Channel

Traffic Channe/Fl

Dedicated Control Channel

Common Control Channel

NCH

Notification Channel

Notifying MSs

Fig. 8 "Classical" logical channels of GSM may be used by GPRS users too

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Use of new logical channels for GPRS

In addition to the nine existing logical radio channels used for signaling (BCCH, SCH,

FCCH, PCH, RACH, AGCH as well as SDCCH, SACCH and FACCH) and the Traffic

Channel (TCH) for circuit switched user information, a new set of logical channels

was defined for GPRS.

Packet traffic is realized by means of the Packet Traffic Channel (PTCH), which in-

cludes the following:

Packet Data Traffic Channel PDTCH.

Packet Associated Control Channel PACCH

Packet Timing advance Control Channel PTCCH

The PDTCH is temporarily assigned to the mobile stations MS. Via the PDTCH, user

data (point-to-point or point-to-multipoint) or GPRS mobility management and session

management GMM/SM information is transmitted.

The PACCH was defined for the transmission of signaling (low level signaling) to a

dedicated GPRS-MS. It carries information relating to data confirmation, resource al-

location and exchange of power control information.

New GPRS signaling channels are mainly specified analogously to GSM Phase1/2.

The Packet Common Control Channel PCCCH has been newly defined. It consists

of a set of logical channels, which are used for common control signaling to start the

connection set-up:

Packet Random Access Channel PRACH

Packet Paging Channel PPCH

Packet Access Grant Channel PAGCH

Packet Notification Channel PNCH

PRACH and PAGCH fulfill GPRS-MS functions, which are analogue to the “classical”

logical channels RACH and AGCH for non-GPRS-users. The PNCH is used for the

initiation of point-to-multipoint multicast (PtM multicast).

For the transmission of system information to the GPRS mobile stations, the

Packet Broadcast Control Channel PBCCH

was defined analogue to the “classical” BCCH.

In a physical channel all different types of logical channels can be contained (no

separation into traffic and signaling channels respectively as is done in conventional

GSM). The differentiation of channel contents is carried out per radio block using the

MAC header, i.e. contents are specified for the four normal bursts of a radio block

sent in each case.

The MAC function, which distributes the physical channel to the various mobile sta-

tions and allocates radio resources to an MS can also use the conventional logical

channels in GSM.


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Logical channels

for GPRS

PBCCH

PRACH

PPCH

PNCH

PAGCH

PACCH

PTCCH/U

PTCCH/D

PDTCH

Packet Broadcast

Control Channel

Packet Random

Access Channel

Packet Paging

Channel

Packet Notification

Channel

Packet Access

Grant Channel

Packet Associated

Control Channel

Packet Timing Advance Control

Channel Uplink/Downlink

Packet Data

Traffic Channel

Packet

Signaling

Packet

Traffic

Broadcast channel

Common

Control

channels

Dedicated channels

Packet System

Information

Access request for

UL packet data

transmission

Paging GPRS-MS

(PtP)

Paging GPRS-MS

(PtM)

Resource allocation

Dedicated signaling

MS-network,

e.g.power control

Timing advance

Determination and

Control

Transmission of

User data

UL

DL

DL

DL

UL&DL

UL&DL

UL

Fig. 9 New logical channels for GPRS

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1.6 Multiframes in GPRS

The GPRS packet data traffic is arranged in 52-type multiframes (GSM Rec. 03.64).

52 TDMA frames in each case are combined to form one GPRS traffic channel multi-

frame, which is subdivided into 12 blocks with 4 TDMA frames each. One block

(B0-B11) contains one radio block each (4 normal bursts, which are related to each

other by means of convolutional coding). Every thirteenth TDMA frame is idle. In the

idle frame the PTACCH is sent. The idles frames are used by the MS to be able to

determine the various base station identity codes BSIC, to carry out timing advance

updates procedures or interference measurements for the realization of power con-

trol.

For packet common control channels PCCH, conventional 51-type multiframes can

be used for signaling or 52-type multiframes. The GPRS users can use "classical"

common control channels of GSM before they will be directed onto their PTCHs. All

mobiles will read the BCCH anyway. Either in case of GSM mobiles to fulfill the same

tasks as before and for GPRS equipment this logical channel will indicate weather

GPRS service is available and if extra logical channels (PBCCH, PPCH, ...) are used.

GSM CS traffic and GPRS subscribers are clearly separated so that there is no con-

flict due to different signaling or multiframe structure.

It is important that there are no "visible" changes for "GSM only mobiles" due to the

introduction of GPRS. GSM CS connections will use for example the same 26 multi-

frame structure for TCH and the 51 multiframe structure for signaling.


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iB0 B1 B2 B3 B4 B5 i B6 B7 B8 i B9 B10B11 i

52 TDMA Frames = PDCH Multiframe

4 Frames 1 Frame

New multiframe

for GPRS

• PDCH follows 52 multiframe structure

• 52 Multiframe: 12 Blocks à 4 TDMA-frames

• PCCCHs: „classical“ 51er Multiframes

or 52er Multiframes

B0 - B11 = Radio Blocks (Data / Signaling)

i = Idle frame (PTCCH)

• BCCH indicates PDCH with PBCCH (in B0)

• DL: this PDCH bears PDCCH & PBCCH

PBCCH in B0 (+ max. 3 further blocks; indicated in B0)

PBCCH indicates PCCCH blocks & further PDCHs with PCCCH

• UL: PDCH with PCCCH: all blocks to be used for PRACH, PDTCH, PACCH

PDCH without PCCCH: PDTCH & PACCH only

Idle frame:

• Identification of BSICs

• Timing Advance Update Procedure

• Interference measurements

for Power Control

Fig. 10 Multiframes for GPRS consist of a certain time slot in 52 consequent TDMA frames

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Procedures Sidemen's

Contents

1 Activation of GPRS Services 23

1.1 GPRS Identities 34

1.2 Mobility Management States 68

1.3 Packet Data Protocol PDP States 192

1.4 GPRS Packet Data Transmission 101 14

1.5 Combined GPRS & IMSI Attach 16

1.6 PDP Context Activation Procedure 18

1.7 Start of Mobile Originated Packet Transfer 20

2 Exercise 23

3 Solution 27

Procedures

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1 Activation of GPRS Services

Activation of

GPRS services

GPRS:

Procedures

Fig. 1

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Sidemen's Procedures

1.1 GPRS Identities

1.1.1 Regional Organization of GPRS

A set of identities were introduced in GSM and GPRS to identify a subscriber, as well

as to keep track of him. Following identities are well known in GSM:

LAI: (Location Area Identity) covers a set of cells, where a subscriber was "seen"

last.

CGI: (Cell Global Identity) the unique number of a cell of a PLMN, composed the LAI

and the CI (cell identity).

Next to the existing GSM identities there is also a new GPRS specific identity, the

RAI (Routing Area Identity). This identity, defined by an operator, comprises one or

several cells. It is broadcasted by the (P)BCCH. If a GPRS mobile leaves a routing

area, a Routing Area Update Procedure has to be taken place. The RAI is used in the

same way as the LAI. The Routing Area is more precise than the location area. A

Routing Area is a subset of one and only one Location Area.

RAI: LAI + RAC (Routing Area Code) = MCC + MNC + LAC + RAC

Regional Organisation of GPRS

Location Area

Routing area

cell

LACMNCMCC

RACLACMNCMCC

CILACMNCMCC

Fig. 2 Regional organization of GPRS

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1.1.2 Subscriber Identities and Subscriber Services

IMSI (International Mobile Subscriber Identity):

This is an unique number allocated to each subscriber in GSM. This was adapted

also for GRPS-only mobile subscribers.

PTMSI (Packet Temporary Mobile Subscriber Identity):

This identity is allocated to each GPRS attached mobile. Its task is similar to the

TMSI. The discrimination between the TMSI and P-TMSI is realized by allocation to

the two most significant bits to 11 for GPRS and to 00, 01, 10 for GSM.

PDP Address:

On the network layer, the subscriber may identified by one or more network layer ad-

dresses, so-called PDP Addresses, which are allocated to the subscriber temporally

or permanently.

One central question in GPRS is: how can a logical link between a mobile and a

SGSN be identified uniquely? This is done with the NSAPI/TLLI pair, which are

unique within a routing area.

NSAPI (Network layer Service Access Point Identifier):

The NSAPI is used as a service access point between the higher level and the

SNDCP. The NSAPI is used to identify the corresponding PDP context, which is as-

sociated with the GPRS MS PDP address on the side of the GSN.

TLLI (Temporary Logical Link Identity):

The TTLI is used to define a one to one correspondence within a Routing Area be-

tween the MS and the SGSN. This is only known by the MS and the SGSN.

TID (Tunnel Identifier):

This identity is used by the GTP to identify a PDP context. The TID is a combination

of the IMSI and the NSAPI. The IMSI/NSAPI pair uniquely identifies a PDP context.

GSN-Address:

The GSN Address is the IP-no. of GSN for the GPRS IP backbone.

The GSN-number is the ISDN-no. for a GSN

Access Point Name:

This name indicates in the NSS backbone, which GGSN shall be used. Furthermore

it can indicate the external network, the subscriber wants to be attached to, for in-

stance the "Internet Service Provider" Name.

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Subscribers Identities

S

G

S

NG

G

S

N

G

G

S

N

TLLI IMSI

Who is the owner of one packet

Which application does the packet belong to

S

G

S

N

G

G

S

N

NSAPI

1 2

3 4

Fig. 3 Subscribers identities in the network

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1.2 Mobility Management States

States of the GPRS services

With regard to point-to-point PtP packet data transmission the GPRS service oper-

ates in two independent state models/circles. One circle describes the mobility man-

agement behavior whereas the other is assigned to the activation of a packet data

protocol PDP.

The circle related to mobility management states in the MS and the associated SGSN

consist of the:

"Idle" state

"Standby" state

"Ready" state

The circle related to a specific packet data protocol has the:

"Inactive" state

"Active" state

Packet Data

Protocol

PDP

States of

GPRS services

2 circles

regarding:

Inactive

State

Active

State

Idle

State

Ready

State

Standby

State

Mobility

Management

Fig. 4 States of GPRS services with regard to mobility management and packet data protocols

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"Idle" state

A mobile station MS in the idle state is detached from the GPRS. Only GPRS sub-

scription data is available in the HLR. No further information exists in other network

units such as SGSN and GGSN. It is not possible to activate a packet data protocol

PDP or to maintain a PDP in its active state. The GPRS MS must monitor the BCCH

to determine the availability of cells, which support GPRS services. Accordingly, the

GPRS MS can carry out PLMN and cell selection procedures. To exit idle state, the

MS must execute the “attach” procedure. Upon successful completion of this proce-

dure, the MS changes to ready state.

"Standby" state

In the standby state the GPRS MS is attached to the GPRS network. The GPRS and

the SGSN have a mobility management context comparable to the circuit switched

connections. The MS monitors the broadcast channel to determine the availability of

cells offering GPRS services and also the paging channel PCH, to be informed about

paging requests. The SGSN recognizes/stores the routing area RA of the GPRS-MS.

The routing area is a sub-unit of the location area LA, in other words a more detailed

determination of the GPRS-MS location. The GPRS-MS informs the SGSN about

changes of the routing area and answers paging requests.

"Ready" state

In the ready state, the SGSN detects the current cell of the GPRS-MS beyond the

routing area RA of the GPRS-MS. If the GPRS-MS changes cells, it informs the

SGSN. Paging is thus superfluous in the ready state. The DL packet data transfer

can be performed any time. Ready state does not mean that a physical connection is

established between SGSN and MS. Only in the ready state, SGSN and MS can

transfer data packets. MS and SGSN exit ready state upon expiry of a ready timer or

in case of a faulty packet data transmission and change to standby state. Upon log-

off, i.e. execution of a detach procedure; MS and SGSN exit ready state and change

to idle state.

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Mobility Management

States

IDLE

state

READY

state

STANDBY

state

GPRS

detach

expiry of mobile

reachable timer

expire READY Timer /

Transmission errors

GPRS

attach

SGSN: Paging /

MS: initiates Transfer

• SGSN & GGSN without

MS information

• only HLR contains subscription data

• no PDP context can be activated

• MS observes BCCH

• PLMN- & Cell Selection

• SGSN knows Routing Area & cell!!

• UL & DL packet transmission possible

• SGSN ↔ MS: MM-Context

• SGSN knows Routing Area

• MS observes BCCH, PCH

• initiates RA-Update

• reacts to Paging Request

• MS initiates Cell Update

Fig. 5 Mobility management states

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1.3 Packet Data Protocol PDP States

There are separate state circles for every authorized PDP of a GPRS-MS

"Inactive" State

The inactive state of a PDP means that this PDP is not operating at that moment.

There is no routing context in the MS, SGSN and GGSN. A transition in the active

state is only possible if there is a mobility management connection and if MS and

SGSN are in the standby or ready state.

No data transfer is possible in the inactive state. Data packets, which reach the

GPRS network are either rejected or ignored.

"Active" State

In the active state the MS, GGSN and SGSN are in a routing context. Data can be

transmitted or received by the MS. The active state is ended explicitly if the MS deac-

tivates a certain PDP. With GPRS detach and expiry of the standby timer, all the acti-

vated PDP are deactivated, too.

PDP States

INACTIVE

state

ACTIVE

state

De-activation PDP context /

GPRS detach

expiry STANDBY timer

Activation

PDP context

• PDP not activated

• no Routing-context

for MS, SGSN & GGSN

• no data transmission possible !

Transition to „Active“ State

only if MM-context exists

( MS & SGSN: STANDBY / READY)

• Routing context

for MS, SGSN & GGSN

• Data transmission possible !

Fig. 6 States of a packet data protocol

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1.4 GPRS Packet Data Transmission

The transmission of GPRS packet data presupposes the execution of

GPRS Attach Procedure as well as of the

PDP Context Activation Procedure.

In the case of a mobile packet data transfer, a one or two-phase packet access is

added. This access procedure is necessary for packet data transfer.

Common Mobility Management / MS-Location

To reduce the signaling load via the radio interface during GPRS and non-GPRS op-

eration, important mobility management MM procedures are carried out jointly (com-

mon MM). This regards the procedures for: attachment / detachment, location & rout-

ing area update and paging.

The result of a GPRS routing area update procedure is stored in the SGSN. The rout-

ing area represents a more exact indication of the MS location, than is actually

needed for non-GPRS services. Triggered by the MS (in the framework of a RA up-

date) the SGSN informs the MSC/VLR via the Gs interface of a change in the loca-

tion areas, which has taken place simultaneously.

Further mobility management procedures are also executed via GPRS procedures. If

possible, all messages containing mobility management information are transferred

through signaling data packets. The MM procedures are defined in the GGM/SM

(GPRS Mobility Management & Session Management).

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Abbreviations Siemen

Contents

1 Abbreviations 23

Abbreviations

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1 Abbreviations

AAL ATM Adaptation Layer

AAL5 AAL Type 5

ABC Administration and Billing Center

ACCG ASN Controller and Clock Generator

ACIS ATM Communication Interface Simulator

ACT Active

ADET Application Database Engineering Team

AGCH Access Grant Channel

ALI Alarm and Interface Module

ALIB Alarm and Interface Module Type B

ALM ATM Layer Module

AMP ATM Bridge Processor

AMX ATM Multiplexer

AMXE AMX Module type E

AP Accounting Probe

APE Abgesetzte Peripherie Einheit (Remote Peripheral Unit)

API Application Programming Interface

APS Application Program System

ASIC Application Specification Integrated Circuit

ASN ATM Switching Network

ASN.1 Abstract Syntax Notation 1

ASNF ASN Module Type F

ASNG ASN Module Type G

ASNH ASN Module Type H

ATM Asynchronous Transfer Mode

ATM230 ATM Interface Asic with 200- and 30-Mbit Interfaces

AUB Access Unit Broadband

BAP Base Processor

BCH Broadcast Channel

BCCH Broadcast Control Channel

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Siemens Abbreviations

BCT Basic Craft Terminal

BG Border Gateway

BigFUT a FUT (functional unit test) including all functional units

BIST Built In Self Test

BOP Basic Operation

BOST Board Self Test

BSC Base Station Controller

BSS Base Station System

BSSGP Base Station System GPRS Protocol

BVC Base Station Virtual Connection

C-ID Charging Identifier

CAP Coordination Processor

CBR Constant Bitrate

CCCH Common Control Channel

CCS7 Common Channel Signaling System No. 7

CCS7E Common Channel Signaling System No. 7 Enhanced

CDB Database for C-based Peripherals

CDC Central Data Collector

CGI Cell Global Identity

CGU Clock Generator Unit

CHILL CCITT High Level Language

CI Cell Identifier

CMISE Common Management Information Service Element

CP113 Co-ordination Processor 113

CT Context Table

CTI Context Table Index

CU(-C) Control Unit (shelf type C)

DBLU DBMS less Unit

DBMS Database Management System

DCCH Dedicated Control Channel

DLCI Data Link Connection Identifier

DNS Domain Name Server

DRAM Dynamic RAM

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DS1 Digital Signal, level 1

DSDL DBMS Specific Definition Language

E1 European PDH Signal, Level 1

ECC Echo Cancellation Circuit

EFD Event Forwarding Discriminator

EIR Equipment Identity Register

EPC External Processor Communication

EPROM Erasable Programmable Read Only Memory

ESGEN Extended MML Syntax Generator

ETSI European Telecommunications Standard Institute

EWSD Siemens Digital Electronic Switching System

EWSD V13 Elektronisches Wählsystem Digital Version 13

EWSX EWSXpress

FACCH Fast Associated Control Channel

FAT Functional Area Test

FCCH Frequency Correction Channel

FEPROM Flash EPROM

FFS For Further Study

FP Frame Relay Processor

FPSM Frame Relay Processor Shared Memory

FR Frame Relay

FR-LIC Frame Relay Line Interface Card

FT1 Functional Test 1 (offline-test)

FT2 Functional Test 2 (online-test)

FT3 FT2 including the HLR

FTP File Transfer Protocol

FUT Functional Unit Test

FW Firmware

GDB GPRS Database

GDMO Guidelines for definition of Managed Objects

GGSN Gateway GPRS Support Node

GMM GPRS Mobility Management

GMM_AF GMM Application Function

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GMM_TF GMM Transport Function

GOAM GPRS Operation and Maintenance Applications

GPRS General Packet Radio System

GR GPRS Release

GR1.0 GPRS Release 1.0

GSN GPRS Support Node

GTP GPRS Tunnel Protocol

GUI Graphical User Interface

HLR Home Location Register

HPDB High Performance Database

HW Hardware

HWT Hardware Tracer

I/O Input / Output

ICA IDS Communication via ATM

ICMP Internet Control Message Protocol

IDS Interactive Debugging System

IMEI International Mobile Equipment Identity

IMSI International Mobile Subscriber Identity

INT_CID Internal Change ID

INT_CID Internal Charging Identifier

IOC Input Output Controller

IOT Interoperability-Test

IOT Interoperability-Test

IP Internet Protocol

IPC Internal Processor Communication

IPv4 IP version 4

ISP Internet Service Provider

ITP Internal Transfer Protocol

ITU International Telecommunication Union

IWE Interworking Entity

L&S Load and Stress Test

L&S Load and Stress Test

LA Location Area

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LAN Local Area Network

LCF Log Control Function

LCT Local Craft Terminal

LDC Local Data Collector

LED Light Emitting Diode

LIC Line Interface Card

LLC Logical Link Control

LLE Logical Link Entity

LM Layer Management

LPS LIC Protection Switch

MAP Mobile Application Part

MBC Message Based Communication

MBS Maximum Burst Size

MCI Maintenance Craft Interface

MDB Maintenance Database

MDD Magnetic Disk Device

MIPS Million Instructions Per Second

MM Mobility Management

MMU Memory Management Unit

MOD Magneto Optical Disk

MP Main Processor

MP-AP Main Processor used for application SW processing

MP-SA Main Processor with Standalone Capabilities

MP:ACC Main Processor for Accounting Management

MP:LM Main Processor for Layer Management

MP:OAM Main Processor for Operation and Maintenance

MP:PD Main Processor for Packet Dispatching

MPC Main Processor (Version C)

MPU Main Processor Unit

MPUB Main Processor Unit B

MPUC Main Processor Unit C

MS Mobile Subscriber

MSC Mobile Services Switching Center

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MSU Message Signaling Unit

MTP Message Transfer Part

N-PDU Network PDU

NC Node Commander

NNI Node Network Interface

NS Network Service

NS-VC Network Service Virtual Connection

NS-VL Network Service Virtual Link

NSAPI Network SAPI

NSS Network Subsystem

O&M Operation and Maintenance

OA&M Operation, Administration and Maintenance

OMC Operation and Maintenance Center

OMC-B OMC for the BSS

OMC-S OMC for the SSS

OS Operations System

P-TMSI Packet Temporary Mobile Subscriber Identity

PCB Printed Circuit Board

PCH Paging Channel

PCM Pulse Code Modulation

PCP Peripheral Control Platform

PCR Peak Cell Rate

PCU Packet Control Unit

PD Packet Dispatcher

PDET Project Database Engineering Team

PDN Packet Data Network

PDP Packet Data Protocol

PDU Packet Data Unit

PLL Phase Locked Loop

PLMN Public Lands Mobile Network

PM Performance management

PRH Protocol Handler

PRH:MGR Protocol Handler Manager

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PRM Packet Routing Management

PRM-S Packet Routing Manager SGSN

PRT Packet Routing and Transfer Function

PSAX Power Supply 5V for Fibre Optic Transceiver type X

PSU Power Supply Unit

PVC Permanent Virtual Connection

Q3 Q interface at the GSN nodes

QoS Quality of Service

RA Routing Area

RAC Routing Area Code

RACH Random Access Channel

RAI Routing Area Identity

RAM Random Access Memory

RB Record Builder

RF Record Formatter

RPC Remote Procedure Call

RSS Radio Subsystem

SA Stand Alone

SAAL Signaling AAL

SACCH Slow Associated Control Channel

SAPI Service Access Point Identifier

SAR Service Access Routines

SCB Sequencer Control Block

SCB SSNC Control Shelf Basic

SCCP Signaling Connection Control Part

SCE SCB-extended

SCE SSNC Control Shelf Extended

SCH Synchronization Channel

SCR Sustainable Cell Bitrate

SDL System Description Language

SDR Symptom Data Recording

SDRAM Synchronous DRAM

SDRT Symptom Data Transport

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SGSN Serving GPRS Support Node

SICAT SDL Integrated Computer Aided Tool set

SLR SGSN Location Register

SM Session Management (GPRS)

SM Signaling Manager (part of #7 application)

SMP Standard Maintenance Protocol

SMU Statistical Multiplexing Unit

SNDCP Subnetwork Dependent Convergence Protocol

SP Synchronization Point

SPOTS Support for Planning, Operation & Maintenance and Traffic analysis

SPU Service Provision Unit

SQS Siemens Q3 Specification

SS7 Signaling System #7

SSNC Signaling System Network Control

SST Sub System Test

STATS Statistics Support

STB Standby

STM-1 Synchronous Transport Module Level 1

SVE System Verification Environment (a tool for proving the formal correctness of a design)

SW Software

SWERR Software Error Report

TCH Traffic Channel

TCP Transmission Control Protocol

TID Tunnel Identifier

TLLI Temporary Logical Link Identifier

TLM Trunk Line Management

TM Traffic Measurements

TMN Telecommunications Management Network

TODE Total Outage Detection

TPL Throughput Limiter

TSC Through Switched Connection

TTY Teletype

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UDP User Datagram Protocol

UNI User Network Interface

VBR Variable Bitrate

VC Virtual Connection

VCPU Virtual Central Processing Unit

VGA Video Graphics Adapter

vGGSN virtual GGSN

VLR Visited Location Register

VOCOC Vision O.N.E. Chill Operating System

VP Virtual Path

WAN Wide Area Network

WWW World Wide Web

xGSN SGSN or GGSN

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principles of-mobile-communication-2011

Documents

pcm signals

transmitted telephone

different telephone

timedivision multiplex

introduction of frequency

waveform samples

pam signals

division multiplex fdm