31-05-2012, 01:12 PM
Design and implementation of a test tool for the GSM traffic channel
[attachment=23675]
Introduction
2.1 Background
This master thesis work was done as part of the programme
for Computer Science and Engineering at Linköping University.
The employer was Enea Epact, but the real customer was
Ericsson Radio Network Center.
With the introduction of GPRS in the GSM network, the
need for verification of the traffic channels increase. For
example, a useful scenario for this new tool would be when
there are a lot of GPRS traffic going on through the BSC,
then a voice call is being set up and GPRS traffic channels
has to be pre-empted. It would also be interesting to use the
system for generating background load on the traffic channels,
while performing other automatic tests.
The thesis consists of two parts. The first part consists of an
analysis to find out the needs and to see how it is possible to
implement the system. The second part involves implementing
part of the system.
Assignment
4 Design and implementation of a test tool for the GSM traffic channel
2.2 Assignment
The customer has a product called Test and Simulation
Solutions (TSS), used for testing nodes in the Global System
for Mobile Communications (GSM) and General Packet
Radio Systems (GPRS). The TSS consists of special hardware
as well as software and is used to simulate traffic in
mobile cell phone networks.
Within TSS there is a problem when it comes to circuit
switched networks like the GSM. The test tool is only able
to set up the connections using the signalling channels, the
physical channels used for traffic are not verified with this
tool. When it comes to packet switched networks, like the
GPRS, this problem has been handled.
The first assignment was to see how the product TSS could
be enhanced with a new tool allowing for verification of the
GSM traffic channels. This includes how to insert traffic
(i.e. simulated speech) into the physical channel, how to
generate it, and how to control the tool in a way that is consistent
with the rest of the product. To do this, several subareas
have to be studied. The GSM network itself has to be
studied in order to find out scenarios where this new tool
can become useful. What standards are there that the new
tool must comply to in order to connect to the GSM network?
And what about the user interface part, what are the
common standard the tool must comply to in order to be easily
integrated with the TSS?
The second assignment for this master thesis was to implement
part of the system proposed. In this case the choosen
part is the enhanced full rate speech codec, which is the
most commonly used codec of cellular phone systems. This
part will involve examining the speech compression and
decompression algorithms to be able to implement them on
the target hardware.
Limitations
Design and implementation of a test tool for the GSM traffic channel 5
2.3 Limitations
Only the speech encoding and decoding functions will be
implemented since this is an area where there is not enough
knowledge at the employer or the customer. This is basically
one function which can be developed detached from the rest
of the test system and easily integrated later.
The full test system, with traffic generator, user interface
etc., will not be implemented within this master thesis due to
limitations in time. Also this area is more well known and
the both the employer and the customer has a lot of experience
in this area.
2.4 Purpose of this report
The purpose of this report is to summarize the work performed
and to document the results achieved. The situation
before is described as well as the gains from implementing
and using the proposed test system.
2.5 Reading instructions
The report is intended to be readable by most senior students
at a Master of Science programme and teachers at those programmes.
However it will also include details about the system
useful for professionals working in the area of
telecommunication using or further developing the system.
Professionals implementing the system should consider
reading [1] and [2].
Chapter 2 gives a brief introduction to the report and defines
the assignment.
Chapter 3 deals with the theory behind the GSM network
and is useful for those not working within the telecommunication
area.
Reading instructions
6 Design and implementation of a test tool for the GSM traffic channel
Chapter 4 gives an introduction to the theory behind speech
coding and the parts of a full featured speech coder.
Chapter 5 gives a deeper analysis of the need and usage of a
test tool.
Chapter 6 describes the design to be used for the proposed
system.
Chapter 7 discusses issues concerning test and implementation.
Problems, improvements and errors are discussed.
Chapter 8 summarizes the thesis and discusses drawn conclusions.
Design and implementation of a test tool for the GSM traffic channel 7
GSM basics
3.1 Telecommunication overview
The following sections will introduce terminology from the
area of mobile telecommunication. The intention is to give
an overview and to define the concepts of mobile telecommunication
and its notion. Details can be found in [3] and
[4].
3.2 Networking elements
The GSM network consists of a hierarchy of elements as can
be seen in figure 3.1. In this particular network there are two
Mobile Switching Centres (MSC), one Gateway MSC
(GMSC) and three different access networks. There are also
some databases to store the information needed in the network.
This public land mobile network (PLMN) is also connected
to the public switched telephone network (PSTN)
and one other PLMN.
The mobile station (MS) can be any type of end user equipment
like a cellular phone, laptop or fax machine with radio
modem.
Networking elements
8 Design and implementation of a test tool for the GSM traffic channel
FIGURE 3.1. GSM network hierarchy.
The Base Transceiver Station (BTS) contains equipment for
the physical radio transmitting and receiving, signal measurements,
encryption and communication with the base station
controller (BSC).
Base station controllers serve as concentrator nodes for several
BTS. It is responsible for allocating radio channels used
by the BTS and to connect speech and signalling channels to
the MSC. It also handles transfers, handovers, of mobile stations
between different BTS. This occurs for example, when
the mobile station is being moved within a limited area, for
example in a city.
The mobile switching centre corresponds to the local station
in a public switched telephone network, though it does not
have any hard wired subscribers. It is responsible for mov-
PSTN GMSC
O ther
PLMN
AUC EIR
HLR
VLR VLR
BSC
MSC MSC
BTS
BTS
BTS MS
A
Abis
Um
Interfaces
Design and implementation of a test tool for the GSM traffic channel 9
ing mobile stations between different BTS and between different
MSC. Transfer of mobile stations between BTS
occurs for example when the end user is travelling longer
distances.
The GMSC is similar to the MSC. It serves as an interface
between the network and other networks, for example the
PSTN or other PLMN. It is responsible for charging
between different network operators. There may be several
GMSC in a network, and they may be integrated with the
MSC.
There are also databases connected to the switching network
elements. The home location register (HLR) contains information
about the end user and under which MSC it currently
is registered. At each MSC there is a visitor location register
(VLR) which contains information about all mobile stations
that are currently in that MSC service area.
Authentication centre (AUC) contains information used for
encryption and authorization.
Equipment identity register (EIR) stores information about
the MS identity. It is used to control that a mobile station is
not blocked.
3.3 Interfaces
The different interfaces between the network elements are
named in figure 3.1. Starting from the MS side, first comes
the Um interface. This is the radio interface between a
mobile station and the BTS [5]. Each logical speech channel
requires 16 kbit/s data rate. With error correction codes,
burst management and encryption the used data rate is 33.8
kbit/s per channel.
The interface between the BTS and the BSC is named Abis.
It uses standard T1 or E1 cables for interconnection. Four 16
Inside the equipment
10 Design and implementation of a test tool for the GSM traffic channel
kbit/s speech channels multiplexed into one 64 kbit/s channel
corresponds to one timeslot in the E1 or T1 link between
the BTS and BSC. The job to transform the speech information
to the correct format is done by the transcoding and rate
adaption unit (TRAU). This is where the speech encoding
and decoding functionality are situated. The Abis interface
is defined in [6].
The A interface between BSC and MSC consists of 64 kbit/s
PCM links on a E1 or T1 interconnection. The speech is
here coded with a-law or μ-law PCM depending on if a
European or American/Pacific system is used. More information
about the A interface is found in [7].
3.4 Inside the equipment
The basic structure inside the switching network elements is
shown in figure 3.2. The main components in the BSC, and
MSC are a number of line interfaces (LIF), i.e. the physical
connections to the other elements, a switching unit and a
control system. There is also some kind of terminal equipment
(TERM) for operation and maintenance.
Inside the BSC there is also a pool of cpu resources for
speech transcoding and rate adaption in the TRAU module,
available for use when connecting traffic channels between
the MSC and BSC.
The BTS consists of a number of transceivers (TRX), which
are multiplexed into the radio interface. The transceivers are
accessed from the BSC via a transceiver radio interface
switching traffic channels from the BSC to the correct transceiver
and vice versa.
Inside the equipment
Design and implementation of a test tool for the GSM traffic channel 11
FIGURE 3.2. Inside the switching elements.
TRX
TRI
CONTROL
TRX
LIF
LIF
TERM
TRAU
CONTROL
M
U
X
LIF
LIF
TERM
CONTROL
BSC
GMSC/MSC
BTS
Other networks
Inside the equipment
12 Design and implementation of a test tool for the GSM traffic channel
Design and implementation of a test tool for the GSM traffic channel 13
Speech coding
4.1 General
A general communication system can be split into a couple
of modules, as in figure 4.1. The transmitter is modelled by
a source encoder, channel encoder and modulator. The
receiver can be modelled by a demodulator, channel decoder
and source decoder as seen in . Between the source and destination
side, there is the transmission media, called a channel.
Since most channels not are perfect, the channels will
add some noise to the signal.
The following sections will concentrate on the features of
the source encoder and decoder.
4.2 Human speech model
Since we are interested in compression of human speech , it
is interesting to study how speech is produced.
Sounds are produced in two different ways. There are
voiced sounds, all vowels for example, that are produced by
the vibrating vocal cords at the glottis. There are also
unvoiced sounds, like ‘p’, ‘t’, and ‘f’, which are the result of
Human speech model
14 Design and implementation of a test tool for the GSM traffic channel
FIGURE 4.1. General communication system.
air flow through a construction in the vocal tract. Note that
consonants can also be voiced sounds like ‘b’, ‘d’ and ‘j’.
Also some sounds are the result of a combination of these.
The excitation signal for these sounds is the vibration of the
vocal cords. The frequency of the vibration does not vary
greatly in time due to the physiology of the glottis, and can
therefore be modelled using a slowly varying model.
The vocal tract is a mechanical system, which means that
the vocal gestures are relatively slow. It can be modelled by
a parametric filter, where the parameters will vary with time.
The bandwidth for these filter parameters are lower than for
the speech signal itself. This is called the short time stationarity
of the vocal tract filter.
The human ear is also to be considered in the models. It is
able to perceive signals between 20 Hz - 20 kHz. However,
it is most responsive to frequencies between 200 and 5600
Hz which usually carries the most important information
when it comes to human speech. Most telephone systems
has a bandwidth limitation to the range of 300 to 3400 Hz,
which will be possible without losing to much of the quality.
Another important aspect is masking, which means that one
sound can be obscured by the presence of another. Masking
Information
source
Source
encoder
Channel
encoder Modulator
Channel
Demodulator
Channel
decoder
Source
decoder
Information
destination
Noise
Different codecs
Design and implementation of a test tool for the GSM traffic channel 15
can be done either as temporal masking or as frequency or
spectral masking. Temporal masking means that signals
with energy in frequencies close to each other can mask
each other, if the time delay between the signals are short.
Frequency masking means that a lower frequency can be
masked by a simultaneous higher frequency. This can be
used to distribute quantization noise to the masked frequency.
4.3 Different codecs
There are different types of speech codecs. The two main
types of coders are waveform coders and vocoders (voice
coders). There are also hybrid coders which combine the
properties of those coders. Some properties of these codecs
are shown in figure 4.2.
FIGURE 4.2. Comparison of compression ratio and
speech quality of different codec categories.
Waveform coding is the process of describing the signal’s
amplitude curve with a number of discrete values. This is
basically done in three steps - sampling, quantization and
coding. The quality of the signal will in general be very
good but requires more bandwidth. Waveform coders tend
to be more robust against unexpected input, like music, than
vocoders.
Low
Excellent
Good
Average
Hybrid coders
Waveform coders
Vocoders
Speech quality
b itrate
2 4 8 16 32 64 (kbit/s)
Enhanced full rate codec
16 Design and implementation of a test tool for the GSM traffic channel
Vocoders use a totally different approach. They try to figure
out which parameters were used to create the signal, given
that it passed through a known filter model. The filter model
in mind is of course a model of the human speech organs.
Vocoders lose a bit of quality of the transferred signal, but
will on the other hand use less bandwidth. The process of
transmitting speech parameters instead of the speech can be
compared to transmitting notes instead of the music itself.
Because of the two contradictionary demands of low bandwidth
usage and high signal quality in cellular phone systems,
hybrid coders are used. The coders use a combination
of parameterizing and waveform coding to achieve this. The
bitrate can be reduced to levels under the 64 kbit/s PCM
coding with good signal quality.
There are several possible speech coders for usage in cellular
phone systems (See also [8]):
• Full rate 13 kbit/s
• Enhanced full rate 12.2 kbit/s
• Half rate 5.60 kbit/s
• Adaptive multi rate 4.75-12.2 kbit/s
Next section will give a further study of the enhanced full
rate (EFR) coder.
[attachment=23675]
Introduction
2.1 Background
This master thesis work was done as part of the programme
for Computer Science and Engineering at Linköping University.
The employer was Enea Epact, but the real customer was
Ericsson Radio Network Center.
With the introduction of GPRS in the GSM network, the
need for verification of the traffic channels increase. For
example, a useful scenario for this new tool would be when
there are a lot of GPRS traffic going on through the BSC,
then a voice call is being set up and GPRS traffic channels
has to be pre-empted. It would also be interesting to use the
system for generating background load on the traffic channels,
while performing other automatic tests.
The thesis consists of two parts. The first part consists of an
analysis to find out the needs and to see how it is possible to
implement the system. The second part involves implementing
part of the system.
Assignment
4 Design and implementation of a test tool for the GSM traffic channel
2.2 Assignment
The customer has a product called Test and Simulation
Solutions (TSS), used for testing nodes in the Global System
for Mobile Communications (GSM) and General Packet
Radio Systems (GPRS). The TSS consists of special hardware
as well as software and is used to simulate traffic in
mobile cell phone networks.
Within TSS there is a problem when it comes to circuit
switched networks like the GSM. The test tool is only able
to set up the connections using the signalling channels, the
physical channels used for traffic are not verified with this
tool. When it comes to packet switched networks, like the
GPRS, this problem has been handled.
The first assignment was to see how the product TSS could
be enhanced with a new tool allowing for verification of the
GSM traffic channels. This includes how to insert traffic
(i.e. simulated speech) into the physical channel, how to
generate it, and how to control the tool in a way that is consistent
with the rest of the product. To do this, several subareas
have to be studied. The GSM network itself has to be
studied in order to find out scenarios where this new tool
can become useful. What standards are there that the new
tool must comply to in order to connect to the GSM network?
And what about the user interface part, what are the
common standard the tool must comply to in order to be easily
integrated with the TSS?
The second assignment for this master thesis was to implement
part of the system proposed. In this case the choosen
part is the enhanced full rate speech codec, which is the
most commonly used codec of cellular phone systems. This
part will involve examining the speech compression and
decompression algorithms to be able to implement them on
the target hardware.
Limitations
Design and implementation of a test tool for the GSM traffic channel 5
2.3 Limitations
Only the speech encoding and decoding functions will be
implemented since this is an area where there is not enough
knowledge at the employer or the customer. This is basically
one function which can be developed detached from the rest
of the test system and easily integrated later.
The full test system, with traffic generator, user interface
etc., will not be implemented within this master thesis due to
limitations in time. Also this area is more well known and
the both the employer and the customer has a lot of experience
in this area.
2.4 Purpose of this report
The purpose of this report is to summarize the work performed
and to document the results achieved. The situation
before is described as well as the gains from implementing
and using the proposed test system.
2.5 Reading instructions
The report is intended to be readable by most senior students
at a Master of Science programme and teachers at those programmes.
However it will also include details about the system
useful for professionals working in the area of
telecommunication using or further developing the system.
Professionals implementing the system should consider
reading [1] and [2].
Chapter 2 gives a brief introduction to the report and defines
the assignment.
Chapter 3 deals with the theory behind the GSM network
and is useful for those not working within the telecommunication
area.
Reading instructions
6 Design and implementation of a test tool for the GSM traffic channel
Chapter 4 gives an introduction to the theory behind speech
coding and the parts of a full featured speech coder.
Chapter 5 gives a deeper analysis of the need and usage of a
test tool.
Chapter 6 describes the design to be used for the proposed
system.
Chapter 7 discusses issues concerning test and implementation.
Problems, improvements and errors are discussed.
Chapter 8 summarizes the thesis and discusses drawn conclusions.
Design and implementation of a test tool for the GSM traffic channel 7
GSM basics
3.1 Telecommunication overview
The following sections will introduce terminology from the
area of mobile telecommunication. The intention is to give
an overview and to define the concepts of mobile telecommunication
and its notion. Details can be found in [3] and
[4].
3.2 Networking elements
The GSM network consists of a hierarchy of elements as can
be seen in figure 3.1. In this particular network there are two
Mobile Switching Centres (MSC), one Gateway MSC
(GMSC) and three different access networks. There are also
some databases to store the information needed in the network.
This public land mobile network (PLMN) is also connected
to the public switched telephone network (PSTN)
and one other PLMN.
The mobile station (MS) can be any type of end user equipment
like a cellular phone, laptop or fax machine with radio
modem.
Networking elements
8 Design and implementation of a test tool for the GSM traffic channel
FIGURE 3.1. GSM network hierarchy.
The Base Transceiver Station (BTS) contains equipment for
the physical radio transmitting and receiving, signal measurements,
encryption and communication with the base station
controller (BSC).
Base station controllers serve as concentrator nodes for several
BTS. It is responsible for allocating radio channels used
by the BTS and to connect speech and signalling channels to
the MSC. It also handles transfers, handovers, of mobile stations
between different BTS. This occurs for example, when
the mobile station is being moved within a limited area, for
example in a city.
The mobile switching centre corresponds to the local station
in a public switched telephone network, though it does not
have any hard wired subscribers. It is responsible for mov-
PSTN GMSC
O ther
PLMN
AUC EIR
HLR
VLR VLR
BSC
MSC MSC
BTS
BTS
BTS MS
A
Abis
Um
Interfaces
Design and implementation of a test tool for the GSM traffic channel 9
ing mobile stations between different BTS and between different
MSC. Transfer of mobile stations between BTS
occurs for example when the end user is travelling longer
distances.
The GMSC is similar to the MSC. It serves as an interface
between the network and other networks, for example the
PSTN or other PLMN. It is responsible for charging
between different network operators. There may be several
GMSC in a network, and they may be integrated with the
MSC.
There are also databases connected to the switching network
elements. The home location register (HLR) contains information
about the end user and under which MSC it currently
is registered. At each MSC there is a visitor location register
(VLR) which contains information about all mobile stations
that are currently in that MSC service area.
Authentication centre (AUC) contains information used for
encryption and authorization.
Equipment identity register (EIR) stores information about
the MS identity. It is used to control that a mobile station is
not blocked.
3.3 Interfaces
The different interfaces between the network elements are
named in figure 3.1. Starting from the MS side, first comes
the Um interface. This is the radio interface between a
mobile station and the BTS [5]. Each logical speech channel
requires 16 kbit/s data rate. With error correction codes,
burst management and encryption the used data rate is 33.8
kbit/s per channel.
The interface between the BTS and the BSC is named Abis.
It uses standard T1 or E1 cables for interconnection. Four 16
Inside the equipment
10 Design and implementation of a test tool for the GSM traffic channel
kbit/s speech channels multiplexed into one 64 kbit/s channel
corresponds to one timeslot in the E1 or T1 link between
the BTS and BSC. The job to transform the speech information
to the correct format is done by the transcoding and rate
adaption unit (TRAU). This is where the speech encoding
and decoding functionality are situated. The Abis interface
is defined in [6].
The A interface between BSC and MSC consists of 64 kbit/s
PCM links on a E1 or T1 interconnection. The speech is
here coded with a-law or μ-law PCM depending on if a
European or American/Pacific system is used. More information
about the A interface is found in [7].
3.4 Inside the equipment
The basic structure inside the switching network elements is
shown in figure 3.2. The main components in the BSC, and
MSC are a number of line interfaces (LIF), i.e. the physical
connections to the other elements, a switching unit and a
control system. There is also some kind of terminal equipment
(TERM) for operation and maintenance.
Inside the BSC there is also a pool of cpu resources for
speech transcoding and rate adaption in the TRAU module,
available for use when connecting traffic channels between
the MSC and BSC.
The BTS consists of a number of transceivers (TRX), which
are multiplexed into the radio interface. The transceivers are
accessed from the BSC via a transceiver radio interface
switching traffic channels from the BSC to the correct transceiver
and vice versa.
Inside the equipment
Design and implementation of a test tool for the GSM traffic channel 11
FIGURE 3.2. Inside the switching elements.
TRX
TRI
CONTROL
TRX
LIF
LIF
TERM
TRAU
CONTROL
M
U
X
LIF
LIF
TERM
CONTROL
BSC
GMSC/MSC
BTS
Other networks
Inside the equipment
12 Design and implementation of a test tool for the GSM traffic channel
Design and implementation of a test tool for the GSM traffic channel 13
Speech coding
4.1 General
A general communication system can be split into a couple
of modules, as in figure 4.1. The transmitter is modelled by
a source encoder, channel encoder and modulator. The
receiver can be modelled by a demodulator, channel decoder
and source decoder as seen in . Between the source and destination
side, there is the transmission media, called a channel.
Since most channels not are perfect, the channels will
add some noise to the signal.
The following sections will concentrate on the features of
the source encoder and decoder.
4.2 Human speech model
Since we are interested in compression of human speech , it
is interesting to study how speech is produced.
Sounds are produced in two different ways. There are
voiced sounds, all vowels for example, that are produced by
the vibrating vocal cords at the glottis. There are also
unvoiced sounds, like ‘p’, ‘t’, and ‘f’, which are the result of
Human speech model
14 Design and implementation of a test tool for the GSM traffic channel
FIGURE 4.1. General communication system.
air flow through a construction in the vocal tract. Note that
consonants can also be voiced sounds like ‘b’, ‘d’ and ‘j’.
Also some sounds are the result of a combination of these.
The excitation signal for these sounds is the vibration of the
vocal cords. The frequency of the vibration does not vary
greatly in time due to the physiology of the glottis, and can
therefore be modelled using a slowly varying model.
The vocal tract is a mechanical system, which means that
the vocal gestures are relatively slow. It can be modelled by
a parametric filter, where the parameters will vary with time.
The bandwidth for these filter parameters are lower than for
the speech signal itself. This is called the short time stationarity
of the vocal tract filter.
The human ear is also to be considered in the models. It is
able to perceive signals between 20 Hz - 20 kHz. However,
it is most responsive to frequencies between 200 and 5600
Hz which usually carries the most important information
when it comes to human speech. Most telephone systems
has a bandwidth limitation to the range of 300 to 3400 Hz,
which will be possible without losing to much of the quality.
Another important aspect is masking, which means that one
sound can be obscured by the presence of another. Masking
Information
source
Source
encoder
Channel
encoder Modulator
Channel
Demodulator
Channel
decoder
Source
decoder
Information
destination
Noise
Different codecs
Design and implementation of a test tool for the GSM traffic channel 15
can be done either as temporal masking or as frequency or
spectral masking. Temporal masking means that signals
with energy in frequencies close to each other can mask
each other, if the time delay between the signals are short.
Frequency masking means that a lower frequency can be
masked by a simultaneous higher frequency. This can be
used to distribute quantization noise to the masked frequency.
4.3 Different codecs
There are different types of speech codecs. The two main
types of coders are waveform coders and vocoders (voice
coders). There are also hybrid coders which combine the
properties of those coders. Some properties of these codecs
are shown in figure 4.2.
FIGURE 4.2. Comparison of compression ratio and
speech quality of different codec categories.
Waveform coding is the process of describing the signal’s
amplitude curve with a number of discrete values. This is
basically done in three steps - sampling, quantization and
coding. The quality of the signal will in general be very
good but requires more bandwidth. Waveform coders tend
to be more robust against unexpected input, like music, than
vocoders.
Low
Excellent
Good
Average
Hybrid coders
Waveform coders
Vocoders
Speech quality
b itrate
2 4 8 16 32 64 (kbit/s)
Enhanced full rate codec
16 Design and implementation of a test tool for the GSM traffic channel
Vocoders use a totally different approach. They try to figure
out which parameters were used to create the signal, given
that it passed through a known filter model. The filter model
in mind is of course a model of the human speech organs.
Vocoders lose a bit of quality of the transferred signal, but
will on the other hand use less bandwidth. The process of
transmitting speech parameters instead of the speech can be
compared to transmitting notes instead of the music itself.
Because of the two contradictionary demands of low bandwidth
usage and high signal quality in cellular phone systems,
hybrid coders are used. The coders use a combination
of parameterizing and waveform coding to achieve this. The
bitrate can be reduced to levels under the 64 kbit/s PCM
coding with good signal quality.
There are several possible speech coders for usage in cellular
phone systems (See also [8]):
• Full rate 13 kbit/s
• Enhanced full rate 12.2 kbit/s
• Half rate 5.60 kbit/s
• Adaptive multi rate 4.75-12.2 kbit/s
Next section will give a further study of the enhanced full
rate (EFR) coder.