GSM Phy Part-2

Cellular Mobile Systems and Services (TCOM1010) 2009-May
Day04_GSM Radio Part 2_Signal Processing.doc Monzur Kabir, Ph.D., P.Eng Page 1 of 13
GSM Radio Part 2
(Speech Processing, Discontinuous Transmission, Timing Advance and Power Control)

1 GSM SPEECH PROCESSING........................................................................................................................................ 2
1.1 RPE-LTP (Regular Pulse Excitation Long-Term Prediction) ....................................................................................... 3
1.2 Channel coding for the GSM data TCH channels ......................................................................................................... 5
1.3 Interleaving ................................................................................................................................................................... 7
1.4 Burst Assembling .......................................................................................................................................................... 7
1.5 Ciphering....................................................................................................................................................................... 7
1.6 Multi-framing................................................................................................................................................................ 8
2 TIMING ADVANCE......................................................................................................................................................... 9
3 POWER CONTROL AND SAVING TECHNIQUES ................................................................................................. 10
4 DISCONTINUOUS TRANSMISSION AND RECEPTION........................................................................................ 12
5 FREQUENCY HOPPING TECHNIQUE IN GSM WHY AND HOW................................................................... 12

1 GSM Speech Processing

GSM supports three speech processing formats:

Full-rate (TCH-FR, 13 kbps) this is the standard format. The TCH-FR is based on RPE-LTP (Regular Pulse
Excitation - Long Term Prediction), which is based on LPC (Linear Predictive Coding).

Half-rate (TCH-HR, 5.6 kbps) Lower bit-rate that makes the capacity doubled and also lowers the processing
power consumption (up to 30%), but have lower voice quality (when compared with TCH-FR). This is an option
where capacity is bigger issue than quality (example: a big rally of thousands of people). The half-rate essentially
accommodate two voice connections per traffic channel. GSM-HR is based on Vector Sum Excited Linear
Prediction (VSELP).

Enhanced-rate (TCH-ER, 12.2 kbps) Bit-rate is little lower (not a significant improvement) but increases the
quality of voice (as good as telephony voice). However, it consumes 5% more processing power. This is a relatively
new addition, and it uses better coding technique. The TCH-ER is based on AMR (Adaptive Multi Rate) coding.

The following figure depicts the speech processing stages (example is based on TCH-FR) and the table calculates the bit-rate.

Ref: Wireless Telecommunications Systems and Networks, By Mullett, Thomson Publisher

Analog Speech to Digital Voice (TCH-FR) Transmission over the air-interface
20 ms slice of voice is digitized with RPE-LTP coder which generates 260 bits (13kbps)
Digitized 260 bits undergo channel-coding which generates 456 bits (22.8 kbps)
456 bits undergoes burst formation
One burst contains 2x57 data bits. That is, 456 bits requires 4 bursts

4 burst = (2x57 + 42.25) x 4 = 625 bits
Eight such voice channels are multiplexed on a single frequency,
which means 625x8 = 5000 bits (250 kbps)
Every 24 bursts have overhead of 2 bursts to form 26-burst multi-frame. This is equivalent to
5000 x 26/24 = 5416.67 bits
5416.67 bits must be transmitted by 20 ms ====> 270.8 kbps

1.1 RPE-LTP (Regular Pulse Excitation Long-Term Prediction)
Ref: http://www.otolith.com/otolith/olt/lpc.html

Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods
for encoding good quality speech at a low bit rate. It provides extremely accurate estimates of speech parameters, and is
relatively efficient for computation.

LPC starts with the assumption that the speech signal is produced by a buzzer at the end of a tube. The glottis (the space
between the vocal cords) produces the buzz, which is characterized by its intensity (loudness) and frequency (pitch). The
vocal tract (the throat and mouth) forms the tube, which is characterized by its resonances, which are called formants.

LPC analyzes the speech signal by estimating the formants, removing their effects from the speech signal, and estimating the
intensity and frequency of the remaining buzz. The process of removing the formants is called inverse filtering, and the
remaining signal is called the residue. The Residue and formant are the two part of the LPC coded voice.

LPC synthesizes the speech signal by reversing the process: use the residue to create a source signal, use the formants to
create a filter (which represents the tube), and run the source through the filter, resulting in speech.

Because speech signals vary with time, this process is done on short chunks of the speech signal, which are called frames.
Usually 30 to 50 frames per second (33 to 20 ms per frame) give intelligible speech with good compression.

The Regular Pulse Excitation (RPE) stage involves reducing the 40 long-term residual samples down to four sets of 13-bit
sub-sequences through a combination of interleaving and sub-sampling. The optimum sub-sequence is determined as having
the least error, and is coded using APCM (adaptive PCM) into 45 bits.

The long-term prediction (LTP) filter models the fine harmonics of the speech using a combination of current and previous
sub-frames. The gain and lag (delay) parameters for the LTP-filter are determined by cross-correlating the current sub-frame
with previous residual sub-frames.

http://cs.haifa.ac.il/~nimrod/Compression/Speech/S4ABYS2004.pdf

Ref. for the following text: http://www.commsdesign.com/article/printableArticle.jhtml?articleID=16501605

As the mathematical model for speech generation in a full-rate codec shows a gradual decay in power for an increase in
frequency, the samples are fed through a pre-emphasis filter that enhances the higher frequencies, resulting in better
transmission efficiency. An equivalent de-emphasis filter at the remote end restores the sound.

The short-term analysis (linear prediction) performs autocorrelation and Schur recursion on the input signal to determine the
filter ("reflection") coefficients. The reflection coefficients, which are transmitted over the air as eight parameters totalling 36
bits of information, are converted into log area ratios (LARs) as they offer more favourable companding characteristics. The
reflection coefficients are then used to apply short term filtering to the input signal, resulting in 160 samples of residual
signal.

The residual signal from the short-term filtering is segmented into four sub-frames of 40 samples each. The long-term
prediction (LTP) filter models the fine harmonics of the speech using a combination of current and previous sub-frames. The
gain and lag (delay) parameters for the LTP-filter are determined by cross-correlating the current sub-frame with previous
residual sub-frames. The peak of the cross-correlation determines the signal lag (LTP-lag) and the gain (LTP-gain) is
calculated by normalising the cross-correlation coefficients. The parameters are applied to the long-term filter, and a
prediction of the current short-term residual is made. The error between the estimate and the real short-term residual signal
the long-term residual signalis applied to the RPE analysis, which performs the data compression.

The Regular Pulse Excitation (RPE) stage involves reducing the 40 long-term residual samples down to four sets of 13-bit
sub-sequences through a combination of interleaving and sub-sampling. The optimum sub-sequence is determined as having
the least error, and is coded using APCM (adaptive PCM) into 45 bits.

The resulting signal is fed back through an RPE decoder and mixed with the short-term residual estimate in order to source
the long-term analysis filter for the next frame, thereby completing the feedback loop (see the table below)

Output Parameters from the Full Rate Codec

http://cs.haifa.ac.il/~nimrod/Compression/Speech/S4ABYS2004.pdf

1.2 Channel coding for the GSM data TCH channels

Channel coding (also called error control coding) is a process of adding redundancy bits (also called error
detection/correction bits) to the original information bits in order to detect/correct errors that might occur during the
transmission. Usually the transmitter side runs an algorithm on the original information bits in order to compute the
redundancy bits. The transmitter sends the redundancy bits along with the original information bits to the receiver. The
receiver runs the peer algorithm on the received bits in order to detect the error and correct, if possible (some code has limited
capability to correct the error). Usually a longer length of the redundancy bits provides better error detection/correction
capability, but adds more overhead.

The channel coding has two major types: block code and convolution code.

A process of block coding receives an input block of k bits, process these block with an error control algorithm to
produce n bits output (n > k). The additional (n k) bits are the error control bits. This is called (n, k) block
code.

A process of convolution coding receives a stream of bits (k bits per clock), applies the error control coding
algorithm on K bits at a time (see the figure below) to produce n bits output per clock (n > k).
This is called (R, K) convolution code, where R = k/n. The following figure depicts a convolution coder with
K = 5, k = 1 and n =2.

http://www.azizi.ca/gsm/coding/index.html

The main differences between above two methods are:
The block code does not relate one block to another in its coding. Convolution coding is a Sliding Window type
coding and every bit participates on a number of error control operations.
A block-code accumulates a large block of data first and then starts coding, whereas a convolution code accepts the
data as a stream (more real-time)
Block code the suitable for non-real time and non-stream type of information (such as web page and email) and
mostly used for ARQ (Automatic Repeat reQuest) scheme which is an error detection followed by retransmission
method. The convolution code is more suitable for real-time data stream and mostly used for FEC (Forward Error
Correction) schemes (and hence more overhead)

The GSM TCH-FR scheme uses both of the above coding schemes. Recall that the speech codec produces a 260 bit block
for every 20 ms speech segment. Not all of the bits have equal significance some of them are more significant than the
others. The bits are thus divided into three classes:

o Class Ia of 50 bits - most sensitive to bit errors
Class Ia bits have a 3 bit Cyclic Redundancy Code added for error detection. If an error is detected, the frame is
judged too damaged to be comprehensible and it is discarded. It is replaced by a slightly attenuated version of the
previous correctly received frame in case of voice communication.

o Class Ib of 132 bits - moderately sensitive to bit errors
Above 53 bits, together with this 132 bits and a 4 bit tail sequence of zero bits (a total of 189 bits), are input into a
1/2 rate convolution encoder of constraint length 4 (K = 5). Each input bit is encoded as two output bits, based on a
combination of the previous 4 input bits. The convolution encoder thus outputs 378 bits.

o Class II of 78 bits - least sensitive to bit errors
These 78 bits are not coded and simply added to the previous 378 bits forming a bit sequence of total 456 bits per 20
ms voice segment (22.8 kbps).

1.3 Interleaving
Interleaving is a process of dispersing the bits of a data burst over multiple bursts in a systematic way (see the figure below).

http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/features.html

Benefit of this technique: when a data-burst is lost (due to burst error in the radio interface) it does not mean a 100% loss of a
single burst rather a partial loss of many bursts. In most of the cases, where the interleaving technique is used, has error
control scheme in place. A partial error (which means error of few bits only) is often error detectable/correctable. The result
is: no loss of data due to a burst error. Thus the interleaving decreases the possibility of losing whole bursts during the
transmission, by dispersing the errors. Being the errors less concentrated, it is then easier to correct them.

A burst in GSM transmits two blocks of 57 data bits each. Therefore the 456 bits corresponding to the output of the channel
coder fit into four bursts (4*114 = 456). The 456 bits are divided into eight blocks of 57 bits. The first block of 57 bits
contains the bit numbers (0, 8, 16, .....448), the second one the bit numbers (1, 9, 17, .....449), etc. The last block of 57 bits
will then contain the bit numbers (7, 15, .....455).

1.4 Burst Assembling
Recall that the interleaving process produces eight 57-bit blocks from 456 bits of 20 ms voice. Two such blocks are fitted into
a normal burst format (see the figure below). Thus four such burst carries all eight 57-bit blocks (that is, 20 ms voice.

1.5 Ciphering
Ciphering is used to protect signaling and user data from intruder. First of all, a ciphering key (K
C
) is computed using the
algorithm A8 stored on the SIM card, the subscriber key (K
I
, also stored in the SIM card) and a random number (RAND)
delivered by the network (this random number is the same as the one used for the authentication procedure). Secondly, a 114
bit sequence is produced using the ciphering key, an algorithm called A5 and the TDMA frame number (F
n
) number provided
by the network. This bit sequence is then XORed (TRUE if only one of the inputs is TRUE) with the two 57 bit blocks of
data included in a normal burst.


http://www.gsm-security.net/gsm-security-papers.shtml

1.6 Multi-framing
Four consecutive burst carries a 20 ms voice-segment (456 bits). However, there is a burst after 3 sets of 20 ms voice which
is used for control signal (SACCH channel) (see the 26-Multiframe format below). That is, there are 2 overhead burst per six
voice segments.

Question: What is the required bit rate for a GSM frequency channel over the radio interface in order to carry voice in a real-
time fashion?

2 Timing Advance
(http://www.mobileshop.org/howitworks/engmode.htm for more)

Suppose that a user is connected through Slot# 5 (see the figure below). The user is, at this moment very close to the base
station antenna. The user of the preceding slot (Slot# 4) is, on the other hand, is very far from the antenna. The burst of Slot#
4 will suffer a longer delay than the burst of Slot# 5 to reach the base station. If the clocks of those two mobile stations are
synchronized with the BTS clock then there will be a delay difference. This difference may cause the Slot# 4 burst crossing
its time boundary and overlapping Slot#5. To avoid this problem the distant user may require a timing advance to make
over the delay.

It is a roundtrip delay:
Let us assume the propagation delay from BTS to distant user (the Slot-4 user) be t
BTS sends time-information at Time t
1
and it arrived at Time (t
1
+ t).
The distant user sends a frame at Time (t
1
+ t) which arrives to BTS at Time (t
1
+ 2t)
The overall delay is 2t though the user is fully in sync.
=> Delay is equal to roundtrip delay for the distance

Quantify the roundtrip delay:
The speed of light in vacuum = 3 x 10
8
m/s
=> 1 sec delay for 3 x 10
8
m => 1 s delay for 300 meter
=> 1 s delay for 150 meter (roundtrip)
GSM radio interface has bit-rate of 270.8 kbps. That is, the bit-duration is 3.69 s.
=> That is, one bit delay means 1105 m distance (one way) or about 553 m (roundtrip)

GSM Timing Advance (TA):
64 steps (Step 0 to 63)

Timing Advance 0 1 2 3 4 5 ....... 63
Distance to BTS < 550 m
550 m-
1100 m
1100 m-
1650 m
1650 m-
2200 m
2200 m-
2750 m
2750 m-
3300 m
....... 35 km
Ref: http://www.tele-servizi.com/janus/engfield2.html
During communications between a mobile station and the base station the network checks the training sequence
arrival time (the sequence is in the data burst) and computes the delay and hence the distance. Based on the
estimated distance the network sends appropriate TA step for the mobile station. For example, if the TA is 4then the
mobile station will set its clock time advanced by 5 bit-times with respect to the BTS clock.

Why up to Step-63 or 35 km roundtrip distance?

First user transmission (for service request with RACH channel using Access Burst)

The first burst from a mobile station (after switch-on) is always without any timing advance since the network does
not have any information about how far the mobile station is and network did not start interacting with the mobile
station at this stage. The access burst shows that the guard-period of access burst is 68.25 bits and hence it can
tolerate a distance up to 553 *68.5 = 37.8 km distance between the BTS and the mobile station. A conservative
number is 35 km. A distance more than that means that the access burst is overlapping the next burst.

3 Power Control and Saving Techniques
There are five classes of mobile stations defined, according to their peak transmitter power, rated at 20, 8, 5, 2, and 0.8 watts.
To minimize co-channel interference and to conserve power, both the mobiles and the Base Transceiver Stations operate at
the lowest power level that will maintain an acceptable signal quality. Power levels can be stepped up or down in steps of 2
dB from the peak power for the class.

The following table defines GSM power classes:

Power Class Maximum Power of a MS
Watt (dBm)

Maximum Power of a BTS
Watt (dBm)
1 20 (43) 320 (55)
2 8 (39) 160 (52)
3 5 (37) 80 (49)
4 2 (33) 40 (46)
5 0.8 (29) 20 (43)
6 10 (40)
7 5 (37)
8 2.5 (34)
http://www.analytek.co.uk/files/GSM_Quick_Ref.pdf
Note: Typically a handheld mobile stations power class is Class 4 (2 W or 33 dBm

The mobile station measures the signal strength or signal quality (based on the Bit Error Rate), and passes the information to
the Base Station Controller, which ultimately decides if and when the power level should be changed. Power control should
be handled carefully, since there is the possibility of instability. This arises from having mobiles in co-channel cells
alternatively increase their power in response to increased co-channel interference caused by the other mobile increasing its
power. Depending on the class and GSM band the MS control level has minimum power level as well. For example, GSM-
900 Class-4 mobile station has maximum power of 33 dBm (2 W) with decrement step 2 dBm down to 13 dBm (20 mW).


http://www.analytek.co.uk/files/GSM_Quick_Ref.pdf

The mobile station receiver sensitivity is the minimum power of the received signal in order to recover the information at
least with minimum quality requirement (defined in terms of Bit Error Rate or BER). The following table provides some
examples of receiver sensitivity.

4 Discontinuous Transmission and Reception
The function of the discontinuous transmission (DTX) is to suspend the radio transmission during the silence periods. This
can become quite interesting if we take into consideration the fact that a person speaks less than 40 or 50 percent during a
conversation. The DTX can help reduce interference between different cells and to increase the capacity of the system. It also
extends the life of a mobile's battery. The DTX function is performed thanks to two main features:
The Voice Activity Detection (VAD): VAD has to determine whether the sound represents speech or noise. If the
signal is considered noise, the system truncates that portion of signal and saves power and bandwidth.
The comfort noise: an inconvenient of the DTX function is that when the signal is considered as noise, the
transmitter is turned off and therefore, a total silence is heard at the receiver. This can be very annoying to the user
at the reception because it seems that the connection is dead. In order to overcome this problem, the receiver creates
a minimum of background noise called comfort noise. The comfort noise eliminates the impression that the
connection is dead.
However, DTX may have inaccuracy in deciding if it is a noise. The inaccuracy may either truncate voice (when voice is
wrongfully detected as noise) or degrade DTX efficiency (when noise is detected as voice).

The discontinuous feature can provide the following advantages:
Minimizing co-channel interference
Conserves power
Produces less traffic per Voice over IP (VOIP) call through a GPRS system

Another method used to conserve power at the mobile station is discontinuous reception (DRX). The paging channel, used by
the base station to signal an incoming call, is structured so that the mobile station knows when it needs to check for a paging
signal. In the time between paging signals, the mobile can go into sleep mode, when almost no power is used.

5 Frequency Hopping Technique in GSM Why and How
Ref: https://styx.uwaterloo.ca/~jscouria/GSM/gsmreport.html

Frequency hopping is the technique of improving the signal to noise ratio in a link by adding frequency diversity. A mobile
station may severely suffer from co-channel interference (especially at the cell boundary) and multipath fading effect, which
is often frequency dependent. Slow Frequency Hopping helps alleviate the problems. When a call uses this technique its TCH
channel hops from one frequency to another frequency according to a predefined (or dynamically defined) frequency-
sequence. It repeats the sequence again and again. Since this technique uses multiple frequencies instead of just one it in
effect randomizes the effect of co-channel interference and fading. The following figure illustrates frequency hopping
concept.

http://www.wicomtech.com/conferences/GSM-NET.pdf

The mobile station already has to be frequency agile, meaning it can move between transmit, receive, and monitor time-slots
within one TDMA frame, which may be on different frequencies. GSM makes use of this inherent frequency agility to
implement slow frequency hopping, where the mobile and BTS transmit each TDMA frame on a different carrier frequency.

The following entities are involved in computation of the hopping sequence:

The frequency hopping algorithm is broadcasted on the Broadcast Control Channel (BCCH).

The Cell Allocation (CA) table lists all the channels available to the cell. The network broadcasts the list in the
BCCH channel. The default CA Table contains all the ARFCNs of GSM system.

A Mobile Allocation (MA) Table is a list of the ARFCNs present in the CA table that share the same frequency band
(such as GSM-900). There is a separate MA Table for each frequency band. Sixteen is the maximum number of
entries for an MA Table. This list contains the frequencies which can be used in the hopping sequence.
The hopping sequence starts from an ARFCN that is included in the MA list. The entry of the MA Table at which
the hopping sequence begins is called the Mobile Allocation Index Offset (MAIO). Note that an MAIO of zero
corresponds to the first entry of the MA Table.
There is another parameter, called Hopping Sequence Number (HSN), is used to compute the hopping sequence.
The HSN can be any number from 0 to 63. An HSN of zero corresponds to the cyclic hopping sequence (default
sequence), and values 1 through 63 correspond to the pseudo random patterns.

The base station commands the mobile station to activate frequency hopping as the mobile station moves toward the edge of
a cell or into an area of high interference or fading. At that time the network provides the mobile station the hopping
sequence IDs (the MAIO and HSN) as shown in the message below. The mobile station computes the hopping sequence
using these IDs, the MA list and the hopping algorithm (obtained from BCCH channel). In a GSM/GPRS/EGPRS network,
frequency hopping is specified in individual cells based on the number of frequencies offered by a specific cell.

Ref: Wireless Communications Systems and Networks, By Mullett, Thomson Publisher

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