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The FDMA part involves the division
by frequency of the total 25 MHz bandwidth into 124 carrier frequencies of 200
kHz bandwidth. One or more carrier frequencies are then assigned
to each base station. Each of these carrier frequencies is then
divided in time, using a TDMA scheme, into eight time slots. One
time slot is used for transmission by the mobile and one for reception.
They are separated in time so that the mobile unit does not receive and
transmit at the same time, a fact that simplifies the electronics.
In the rest of this section, the procedure involved in digitally transmitting
a voice signal in a GSM network is examined, along with some of the features,
such as discontinuous transmission and reception, used to improve voice quality,
reduce the mobile unit's power consumption, and increase the overall capacity of
the network. ,------------------------------------------------------------------------,
| |
| ,---,---------------,---,-----------,---,----------------,---,------, |
| | 3 | 57 | 1 | 26 | 1 | 57 | 3 | 8.25 | |
| `---'---------------'---'-----------`---'----------------`---`------' |
| Tail Data bits Training Data bits Tail Guard |
| bits sequence bits bits |
| |
`------------------------------------------------------------------------'
FIGURE 2
1 Channel structure
The structure of the most common timeslot burst is shown in Figure 2.
A total of 156.25 bits is transmitted in 0.577 milliseconds, giving a gross
bit rate of 270.833 kbps. There are three other types of burst
structure for frame and carrier synchronization and frequency correction.
The 26bit training sequence is used for equalization, as described below.
The 8.25 bit guard time allows for some propagation time delay in the
arrival of bursts.
Each group of eight time slots is called a TDMA frame, which is transmitted
every 4.615 ms. TDMA frames are further grouped into multiframes
to carry control signals. There are two types of multiframe,
containing 26 or 51 TDMA frames. The 26frame multiframe
contains 24 Traffic Channels (TCH) and two Slow Associated Control Channels (SACCH)
which supervise each call in progress. The SACCH in frame 12
contains eight channels, one for each of the eight connections carried by the
TCHs. The SACCH in frame 25 is not currently used, but will carry
eight additional SACCH channels when halfrate traffic is implemented.
A Fast Associated Control Channel (FACCH) works by stealing slots from a
traffic channel to transmit power control and handoversignalling messages.
The channel stealing is done by setting one of the control bits in the time
slot burst.
In addition to the Associated Control Channels, there are several other
control channels which (except for the Standalone Dedicated Control Channel)
are implemented in time slot 0 of specified TDMA frames in a 51frame
multiframe, implemented on a nonhopping carrier frequency in each cell.
The control channels include:
- Broadcast Control Channel (BCCH): Continually broadcasts, on the downlink,
information including base station identity, frequency allocations, and
frequencyhopping sequences.
- Standalone Dedicated Control Channel (SDCCH): Used for registration,
authentication, call setup, and location updating. Implemented
on a time slot, together with its SACCH, selected by the system operator.
- Common Control Channel (CCCH): Comprised of three control channels used
during call origination and call paging.
- Random Access Channel (RACH): A slotted Aloha channel to request
access to the network
- Paging Channel (PCH): Used to alert the mobile station of incoming
call.
- Access Grant Channel (AGCH): Used to allocate an SDCCH to a mobile for
signalling, following a request on the RACH.
2 Speech coding
GSM is a digital system, so speech signals, inherently analog, have to be
digitized. The method employed by ISDN, and by current telephone
systems for multiplexing voice lines over high speed trunks and optical fiber
lines, is Pulse Coded Modulation (PCM). The output stream from
PCM is 64 kbps, too high a rate to be feasible over a radio link. The
64 kbps signal contains much redundancy, although it is simple to implement.
The GSM group studied several voice coding algorithms on the basis of
subjective speech quality and complexity (which is related to cost, processing
delay, and power consumption once implemented) before arriving at the choice of
a Regular Pulse Excited - Linear Predictive Coder (RPELPC) with a Long Term
Predictor loop. Basically, information from previous samples,
which does not change very quickly, is used to predict the current sample.
The coefficients of the linear combination of the previous samples, plus an
encoded form of the residual, the difference between the predicted and actual
sample, represent the signal. Speech is divided into 20
millisecond samples, each of which is encoded as 260 bits, giving a total bit
rate of 13 kbps.
3 Channel coding and modulation
Due to natural or manmade electromagnetic interference, the encoded speech or
data transmitted over the radio interface must be protected as much as is
practical. The GSM system uses convolutional encoding and block
interleaving to achieve this protection. The exact algorithms
used differ for speech and for different data rates. The method
used for speech blocks will be described below.
Recall that the speech codec produces a 260 bit block for every 20 ms speech
sample. From subjective testing, it was found that some bits of
this block were more important for perceived speech quality than others.
The bits are thus divided into three classes:
Class Ia 50 bits - most sensitive to bit errors
Class Ib 132 bits - moderately sensitive to bit errors
Class II 78 bits - least 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. These
53 bits, together with the 132 Class Ib bits and a 4 bit tail sequence (a total
of 189 bits), are input into a 1/2 rate convolutional encoder of constraint
length 4. Each input bit is encoded as two output bits, based on
a combination of the previous 4 input bits. The convolutional
encoder thus outputs 378 bits, to which are added the 78 remaining Class II
bits, which are unprotected. Thus every 20 ms speech sample is
encoded as 456 bits, giving a bit rate of 22.8 kbps.
To further protect against the burst errors common to the radio interface,
each sample is diagonally interleaved. The 456 bits output by the
convolutional encoder are divided into 8 blocks of 57 bits, and these blocks are
transmitted in eight consecutive timeslot bursts. Since each
timeslot burst can carry two 57 bit blocks, each burst carries traffic from
two different speech samples.
Recall that each timeslot burst is transmitted at a gross bit rate of
270.833 kbps. This digital signal is modulated onto the analog
carrier frequency, which has a bandwidth of 200 kHz, using Gaussianfiltered
Minimum Shift Keying (GMSK). GMSK was selected over other
modulation schemes as a compromise between spectral efficiency, complexity of
the transmitter, and limited spurious emissions. The complexity
of the transmitter is related to power consumption, which should be minimized
for the mobile station. The spurious radio emissions, outside of
the allotted bandwidth, must be strictly controlled so as to limit adjacent
channel interference, and allow for the coexistence of GSM and the older
analog systems (at least for the time being).
4 Multipath equalization
At the 900 MHz range, radio waves bounce off everything - buildings, hills,
cars, airplanes, etc. Thus many reflected signals, each with a
different phase, can reach an antenna. Equalization is used to
extract the desired signal from the unwanted reflections. Equalization works by
finding out how a known transmitted signal is modified by multipath fading, and
constructing an inverse filter to extract the rest of the desired signal.
This known signal is the 26bit training sequence transmitted in the
middle of every time slot burst. The actual implementation of the
equalizer is not specified in the GSM specifications.
5 Frequency hopping
The mobile station already has to be frequency agile, meaning it can move
between a transmit, receive, and monitor time slot 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
frequency hopping algorithm is broadcast on the Broadcast Control Channel.
Since multipath fading is (mildly) dependent on carrier frequency, slow
frequency hopping helps alleviate the problem. In addition, cochannel
interference is in effect randomized.
6 Discontinuous transmission
Minimizing cochannel interference is a goal of any cellular system, since
it allows better service for a given cell size, or the use of smaller cells,
thus increasing the overall capacity of the system. Discontinuous
transmission (DTX) is a method that takes advantage of the fact that a person
speaks less that 40 percent of the time in normal conversation , by turning the
transmitter off during silence periods. An added benefit of DTX
is that power is conserved at the mobile unit.
The most important component of DTX is, of course, Voice Activity Detection.
It must distinguish between voice and noise inputs, a task that is not as
trivial as it appears, considering background noise. If a voice
signal is misinterpreted as noise, the transmitter is turned off and a very
annoying effect called clipping is heard at the receiving end. If,
on the other hand, noise is misinterpreted as a voice signal too often, the
efficiency of DTX is dramatically decreased. Another factor to
consider is that when the transmitter is turned off, there is a very silent
silence heard at the receiving end, due to the digital nature of GSM. To
assure the receiver that the connection is not dead, comfort noise is
created at the receiving end by trying to match the characteristics of the
transmitting end's background noise.
7 Discontinuous reception
Another method used to conserve power at the mobile station is discontinuous
reception. 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.
8 Power control
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 cochannel
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 down to a minimum of 13 dBm (20
milliwatts).
The mobile station measures the signal strength or signal quality (based on
the Bit Error Ratio), 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 cochannel
cells alternatingly increase their power in response to increased cochannel
interference caused by the other mobile increasing its power. This
in unlikely to occur in practice but it is (or was as of 1991) under study.
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