Design and Simulation of ADSL/ADSL2 Initialization
Process
Abstract- in this paper , a design and
simulation for Asymmetric Digital
Subscriber Line (ADSL)/Asymmetric
Digital Subscriber Line 2 (ADSL2)
Initialization Process is presented which
can be applied to different telephone
network. The Initialization process is
designed and simulated under the
MATLAB v7 environment. ADSL channel
faces different types of noise, the most
important types are Background and
Crosstalk noises and Intersymbol
interference (ISI) which can be eliminated
by using Time Domain Equalizer (TEQ).
Minimum Mean Square Error (MMSE)
algorithm is implemented as TEQ
algorithm. The initialization process where
tested on American National Standard
Institute (ANSI ) define 8 Carrier Serving
Area (CSA) test loops for ADSL service As
a result , 9.02 Mbps and 9.42 Mbps were
achieved over CSA loop-2 for ADSL and
ADSL2 respectively. The TEQ efficiency
for 8 CSA test loop are calculated where it
achieved 93.1% of zero ISI for CSA loop-
3.a Comparison between Near End Cross-
Talk (NEXT) and Far End Cross-Talk
(FEXT) power shows that NEXT have
higher power and narrow band where
NEXT powers were (-45.593) dBm for all
loops and FEXT were ranging from (-
76.353 to -68.261) dBm for the 8 loops.
Finally the results show that ADSL2
outperforms ADSL by about 400Kbps
which about 1.1%.
I. Introduction
Broadband or high-speed Internet access is
provided by a series of technologies that
give users the ability to send and receive
data at volumes and speeds far greater than
current Internet access over traditional
telephone lines. In addition to offering
speed, broadband access provides a
continuous, “always on” connection (no
need to dial-up) and a “two way”
capability, that is, the ability to both receive
(download) and transmit (upload) data at
high speeds. Broadband access, along with
the content and services it might enable, has
the potential to transform the Internet: both
what it offers and how it is used. It is likely
that many of the future applications that
will best exploit the technological
capabilities of broadband have yet to be
developed [1].
Telephone line based technologies
provide dedicated access to the individual
users. One of the best solutions is Digital
Subscriber Line (DSL) access, which is
targeted for residential users and has
received much attention by many telephone
companies. The architecture of DSL
systems allows telephone companies to use
existing twisted-pair infrastructures, by
which there is no need to lay extra lines for
new services, for their next-generation
broadband access networks.
The most promising of the xDSL
technologies for integrated Internet access
is Asymmetric Digital Subscriber Line
(ADSL) by which ADSL is designed to
interoperate with the telephone, i.e.
allowing voice and high-speed data to be
sent simultaneously over the same line.
Unlike today's computer dialup modems,
the customer may use the telephone while
the computer is connected to the Internet
Service Provider (ISP). The achievable data
rate is significantly higher in ADSL by
almost 100 times compared to today's
fastest 56 Kbps modems [2].
A recent development is ADSL2 and
ADSL2+. It defines new applications,
services and deployment scenarios.
The first issue it has to deal with is a
wide range of used cables in telephone loop
plant also a varying length of loops from
few hundred of feet to 18000 feet. The
modem should adept well with these
changes [3].
Other users using ADSL or other
communication service running on the same
binder causes noises, these noises is called
crosstalk noises, where background noise
represents the thermal and environment
effect. Crosstalk noises have the dominant
effect on ADSL system performance.
Other ADSL impartment when
operating on long loops is dispersive
behavior of the channel, resulting in a wider
received pulse. This causes a time sample
to spread into the neighboring time slots
that causes InterSymbol Interference (ISI).
To eliminate this effect ADSL modem
should shorten the channel impulse to
desirable length.
Cioffe [4], described fundamentals of
MCM and how it is analyzed for channels
with ISI and additive Gaussian noise. Inoue
[5] ,was concerned with an improvement to
previously proposed ADSL echo canceller
using modified conjugate gradient for
adaption schema. Arslan [6], studied
different kinds of equalization technique for
discrete multitone transceiver and
developed a time domain equalizers (TEQ)
design method to optimize the channel
capacity at the output of the TEQ. Nadhim
[7] studied MCM technique in DSL
systems using the Fourier transform and
then he used the discreet wavelet transform
(DWT) and studied the system
performance. DALY [8], dealt with the way
of improving the efficiency of multicarrier
communication on the digital subscriber
loop and also examined bit and power
loading algorithms for multitone systems.
Since ADSL channel does not change
during data transmission and only minor
noises level changes occurs, all problems
are handled during ADSL modem start-up
process, which is known as ADSL
initialization process. In this work
ADSL/ADSL2 initialization process in
simulated under MATLAB version 7
environment.
II. ADSL/ADSL2 Initialization
The initialization process allows the ATU-
C and ATU-R to establish their
communications. The process allows the
two modems to identify themselves to each
other, determine line conditions available to
support communication, exchange
parameters that define the request
connection, allocate resources, and begin
normal communication. The process is
divided into four phases as described in
ANSI standard:
Activation and Acknowledgment: The
ATU-R begins the initialization process by
transmitting the appropriate tones to the
ATU-C. When this segment of initialization
is complete, the ATU-R and ATU-C have
negotiated the timing method used between
them and have determined which device is
the master. At the end of this procedure, the
ATU-R and ATU-C are in state capable of
analyzing the line condition. In this paper,
the ATU-C and ATU-R are assumed to
have negotiated the timing method and
perfectly handshaking and all activation and
acknowledgment process is done perfectly.
Transceiver Training: During this process,
the ATU-R and the ATU-C send signals
that allow their partner to determine line
conditions and adjust the equalization of
their transceiver. Transceiver training also
determines if ADSL is operating in FDM or
Echo cancellation mode.
Channel Analysis: The modem exchange
information on the upstream and
downstream bearer channel required for the
connection the latency paths they will be
placed in, and the bandwidths for each
channel requested. Information about
specific features supported or requested is
also exchanged. The modems then perform
test that determine the loop quality and
SNR for each specific 4 kHz DMT tone.
Exchange: Having gathered the
information about quality of the connection
and the requested configuration, the
modems configure themselves and
exchange information about their
configuration, the specific bandwidth
allocated to the requested bearer channels is
assigned, the specific DMT tones and the
amount of the data encoded in each tone are
determined and assigned. The connection is
tested in both directions after which each
modem notifies its peer that it is ready to
enter normal communications. In this
thesis, this phase is represented by bit
loading process.
The ADSL test loop modeling
process simulate the ANSI CSA loops as
ADSL modem test loop, noise modeling
process simulate AWGN, NEXT and FEXT
noises for ADSL service. Channel analysis
process measure ADSL modem
performance on test loops with the
existence of noises by calculating signal to
noise ratio (SNR) assuming free ISI
channel. Since the used test loop is not free
ISI, therefore ISI elimination process is
required. Channel equalization process
eliminate channel ISI by using Time
Domain Equalizer (TEQ) and measure
modem performance on test loop after
equalization by calculating equalizer output
signal to noise ratio (SNR). Bit loading
processes calculates modem performance in
terms of bit rate and calculate modem bit
loading table using equalizer output (SNR).
A. DSL channel
The transmission characteristics of DSL
loops determine the performance of DSL
systems. DSL loops are based on existing
analog telephone subscriber loops which
were originally developed for voice
communication. A subscriber loop consists
of twisted pair cables that connect a local
central office to customer premises. Two
subscribers are connected to each other
through central offices.
In order to facilitate the development
of broadband communication on the DSL,
the ANSI and ESTI standardization bodies
specified test loops which would hopefully
be representative of a large cross-section of
the DSL links encountered in practice.
The ANSI introduced the Carrier
Serving Area (CSA) engineering guidelines
in the early 1980s to shorten subscriber
loop length, which reduces loop
deployment cost and supports all future
digital services. A carrier serving area
(CSA) is a plant administration area
subsection of the main loop plant. [4].
These generic lines are referred to as ADSL
CSA test-loops [9].

Figure 2. Configuration of the Eight

Figure 1. Transmission Line
Schematic.
The ANSI CSA test loop is modeled and
used as test loops as is shown in Fig 1. The
load impedance shown in Fig. 1 is
considered to be a real constant (100 in
ANSI loops). The eight standard CSA loops
are used as test channels in the
ADSL/ADSL2 initialization simulation.
A splitter is used to separate the voice
signal from ADSL signal, hence a highpass
filter with low transition frequency is
needed. For the purposes of modling, a fifth
order Chebyshev high-pass filter with cut-
off frequency of 4.8 kHz was used to
simulate splitter effect.
Standard CSA loops. Number Represent
Length/Thickness in Feet/Gauge. Vertical
Lines Represent Bridge-Taps.
B. Telephone Line Impairments
Subscriber loops which connect the
customer premises to a Central switching
Office (CO), were developed and deployed
for voice transmission. The term loop refers
to the twisted copper pair telephone line
from a CO to the customer. This term
originates from current flow through a
looped circuit from the CO on one wire and
returning on another wire [10].
All signals sent over conventional
pair-cable telephone lines are subject to line
attenuation, dispersion and electrical noise.
Line attenuation and some forms of in-band
noise both increase with frequency [11].
Intersymbol interference (ISI) is an
unavoidable consequence of both wired and
wireless communication systems. ISI
causes spreading of received signal.
As transmissions occur more frequently
(that is, the symbol rate is increased, which
corresponds to decreasing symbol time T
for each successive), then a given channel
shape exhibits increasing overlap of
transmissions and thus more ISI.
Equalization methods adaptively configure
the receiver to mitigate ISI [12].
Multicarrier Modulation (MCM)
systems use Cyclic Prefix (CP) to separates
the symbols in time in order to decrease ISI.
As it is well known, the signal going
through the line is linearly convolved with
the impulse response of the line. If the
impulse response is shorter than the
duration of the cyclic prefix, each symbol
can be processed separately, and there is no
ISI [11].
Another impairment of data transmission is
white noise. One special kind of noise is
Additive White Gaussian Noise (AWGN)
which represents background noise as
already specified by its name. It has a flat
spectrum and a Gaussian amplitude
probability. Normally it is characterized by
its power spectral density.
Since AWGN have constant PSD
( /2), it is easly built by generating a
random signal process and muliplying it by
the AWGN power, where AWGN power is
the AWGN PSD (Watt/Hz) multiplied by
the used bandwidth.
For simulation, different levels of
AWGN have been used, all moving in a
range between (-170 to -140 dBm/Hz)
single sided power spectral density [13].
For ADSL, however, -140 dBm/Hz seems
to be the most used value for AWGN [14].
The performance of DSL transceivers
can be impaired by interference from. The
twisting of the wire pairs reduces this
inductive coupling (also referred as
crosstalk), but some signal leakage remains.
Crosstalk is most pronounced at the
segment of cable near the interfering
transmitters. The crosstalk resulting from
other transmission systems in the same
cable (and especially the same binder group
with the cable) is a primary factor limiting
the bit rates and loop reach achievable by
DSLs. There are two types of crosstalk
[12].
Near End Cross-Talk (NEXT) is a
major impairment for systems that share the
same frequency band for upstream and
downstream transmission. NEXT noise is
seen by the receiver located at the same end
of the cable bender of the transmitter that is
the noise source.
Far End Cross-Talk (FEXT) is the
noise detected by the receiver located at the
far end of the cable from the transmitter
that is the noise source. FEXT is less severe
than NEXT because the FEXT noise is
attenuated by traversing the full length of
the cable [12].
In order to evaluate a NEXT or a
FEXT crosstalk signals, the PSD function
of the interfering service has to be known.
The PSD of downstream and upstream for
ADSL will be discussed in this subsection.
This paper will focus on Standard
T1.413 by ANSI as a reference for the
ADSL service. The ADSL upstream
according to ANSI can be given by

the is the same for both the
download and upload stream except for the
differences between the and
coefficient. The NEXT for
ADSL service can be calculated using
where U is the number of disturbers in the
cable, is the coupling
coefficient for 49 NEXT disturbers and
is the frequency in Hz. The FEXT for
ADSL service can be calculated using
Where is the channel gain (frequency
response), 8 x 10-20 is the coupling
coefficient for 49 FEXT disturbers and is
the coupling path distance.
Generating a crosstalk signal is
conveniently accomplished by defining a
filter transfer function in terms of the
desired power spectral density (PSD). If
the input to the filter is a white noise
process with unit variance, then the PSD is
equal to the magnitude squared of the filter
transfer function. The filter transfer
function is
Where is the PSD in Watts/Hz.
The total Noise PSD will be
Since there are different PSD origins, each
PSD is can be simulated independently and
the resulted signal is summed to generate
the total noise signal.
C. Equalization for Discrete
Multitone Modulation
With DMT, the problem of fully equalizing
a channel is converted into partitioning the
channel into small subchannels which is
more efficient to implement in high-speed
transmission. However, this does not imply
that equalization is not required in a DMT
system. The spectra of each inverse FFT
(IFFT) modulated subchannel is a sampled
sinc function which is not bandlimited.
Demodulation is still possible due to
the orthogonality between the sinc
functions. An ISI channel, destroys
orthogonality between subchannels so that
they cannot be separated at the receiver.
One way to reduce ISI with a shorter
cyclic prefix is to use an equalizer. Since
the length of a DMT symbol is longer than
a symbol in single carrier modulation,
equalization is simpler. Also, noise
enhancement by the equalizer is not an
issue because the equalizer does not affect
the SNR in each subchannel, which are the
primary parameters to determine the
performance of a DMT system.
The ADSL standard uses a guard
period, time-domain equalization, and
frequency-domain equalization. The Time-
Domain Equalizer (TEQ) shortens the
channel to a length of a predetermined but
short guard period. The TEQ can be
implemented as an FIR filter whose filter
coefficients are trained during initialization.
This combination has been standardized
and is implemented in practical systems [4].
Minimum Mean-Squared Error Impulse
Response Shortening Algorithm
Chow and Cioffi [15] are the first to apply
channel shortening equalization to
multicarrier modulation. They use the
MMSE design method to shorten a given
channel to the length of the cyclic prefix.
The idea behind the MMSE TEQ
design method may be explained by Fig. 3.
The structure consists of an FIR equalizer
in cascade with the channel and a parallel
branch that consists of a delay and an FIR
filter with a target impulse response (TIR).
The goal in the MMSE design of the vector
of TEQ taps (w) is to minimize the mean
square of the error between the output of
the equalizer and the output of the TIR.
Assume that the error is zero for any given
input signal. That means the impulse
responses of both branches are equal. In
other words, the equalized channel impulse
response (upper branch) would be equal to
a delayed version of the TIR. Setting the
number of taps of the TIR to a desired
length forces the equalizer channel impulse
response to have the same length.
Figure 3. Block diagram of Minimum
Mean-Squared Error (MMSE) Equalizer[6].
Fig 4. shows a TIR and Shortened
Impulse Response (SIR). The MMSE
design method formulates the square of the
difference between the TIR and SIR as the
error and minimizes it. The method
minimizes the difference between the TIR
and SIR both inside and outside the target
window.
In fact, the difference between the
TIR and SIR inside the target window does
not cause ISI. Both the TIR and SIR inside
the target window have higher amplitudes,
which mean that difference inside the target
window might contribute more to the MSE
than the difference outside.
Since the MMSE method in general
cannot force the error to become exactly
zero, some residual ISI will remain. To
maximize channel capacity, the residual ISI
should be placed in frequency bands with
high channel noise.
Figure 4. Target Impulse Response (TIR)
and Shortened Impulse Response (SIR)[6].
This ensures that the residual ISI
would be small compared to the noise and
the effect on the SNR would be negligible
[4]. The MMSE design method does not
have a mechanism to shape the residual ISI [12]. As was described before, the encoder
in frequency. MMSE algorithm is takes the data bit stream and encodes it into
implemented in chapter five and used to N QAM constellation points for each
eliminate ISI. subchannel. This encoding is done
according to the bit loading table which
Table 1. Performance Comparison defines the number of bits carried by each
between Selected TEQ algorithems [16]. tone.
Bit Rate as a percentage The ADSL system has the following rules
Loop of Match Filter Bound Rb(MFB) for bit loading as given in ANSI T1.413:
CSA (MFB) (Mbps)  Only integer number of bits is
General MSSNR Min allowed, if the resulted is not
MMSE ISI integer, it is rounded to the least
1 92% 62% 98% 8.45 integer.
 The minimum number of bits that
2 90% 75% 97% 9.68
can be carried on any sub channel is
3 91% 82% 99% 8.11 2 (4-QAM), so any subchannel that
4 92% 61% 98% 8.05 cannot carry 2 bits is turned off.
5 85% 72% 98% 8.53  The maximum number of bits that
6 93% 80% 99% 7.77 each subchannel can carry is 15 bits,
7 77% 74.2% 96.0% 7.75 if the resulted is greater than 15
8 56% 71% 99% 6.90 bits, it is replaced by 15 bits
In ADSL2 system, the same rule can be
A comparison of selected TEQ algorithms applied except for the following, as
can be shown in Table 1 as percentage to described in ITU G.992.3:
Match Filter Bound (MFB) which  The minimum number of bits that
represents perfect equalization, the number can be carried on any sub channel is
of TEQ taps is 17. The Minimum Mean 1bit (2-QAM), rather than 2 bits for
Squared Error (MMSE) is the most ADSL, any subchannel that cannot
commonly used in commercial ADSL carry 1 bit is turned off.
modems. MMSE design methods are
relatively easier to implement with adaptive
algorithms and are efficient in the sense of E. Channel Analysis
computational complexity [4]. MMSE
algorithm is implemented in chapter five This section will discuss the process of
and used to shorten test loop impulse calculating ADSL signal to noise ratio
response. (SNR) which is needed to determine the
channel capacity.
D. Bit Loading During channel analysis process
ATU-C send signal know as Medly. This
The process of assigning information and signal consists of binary random data
energy to each of the subchannels is called transmitted to ATU-R. All subchannels are
bit loading in multichannel transmission
used and each subchannel is modulated
with 4-QAM signals at maximum allowable
power.
On the other side, the ATU-R will
receive the Medly signal but after the signal
was filtered by the channel. The ATU-R
will need also to estimate channel noises.
The received Medly signal power and its
calculated noise are used to calculate each
subchannel SNR value.
Fig. 5 shows the ADSL downstream
mask. Using ADSL PSD mask, the total
power can be calculated, as stated
previously only ADSL downstream ATU-C
to ATU-R will be simulated. The used PSD
mask that is the downstream PSD over
POST with overlap spectrum so the
bandwidth that will be used for
transmission is from 25.875 KHz to 1104
KHz.
Figure 5. ADSL Downstream PSD Mask.
The maximum transmitted power
PSD is -36.5 dBm/Hz but a margin of 3.5
dBm is used as the filter pass band ripple as
was described in ANSI T1.413[17]. The
resulted maximum power will be -40
dBm/Hz. Hence the total power is given by:
Also
where is the transmitted signal power
spectrum and represent subchannel
power.
Using the generated noise and the
known channel impulse response, the SNR
for each of the DMT subchannels can be
calculated. Hence
where is the SNR for the
subchannel, is the channel gain
(frequency response) for the subchannel
(which can be calculated by taking the
square of the absolute value of the Fourier
transform of the channel impulse response),
is the noise PSD for the
subchannel, assuming free ISI channel. The
SNR for a channel without ISI is referred to
as , where MFB stand for Match
Filter Bound. The average SNR for DMT
channel can be calculated using.
Where term is called the average SNR
or geometric SNR ( .
The Time domain equalizer (TEQ)
algorithm is not perfect and do not shorten
the channel to (v+1). The residual will
cause ISI and that will affect on the system
performance. To calculate SNR that include
the residual ISI the following equation can
be used :
where is the equalizer output SNR,
is the equalizer gain (frequency
response) and is the ISI gain of the
residual ISI.
 The represents the equalizer III. Simulation of ADSL/ADSL2
output SNR which is used to calculate Process
channel capacity. In order to calculate the
efficiency of the equalizer, the total bit rate The CSA loops modeled with impulse
for equalized channel and Match Filter responses consist of 512 samples and
Bound channel using the geometric SNR of sampled at a rate of 2.208 MHz using
the equalizer output and the Linemod[16], the ADSL channel noises, the
MMSE TEQ algorithm were all modeled
geometric SNR of the free ISI channel
using MATLAB then ADSL channel
are used:
analysis is calculated finally the bit-loading
table is built, The MATLAB code were
assembled to create a program that
To calculate the number of bits that each
simulate the initialization process for
subchannel can carry, the default bit-
ADSL/ADSL2 modems, as stated before
loading algorithm is used to calculate the
the program simulate the downstream
bit loading table as explained in subsection
ATU-C to ATU-R only.
D where :
As shown in Fig. 6, the program
consists mainly from 6 MATLAB file
Here, is the number of bits that can be codes, that simulate the ADSL/ADSL2
carried on the subchannel and is the initialization process. Each file represent a
modulation gap. process that was explained in previous
For DSL application the targeted bit subsections.
error rate is , the modulation Table 2 shows the used parameters
gap for QAM ( ) is 9.8 dB as defined for ADSL/ADSL2 initialization process
in ANSI, and thus, the for a DSL system simulation.
is Table 2. ADSL/ADSL2 Test Parameters.
where is the DSL margin, for standard Parameters Value
ADSL system the margin is 6 dB, is the Name
coding gain for the used error correcting Input Power 20 dBm
system, the code gain for ADSL system
with RS code is 4.2 dB and for ADSL2 Number of 19 tap
where Trills code is used the code gain is Equalizer Tap
5.5 dB as defined in section 4.5. The total Margin 6 dB
bit rate ( ) can be calculated by using: Code Gain 4.2 dB for ADSL
5.5 dB for ADSL2
where is the total number of transmitted AWGN Power -140 dBm/Hz
bits in one frame and is the frame period.
NEXT Users 10
FEXT Users 10
 Figure 6. ADSL/ADSL2 Initialization Program Structure.

A GUI (graphical user interface) was built IV. Simulation Results and
to ease the use of the initialization program,
Discussion
In this subsection the result for the ADSL response to the required length (v+1), the
initialization on CSA test loop-1 is remaining impulses will cause ISI as shown
presented. ADSL2 have similar results in Fig 8. Table 4 shows the equalizer
except for bit loading table since ADSL2 efficiency for all CSA loops.
modem has higher code gain.
Table 3. CSA Loops NEXT and FEXT
A. Channel Modeling Results Power.
NEXT Power FEXT Power
ADSL CSA loop-1 was simulated as shown CSA (dBm) (dBm)
in figure (5.19) and filtered with high pass Loop 10 24 10 24
filter to simulate POST splitter effect on the Users Users Users Users
channel effect. The impulse response of the Loop-1 -45.593 -43.28 -71.475 -69.143
channel is much greater than 33 (v+1) Loop-2 -45.593 -43.28 -72.924 -70.5
which is the maximum allowable value for Loop-3 -45.593 -43.28 -76.353 -74.157
ADSL. For this reason, the Time Domain Loop-4 -45.593 -43.28 -72.17 -69.86
Equalizer (TEQ) is required. Loop-5 -45.593 -43.28 -72.58 -70.373
Loop-6 -45.593 -43.28 -71.7 -69.497
Loop-7 -45.593 -43.28 -68.261 -65.981
B. Noise Modeling Results Loop-8 -45.593 -43.28 -68.901 -66.60
Using Eqs (1-5), the noise signals AWGN,
NEXT, FEXT were modeled. Fig. 7 shows
AWGN, NEXT and FEXT power spectrum.
The Next Power is -45.593 dBm and FEXT
power is -71.475 dBm. Table 3 shows the
NEXT and FEXT power for all loops.
It is noticed that NEXT power did
not change with the change of loops. This is
because NEXT is channel independent,
while FEXT is channel and couple length
dependent.
Since equalizer effect is not only on
the channel and signal but it also affect the
received noise and its residual ISI cause
noise, it called ISI noise.

C. Channel Equalization Results Figure 7. AWGN, FEXT and NEXT Power

By implementing MSSE an equalizer with
19 tap using MMSE algorithm was used to
shorten the CSA loop-1. The equalizer
shortens most of the channel impulse
Spectrum.
It is noticed that the equalizers efficiency is
less than that given in section II, since the
equalizer efficiency depends on the
transmitted power, AWGN power, NEXT Using Eq. 10, the Geometric MFB
users and FEXT users. Changing of these (Match Filter Bound) SNR ( ) for
parameters will change the equalizers CSA loop-1 is 40.182 dB, Geometric
efficiency. Equalizer SNR ( ) is 32.58 dB and
the Equalizer Efficiency is 81.2 %. Table 5
shows the achieved equalizer geometric
SNR ( ) and Match Filter bound
SNR ( ) for all CSA loops.

Figure 8. Comparison between Channel

and Equalized Channel Impulse Response.

Table 4. CSA Loops Equalizer Efficiency.

CSA Loop Equalizer Efficiency
Name ( ) (%) Figure 9. Transmitted Signal Power
Loop-1 81.2 Spectrum.
Loop-2 89.8
Loop-3 93.1 Table 5 CSA Loops Calculated
Loop-4 80.0 and .
Loop-5 85.1
Loop-6 87.1 CSA Geometric Match
Loop-7 70.8 Loop SNR Geometric
Loop-8 85.1 ( ) SNR
(dB) ( )
(dB)
D. Channel Analysis Results Loop-1 32.58 40.123
Loop-2 39.57 44.1
The used transmitted signal spectrum
Loop-3 36.5 39.21
during the SNR calculation is shown in Fig
9. The average transmitted is 20 dBm Loop-4 31.36 39.2
which is the maximum allowable Loop-5 34.39 40.41
transmitted power. Loop-6 33.31 38.24
The signal to noise ration SNR can Loop-7 26.96 38.08
be calculated using Eq. 9 and Eq. 11 Loop-8 29.8 35.01
for and respectively.
 before real system installation, it provide
The loops equalizer output SNR information about potential system
depends on the equalizer efficiency and performance, Time Domain Equalizer
more efficient equalization algorithm can (TEQ) can shorten the impulse response to
achieve better SNR. desirable system impulse response length,
. for the MMSE algorithm as TEQ algorithm,
TEQ can achieve up to 93% of zero ISI
E. Bit Loading Table Results channel. Also Time Domain Equalizer
(TEQ) increase noise level especially
The bit loading table is calculated by using higher frequencies which cause BER, using
Eq. 13 and applied the rules of adaptive TEQ can mitigate noise
ADSL/ADSL2. Using Eq. 15 the total bit enhancement, Noise effects on bit rate,
rate for CSA Loop-1 is 6.62 Mbps. Table 6 where higher AWGN level, more FEXT
shows a comparison between ADSL and and NEXT interfered user’s cause’s higher
ADSL2 Total Bit Rate. noise power which leads to reduce bit rate.
NEXT is the dominant channel impartment
Table 6 Comparison between ADSL and compared to AWGN and FEXT, where
ADSL2 Total Bit Rate over CSA Loops. NEXT have higher power and narrow
CSA ADSL ADSL2 bandwidth, Bit loading algorithm determine
Loops Total Bit Rate Total Bit Rate the number of bits that can be carried on
Rb (Mbps) Rb (Mbps) each subchannel, using Default bit loading
Loop-1 6.620 7.08 algorithm ensure equal probability of error
Loop-2 9.02 9.42 on all subchannels, where it is fast and
Loop-3 7.94 8.36 simple and can be done offline, ADSL2
Loop-4 6.2 6.68 achieve higher bit rate than ADSL by more
Loop-5 7.22 7.68 than 400 Kbps on CSA loops also ADSL2
Loop-6 6.86 7.32 have higher code gain and more flexible bit
Loop-7 4.6 5.09 loading role.
Loop-8 5.66 6.07
VI. References
It is noticed that ADSL2 achieves
better bit rate than ADSL in about 400 [1] Angele A. and Lennard G. Kruger, ”
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[2] M. Rehman, "DSL vs. Cable Modem",
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[3] M. Ding, ” Channel Equalization To Achieve

Modling standard loops has important role High Bit Rates In Discrete Multitone
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[6] Z. Cai , “Enhancement of ADSL design
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[9] NEXTEP Broadband Networks Group, "xDSL
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[10] T. J. Willink and P. H. Wittke, “Optimization
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[11] J. Campello, “Practical bit loading for DMT”,
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[12] D. Daly , C. Heneghan and A.D. Fagan,
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multitone communications”, IEEE Int.,
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[13] Dr. B. Z. Hassan, "Error Detection and
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Lecture, Al Nahrain University, Iraq, 2004.
[14] Comtech AHA corporation Primer, "Reed-
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[15] J. A. C. Bingham, "ADSL, VDSL, and
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[16] D. G. Messerschmitt and A. Salvekar,
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analysis", Dept. of Electrical Eng. and
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[17] F. Van der Puteen, Alcatel Telecom and F.
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