(IJCSIS) International Journal of Computer Science and Information Security, 2009
Vol. 4, No. 1 & 2, 2009
Channel Equalization in Digital Transmission
Kazi Mohammed. Saidul. Huq #1, Miguel Bergano#1,Atilio Gameiro #1, Md. Taslim Arefin *2
# *
Institute of Telecommunications
Aveiro, Portugal
kazi.saidul@av.it.pt
Abstract— Channel equalization is the process of reducing
amplitude, frequency and phase distortion in a radio channel
with the intent of improving transmission performance. Different
types of equalizers, their applications and some practical
example is given. Especially, in the digital communication
scenario how equalization works is shown. This paper presents a
vivid description on channel equalization in digital transmission
system.
Keywords— ISI, Baseband, Passband, equalization
1. INTRODUCTION
A communication system is basically a way of transmitting
information trough a communication channel and usually
associated with it are the transmitter and a receiver. The main
function of it is to guarantee that information, or message,
from the transmitter should be available at the receiver
without perturbations. A communication system is completed
when joining these three parts, the transmitter, the receiver
and the communication channel. Examples of communication
channels are telephone channels, coaxial cables, optical fiber,
wireless broadcast channel, mobile radio channel and satellite
channels. The signal to be transmitted could be analog or
digital. The first one implies the use of fewer hardware on the
receiver and transmitter, on the contrary digital signals need
more hardware, although digital systems are more stable,
flexible and more reliable. It should be noted, however, that
we can implement much of an analog communication system
using digital hardware and the appropriate ADC and DAC
steps, and thereby secure an analog system many of the
advantages of a digital system.
Ideally a system like this would work perfectly but due to
imperfections of the channel it can be defined by a more
complete diagram, represented in Fig. 1.
Lecturer, Dept. of Computer Science and Engineering
University of Development Alternative (UODA)
Dhanmondi, Dhaka-1209, Bangladesh.
2
arefin.taslim@cse.uoda.edu.bd
Figure 1: Elements of a communication System
In a real communication system the communication channel is
not perfect, perturbations caused by imperfections on the
transmission channel or interference from outside world, for
example, can generate a bad functionality of the channel.
Having these issues the channel will not perform a flat
frequency response and linear phase shift mainly because of
distortion. Interference and noise are contaminations that
occur from other radio systems and from random electrical
signals produced by natural processes, respectively. In order
to perform a good way of conveying information from
transmitter to receive the problems mentioned earlier should
be considered in modeling a communication system. The main
task in this procedure is to take the channel conditions and in
some way invert it, or in other words, a channel can be
mathematically estimated by a transfer function, at the output,
or at the receiver, it would be a system with an inverse of that
transfer function. Some problems arise in modeling the
channel; issues like nonlinearity or time variance induce
difficulties. All these mentioned issues are obstacles to
approach an ideal frequency response of the communication
system or to identify the channel characteristics exactly.
1.1. Digital Transmission
A digital transmission performs digital messages that are
basically ordered sequence of symbols produced by a discrete
information source. Here the task is to transfer a digital
message from the source to the destination. In an analog
communication system problems like the channel frequency
bandwidth and the signal to noise ratio cause errors that
appear in the received message, similarly, signaling rate and
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Vol. 4, No. 1 & 2, 2009

error probability play roles in digital communication namely

in the output messages.

Digital signal has usually a form of an Amplitude Modulated

Pulse Train and are commonly expressed by:
x(t) = ∑ ak p (t − kD) (1)
k
where ak represents the modulated amplitude of each symbol k,
D is for pulse duration or pulse to pulse interval and p(t) is the
unmodulated pulse that has the values 1 or 0 periodically. In
the case of binary signaling D indicates bit duration so D=Tb,
and the bit rate is rb=1/Tb measured in symbols per second or
baud. Digital PAM signals can take several formats, basically
a simple on off with a defined duration generates a format
called RZ (return to zero), but others exist like NRZ (Non
return to Zero), both polar having a DC component that
Figure 2: Binary PAM formats
wastes power, an alternative is bipolar NRZ or split-phase
Manchester or even quaternary. In a transmission process
there are several perturbations, noise contamination cross talk Always associated in all these formats are the noise reduction
or spill over from other signals a phenomena described as ISI by introducing a filter, this filter should not introduce ISI, like
– Inter Symbol Interference – that basically is a form of showed in Fig. 3.
distortion of signals in which symbols interfere with
subsequent symbols. Reducing the bandwidth of filter will
reduce noise but would increase the ISI, for that Nyquist
stated that the symbol rate r must be lower than the twice of
channel bandwidth.
r ≤ 2B (2)
On the list of the digital transmission limitations is obvious Figure 3: Baseband transmission system
the channel, so to approach an ideal frequency response the The amplifier compensates losses in the channel and the filter
channel must be equalized. The equalizer is usually inserted LPF removes out of band contaminations, the output message
between the receiver and the channel regenerator. With this, it is the recovered message from the digital signal.
will increase the knowledge of the channel characteristics that
sometimes results in some residual ISI. An equalizer is based To transmit in longer distances passband digital transmission
on the structure of a transversal filter, like it will be shown is used, and requires modulation methods applied in analog
later. signals. Digital information has a lot of ways to be performed
in a carrier wave, it can modulate amplitude, frequency or
1.2. Baseband and passband digital transmission phase of a sinusoidal carrier waive.

At baseband a digital message is represented by a PAM and
expressed like equation (1). Above the modulated forms that a
baseband signal can take was already mentioned, RZ,
NRZ(NRZ-L, NRZ-M, NRZ-S), Bipolar, Biphase (Biphase-L,
Biphase-M, Biphase-S), Differential Manchester:
Any modulated passband signal may be expressed in the
quadrature-carrier form:
x c (t) = Ac [x i (t)cos(wc t + θ ) − x q (t)sin(wc t + θ )] (3)
The carrier frequency fc, amplitude Ac and phase are constant.
The message is contained in the phase – i – and quadrature – q
– components. An amplitude modulation (ASK – Amplitude
Shift Keying) can be achieved simply using a NRZ signal,
another example is QAM (Quadrature Amplitude Modulation)
that achieves higher modulation speed. Phase Shits can also
perform phase modulation often described as BPSK (Binary
Phase Shift Keying, if the signal has four elements in the
alphabet the modulation is QPSK (Quaternary Phase Shift
Keying). An example of transmitter is in Fig. 4:

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Vol. 4, No. 1 & 2, 2009

x(t) is for the output of the filter. At the receiver is the filter,
with frequency response HR(t) or impulse response hR(t), a
sampler and a comparator. So at the ouput it will be:
+∞
v(t) = y(t) ∗ hR (t) = ∑ Aa p (t − t
k r d − KTb ) + n 0 (t) (7)
k=−∞
The value A is a scale factor such that pr(0)=1, n0(t) is the
noise component at the receiver filter output, and pr(t-td) is the
pulse shape at the receiver filter output, delayed by an amount
td due to filtering.
Having all the information of the response of a
communication system it is possible to develop forms to
Figure 4: QPSK Transmitter minimize problems in the system, like ISI and SNR reduction.
Zero ISI and and noise can be achieved by choosing the
correct HT(f) and HR(f). The equations, given in [1][2],
A frequency modulation (FSK – Frequency Shift Keying) is demonstrate that is a hard task to create such frequency
obtained when the input signal x(t) selects the frequency of an responses mainly because of the channel conditions, in a
oscillator. baseband transmission, or PAM, like a modem, it must have
information about the channel. For passband transmission,
2. PRINCIPLES like cellular radio there are several obstacles to the
transmission, or for microwave links that depend on the
Previously was defined several concepts of digital atmosphere conditions. So the best filter to use at the receiver
transmission systems, namely its limitations (bandwidth, noise, must be adjustable improving the performance of a
distortion and ISI), and formats of transmitting (modulation) transmission. Such filter is called equalizer. There are two
for baseband and passband transmission. In order to avoid the types of equalizers: preset and adjustable. The first one its
issues related to these types of communications systems it parameters are determined by making measurements on the
must properly designed. channel and determining these parameters using these
measurements. The adaptive, is automatic, its parameters are
adjusted by sending a known signal, called training signal.

Transmitter Channel
Source Equalizer
Filter Filter
Figure 5: Ideal model of a communication System
Figure 6: Block Diagram of PAM Communication System with
The signal source input has the regular input represented by (1) equalization
this time p(t) has the form of a unit pulse δ(t). The next The previous figure illustrates the process of equalization. The
subsystem is transmission filter with low pass frequency overall frequency response is:
response HT(f) or impulse response hT(t). The transmitted
signal is given by:
+∞ +∞
H ( f ) = H ( f )H ( f )H ( f ) (8)
0 T C E

x t (t ) = ∑ a δ(t − kT )∗ h
k b T (t) = ∑ a h (t − kT )
k T b
(4)
In theory an equalizer should have an impulse response that is
k=−∞ k=−∞
the inverse of that on the channel, and the design of this
Where the asterisk represents convolution. The channel systems involves a compromise between ISI reduction and
can be considered as a filter, due to its bandwidth limitations noise reduction of the channel.
and imposes a frequency response function HC(f) or impulse
response hC(t) and additive Gaussian Noise represented by n(t).
At the receiver will be: 3. TYPES OF EQUALIZERS

y(t) = x(t) + n(t) (5) 3.1. Zero forcing

x(t) = x t (t) ∗ hC (t) (6)
The basic idea of a Zero-Forcing Equalization – ZFE – is to
implement a filter (equalizer) that follows the channel
response, or like already said, the channel filter. The system of

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a ZFE has a frequency response indicated in (8). Asuming that
the first Nyquist critereon is satisfied by the sampler a ZFE is
like a inverse filter, the inverse frequency response of the
chanell frequency response and is usually approximated by a
set of FIR filters like is presented in Fig. 7.
To formulate a set of FIR inverse filter coefficients, a training
signal consisting of an impulse is transmitted over the channel.
By solving a set of simultaneous equations based on the
received sample values, a set of coefficients can be
determined to force all but the center tap of the filtered
response to 0. This means the N–1 samples surrounding the
center tap will not contribute ISI. The main advantage of this
technique is that the solution to the set of equations is reduced
to a simple matrix inversion.
The major drawback of ZFE is that the channel response may
often exhibit attenuation at high frequencies around one-half
the sampling rate (the folding frequency). Since the ZFE is
simply an inverse filter, it applies high gain to these upper
frequencies, which tends to exaggerate noise. A second
problem is that the training signal, an impulse, is inherently a
low-energy signal, which results in a much lower received
signal-to-noise ratio than could be provided by other training
signal types [3][6].
Figure 7: Filter Structure of a ZFE
3.2. Minimum Mean Square
Since ZFE ignore the additive noise and may significantly
amplify noise for channels with spectral nulls another type of
equalizer my be used to partially avoid this problem. The
Minimum-mean-square error (MMSE) equalizers minimize
the mean-square error between the output of the equalizer and
the transmitted symbol. They require knowledge of some auto
and cross-correlation functions, which in practice can be
estimated by transmitting a known signal over the channel. In
such an equalizer the Coefficients in Fig. 7 are chose to
minimize the mean square error (MMSE). The error consists
on the sum of the squares of ISI terms plus noise.
The MMSE at the equalizer is the expected value of the
square of the error.
Vol. 4, No. 1 & 2, 2009
MMSE = E [(error) 2 ] (9)
Analitically the error represents the difference between the
desired value and the real value.
{
MMSE = E [z( t ) − d (t )]
2
} (10)
Following this concept of obtaining the minimum error, the
task is to determine the taps of the filter in Fig. 7 in order to
perform a transmission with minimum errors. In Fig. 8 is
presented a scheme points out the interesting signals used in
the process[10].
y(t)
Transverse z(t)
Channel and
Receiver ∑ Filter
Decision
Filters
Equalizer
Noise ∑
d’(t)
Transverse
Filter
Equalizer
Figure 8: MMSE Equalizer Circuit
3.3. Adaptive equalizers
Most of the times the channel, besides being unknown, it is
also changing with time, a solution can be achieved by
creating an algorithm that adjust the taps of the filter by
following the channel and lead to the optimum values of the
equalizer. Adaptive equalization has different ways to perform
automatic algorithms.
3.3.1 Decision Directed Equalization
The previous equalizer systems are linear in that they employ
linear transversal filter structures. The filters implement a
convolution sum of a computed impulse response with the
input sequence. Often with data communication systems, one
can take advantage of prior knowledge of the transmit signal
characteristics to deduce a more accurate representation of the
transmit signal than can be afforded by the linear filter. It is
possible to devise a decision device (a predictor or a slicer)
that estimates what symbol value was most likely transmitted,
based on the linear filter continuous output. The difference
between the decision device input and output forms an error
term which can then be minimized to adapt the filter
coefficients. This is true because a perfectly adapted filter
would produce the actual transmitted symbol values, and,
therefore, the slicer error term would go to 0. In practice, the
error is never 0, but if the adapted filter is near ideal, the
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Vol. 4, No. 1 & 2, 2009

decisions are perfect. In this case, the slicer is effectively channel filter are calculated. It verifies a good estimation of
throwing away received noise with each decision made. the channel filter parameters as demonstrated by the error
curve, that present values of 10-1. Also it verifies very
3.3.2 Decision-Feedback Equalization approximate values of the weights.

Another nonlinear adaptive equalizer should be considered:

the decision feedback equalization (DFE). DFE is based on
the principle that once we have determined the value of the
current transmitted symbol, we can exactly remove the ISI
contribution of that symbol to future received symbols (see
Figure 5). The nonlinear feature is again due to the decision
device, which attempts to determine which symbol of a set of
discrete levels was actually transmitted. Once the current
symbol has been decided, the filter structure can calculate the
ISI effect it would tend to have on subsequent received
symbols and compensate the input to the decision device for
the next samples. This postcursor ISI removal is accomplished
by the use of a feedback filter structure.
Figure 9: Real and Estimated Output Signals

4. DESIGN IN BASEBAND

In baseband the frequency bandwidth of transmission is equal

to the symbol rate. In this case the samples are real numbers
while passband the samples are complex numbers. A first
consequence of baseband equalization is the delay
introduced by the equalizer in the carrier recovery loop. This
delay affects the loop stability, steady-state jitter
performance as well as its acquisition behavior. An example
of a Least Mean Square Algorithm is presented next.

4.1. Program in Matlab

The least mean squared (LMS) equalizer is a more general Figure 10: Error
approach to automatic synthesis. The coefficients are
gradually adjusted to converge to a filter that minimizes the
error between the equalized signal and the stored reference.
The filter convergence is based on approximations to a
gradient calculation of the quadratic equation representing the
mean square error. The only parameter to be adjusted is the
adaptation step size αa. Through an iterative process, all filter
tap weights are adjusted during each sample period in the
training sequence. Eventually, the filter will reach a
configuration that minimizes the mean square error between
the equalized signal and the stored reference. As might be
expected, the choice of αa involves a tradeoff between rapid
convergence and residual steady-state error. A too-large
setting for αa can result in a system that converges rapidly on
start-up, but then chops around the optimal coefficient settings Figure 11: Estimated and Calculated Weights of Channel Filter
at steady state.
In this algorithm the input signal considered was noise, and A MMSE algorithm was also tested and presented a clear way
the channel filter parameters were previously determined in a of the implementation of this type of equalizer. The plots
practical experience. Noise (White Noise) was added to the show the equalizer results for 1000 samples and using 500 for
output of the channel. Then it takes N samples for training training. The input is a QAM signal.
sequence (for the plots N=60) and finally the weights of the

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Vol. 4, No. 1 & 2, 2009

5. BASEBAND VS PASSBAND EQUALIZATION

5.1. Examples

Next is an example of a Baseband and PassBand Equalization

in the context of QAM (or multiphase PSK).[8]
In a practical implementation, an equalizer can be realized
either at baseband or at passband. For example Fig. 15
illustrates the demodulation the demodulation of QAM (or
multiphase PSK) by first translating the signal to baseband
and equalizing the baseband signal with an equalizer having
complex-valued coefficients.

cos ωi t
Figure 12: Input Signal
Re ⎡ˆIn ⎤
⎣ ⎦

Im ⎡ˆIn ⎤
⎣ ⎦

Re [ε n ]

Im [ε n ]

sin ωi t

Figure 15: QAM (Multiphase QPSK) signal demodulation

In effect, the complex equalizer with complex-valued (in-

phase and quadrature components) inputs is equivalent to four
parallel equalizers with real-valued tap coefficients as shown
in Fig. 16.
Figure 13: Received Samples

[ Re(cn )]
Re ⎡ ˆI n ⎤
⎣ ⎦

[ Im(cn )]

Im ⎡ ˆI n ⎤
[ Im(cn )] ⎣ ⎦

[ Re(cn )]

Figure 16: Complex-valued baseband equalizer for QAM (Multiphase

QPSK) signals

On the other hand, we may equalize the signal at passband.

Figure 14: Equalized Symbols This is accomplished as shown in Fig. 17 for a two-
It performs a good estimation of the weights of the dimensional signal constellation such as QAM and PSK. The
transversal filter, and provides the optimum values for the received signal is filtered and, in parallel, it is passed through
filter. a Hilbert transformer, called a phase-splitting filter.

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Vol. 4, No. 1 & 2, 2009

e − jω t system to implement in a consumer system. As discussed

below, there are other designs that outperform the DFE in
terms of convergence or noise performance, but these
generally come at the expense of greatly increased system
complexity. Today, most TDMA phones employ DFE running
on fixed-point DSPs such as those in the TMS320C5x [4]
family.

6.3. Equalization used in GSM

e jω t

Figure 17 – QAM or QPSK signal equalization at passband signals An adaptive equalizer is used in the demodulator of the
receiver to compensate for its difficulty in recognizing the
Thus, we have the equivalent of in-phase and quadrature
original bit pattern from the distorted signal. Distortion of the
components at passband, which are fed to a passband complex
signal is caused by the fact that the Doppler shift and the delay
equalizer. Following the equalization, the signal is down-
time for each path varies continuously. As a result, the
converted to a baseband and detected. The error sigbal
channel characteristic (the impulse response) changes over
generated for the purpose of adjusting the equalizer co-
time. The equalizer used for GSM is specified to equalize
efficients is formed at baseband and frequency-translated to
echos up to 16 ms after the first signal received. This
passband as illustrated in Fig. 17.
corresponds to 4.8 km in distance. One bit period is 3.69 ms.
Hence, echos with about 4 bit lengths delay can be
6. APPLICATIONS (EXAMPLES)
compensated [5].
6.1. Equalization in Modem (ADSL) Applications
6.4. Equalization in HSPA and 3GPP
Today, automatic equalization is used on just about all
Receiver-side equalization [6] has for many years been used
modems designed for operation over the switched telephone
to counteract signal corruption due to radio-channel frequency
network. With automatic equalization, a certain initialization
selectivity. Equalization has been shown to provide
time is required to adapt the modem to existing line conditions.
satisfactory performance with reasonable complexity at least
This initialization time becomes important during and after
up to bandwidths corresponding to the WCDMA bandwidth
line outages, since line initial equalization times can extend
of 5MHz [7]. However, if the transmission bandwidth is
otherwise short dropouts unnecessarily. Recent modem
further increased up to, for example 20 MHz, which is the
developments shortened the initial equalization time to
target for the 3GPP Long-Term Evolution, the complexity of
between 15 and 25 ms, whereas only a few years ago a much
straightforward high-performance equalization starts to
longer time was commonly required. After the initial
become a serious issue. One option is then to apply less
equalization, the modem continuously monitors and
optimal equalization, with a corresponding negative impact on
compensates for changing line conditions by an adaptive
the equalizer capability to counteract the signal corruption due
process. This process allows the equalizer to ‘track’ the
to radio-channel frequency selectivity and thus a
frequently occurring line variations that occur during data
corresponding negative impact on the radio-link performance.
transmission without interrupting the traffic flow. On one
The use of specific single-carrier transmission schemes,
9600 bps modem, this adaptive process occurs 2400 times a
especially designed to allow for efficient but still reasonably
second, permitting the recognition of variations as they
low-complexity equalization.
occur[9].
Linear time-domain (frequency-domain) filtering/equalization
implies that linear processing is applied to signals received at
6.2. Equalization for Digital Cellular Telephony
different time instances (different frequencies) with a target to
maximize the post-equalizer SNR (MRC-based equalization),
The direct sequence spreading employed by CDMA (IS-95)
alternatively to suppress signal corruption due to radio-
obviates the need for a traditional equalizer. The TDMA
channel frequency selectivity (zero-forcing equalization,
systems (for example, GSM and IS-54), on the other hand,
MMSE equalization, etc.).
make great use of equalization to contend with the effects of
multipath-induced fading, ISI due to channel spreading,
7. CONCLUSION
additive received noise, and channel-induced spectral
distortion, etc. Because the RF channel often exhibits spectral
nulls, the linear equalizers are not optimal due to their Of particular interest today is the area of digital cellular
tendency to boost noise at the null frequencies. Of the communications, which has seen wide use of fixed-point
nonlinear equalizers, the DFE is currently the most practical DSPs. DSP-based equalizer systems have become ubiquitous

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in many diverse applications including voice, data, and video
communications via various transmission media. Typical
applications range from acoustic echo cancelers for full-
duplex speakerphones to video echo-canceling systems for
terrestrial television broadcasts to signal conditioners for
wireline modems and wireless telephony.
The effect of an equalization system is to compensate for
transmission-channel impairments such as frequency-
dependent phase and amplitude distortion. Besides correcting
for channel frequency-response anomalies, the equalizer can
cancel the effects of multipath signal components, which can
manifest themselves in the form of voice echoes, video ghosts
or Rayleigh fading conditions in mobile communications
channels. Equalizers specifically designed for multipath
correction are often termed echo-cancelers. They may require
significantly longer filter spans than simple spectral equalizers,
but the principles of operation are essentially the same.
This article attempts to familiarize you with some basic
concepts associated with channel equalization and data
communication in general. This report is intended to give an
introduction to equalization, their types and examples and
applications in digital transmission. We have provided a brief
survey of equalization techniques and describe their
characteristics using some examples. Baseband and Passband
equalization has been discussed in terms of Multiphase QPSK.
Some Matlab driven examples also shown using plot to better
understand.
REFERENCES
[1]. B. P. Lathi, Modern Digital and Analog
Communication Systems, Third Edition: Oxford
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[2]. Ziemer, R.E., and Peterson, R.L., Introduction to
Digital Communication, Second Edition, Prentice Hall,
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[3]. J. Kurzweil, An Introduction to Digital
Communications, John Wiley, 2000.
[4]. TMS320C5x User’s Guide, Texas Instruments, 1993.
[5]. GSM Introduction WL9001student guide Lucent
Technologies, 1998.
[6]. J.G. Proakis, Digital Communications, McGraw-Hill,
New York, 2001.
[7]. G. Bottomley, T. Ottosson and Y.-P. Eric Wang, ‘A
Generalized RAKE Receiver for Interference
Suppression’, IEEE Journal on Selected Areas in
Communications, Vol. 18, No. 8, August 2000, pp.
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[8]. Qureshi, S.,“Adaptive Equalization”, IEEE
Communications Magazine, March 1992, pp. 9–16.
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[9]. Peebles, P.Z., Communication System Principles,
Addison-Wesley, 1976.
[10]. Samueli, H., Daneshrad, B., Joshi, R., Wong, B., and
Nicholas, H., “A 64-Tap CMOS Echo
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AUTHORS PROFILE
Kazi Mohammed Saidul Huq received B.Sc. in CSE from
Ahsanullah University of Science & Technology, Bangladesh
in 2003. He obtained his M.Sc. in EE - specialization
Telecommunications from Blekinge Institute of Technology,
Sweden in 2006. Since April 2008, he started working at
Instituto de Telecomunicações, Pólo de Aveiro, Portugal. His
research activities include integration of heterogeneous
wireless systems (in CRRM, cross-layer design, DBWS &
system level simulation paradigm) and integration of RFID.
Atílio Gameiro received his Licenciatura (five years course)
and his PhD from the University of Aveiro in 1985 and 1993
respectively. He is currently a Professor in the Department of
Electronics and Telecommunications of the University of
Aveiro, and a researcher at the Instituto de Telecomunicações
- Pólo de Aveiro, where he is head of group. His main
interests lie in signal processing techniques for digital
communications and communication protocols. Within this
research line he has done work for optical and mobile
communications, either at the theoretical and experimental
level, and has published over 100 technical papers in
International Journals and conferences. His current research
activities involve space-time-frequency algorithms for the
broadband component of 4G systems and joint design of
layers 1 and 2.
Md. Taslim Arefin received B.Sc. in Computer Engineering
from American International University –Bangladesh (AIUB)
in 2005. He obtained his M.Sc. in Electrical Engineering –
Specialization Telecommunications from Blekinge Institute of
Technology (BTH), Sweden in 2008 . At the present time he
is working as lecturer in the Dept. of Computer Science &
Engineering at University of Development Alternative
(UODA), Dhaka, Bangladesh from January, 2009. His
research interest includes BSS, communication engineering
and computer networking like development over cellular
network, routing related issue and wireless communication
etc.
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