Probabilidad de Erlang

ESCUELA SUPERIOR POLITCNICA DE CHIMBORAZO
F FA AC CU UL LT TA AD D D DE E I IN NF FO OR RM MA AT TI IC CA A Y Y E EL LE EC CT TR RO ON NI IC CA A

T TR RA AB BA AJ JO O D DE E I IN NV VE ES ST TI IG GA AC CI I N N

T TE EL LE EF FO ON NI IA A D DI IG GI IT TA AL L

T TE EM MA A: :
D Di is st tr ri ib bu uc ci i n n d de e p pr ro ob ba ab bi il li id da ad d d de e E Er rl la an ng g y y s su us s
a ap pl li ic ca ac ci io on ne es s

NOMBRE:
Sebastin Crdenas (264)
Luis Tumalli (190)

S SE EM ME ES ST TR RE E: : O OC CT TA AV VO O

E ES SC CU UE EL LA A D DE E I IN NG GE EN NI IE ER RI IA A E EL LE EC CT TR RO ON NI IC CA A E EN N
T TE EL LE EC CO OM MU UN NI IC CA AC CI IO ON NE ES S Y Y R RE ED DE ES S
Ingenieria Electronica, Telecomunicaciones y redes
T E L E F O N I A D I G I T A L

1

TEMA: Distribucin de probabilidad de Erlang y sus aplicaciones

OBJETIVOS

OBJETIVO GENERAL

Realizar una investigacin acerca de la distribucin de probabilidad de Erlang
adems de sus principales aplicaciones para entender su importancia en las
redes telefnicas.

OBJETIVOS ESPECFICOS

Definir la probabilidad de Erlang adems de las principales aplicaciones
que esta pueda tener.
Determinar el uso de las probabilidad de Erlang B y C dentro de las redes
de telefona
MARCO TEORICO

La distribucin de probabilidad de Erlang puede definirse como una distribucin
de probabilidad continua, cuya funcin principal se centra en determinar el
nmero de llamadas telefnicas que se podrn realizar en un tiempo por parte
de los operadores. [1]

La distribucin de Erlang se aplicara principalmente en redes telefnicas que
tengan un trfico masivo, es asi que para esta se tiene la ecuacin general que
se plantea a continuacin [1]:

Mediante esta probabilidad se mide el tiempo transcurrido entre la recepcin de
llamadas, sobre todo en caso en los que existen elevados tiempos de espera
para la ocurrencia de cada uno de los eventos, es asi que me mediante este
indicador podremos determinar la probabilidad de retardo y bloqueo, para
conseguir esto se plantean dos tipos de distribuciones de Erlang, que sern las
que ms aportan en la telefona asi estas son:

2

1. Erlang B: Esta nos permitir calcular la probabilidad de bloqueo, si se tiene
una prdida del sistema, es decir no se atiende a una solicitud, por fallas
del sistema o colapso de los servidores, se considerara que le nmero de
usuarios es infinito: [2]

B: probabilidad de bloqueo.
N: nmero de circuitos.
A: intensidad de trfico

2. Erlang C: Esta nos permite calcular la probabilidad de que una llamada
entretrante no sea atendida inmediatamente, de igual manera se
considerara que el nmero de usuarios es infinito y que se tiene un numero
N de fuentes, para calcular esta probabilidad tenemos la siguiente
ecuacin [3]:

Pc: probabilidad de retardo.
N: nmero de circuitos.
A: intensidad de trfico

Tanto la probabilidad de Erlang B como la C sern de gran utilidad para lograr
determinar el rendimiento de la red, sobre todo considerando que el trfico es de
gran volumen.

CONCLUSIONES:
La probabilidad de Erlang se utilizara en redes telefnicas que poseen
trfico de gran volumen o que tienen un elevado nivel de crecimiento.
Erlang B nos permitir conocer el nivel o probabilidad de bloqueo en una
red que se considera de infinitos usuarios, mientras que Erlang C nos da
una idea de la probabilidad de que una llamada sufra retardo cuando
hay fallas del sistema.

RECOMENDACIONES:
Buscar ejemplos en los que se aplique la probabilidad de Erlang para de
esta forma entender el funcionamiento de estas formulas
Utilizar estas ecuaciones en redes que tengan un alto nivel de trfico.

3

REFERENCIAS:
[1] S. Rappaport, Calculation of Some functions Arising in Problems of
Queueing and Communications Traffic, 2013.
[LINK]:http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=&arnumber=1094249&queryText%3
DCalculation+of+Some+functions+Arising+in+Problems+of+Queueing+and+Communication
s+Traffic

[2] Insoo Koo, Erlang Capacity Analysis of CDMA Systems Supporting Voice
and Delay-Tolerant Data Services Under the Delay Constraint
[LINK]:http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=&arnumber=4277079&queryText%
3DErlang+Capacity+Analysis+of+CDMA+Systems+Supporting+Voice+and+Delay-
Tolerant+Data+Services+Under+the+Delay+Constraint

[3] E, Chromy. Erlang C formula and its use in the call centers
[LINK]:https://dspace.vsb.cz/bitstream/handle/10084/84489/AEEE-2011-9-1-
7chromy.pdf?sequence=1

I EEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-27, NO. 1 , JANUARY 1979 249
Calculation of Some Functions Arising in Problems of Queueing
and Communications Traffic
STEPHEN S. RAPPAPORT
Abstract-Useful formulas are developed for the iterative calculation
of the inverse of blocking probability for Erlang loss and Erlang delay
systems, and for the determination of offered traffic and blocking prob-
ability for Erlang loss systems when the carried traffic and number of
servers are given. These formulas which arise in many problems of
queueing and communications traffic can be easily programmed. In
tests they gave rapid convergence to high accuracy over a broad range
of parameter values much greater than the range of usual display.
1. INTRODUCTION
In various problems of queueing or communications traffic
it is often required or convenient to calculate certain inverse
functions for which it is not possible to obtain an explicit
expression. Examples of wide interest include, 1) determina-
tion of offered load and blocking probability for an Erlang loss
system when carried load and number of (servers) trunks are
given, 2) determination of carried load and offered load for an
Erlang loss system when blocking probability and number of
servers are given, and 3) determination of carried load for an
Erlang delay system when blocking probability and number of
servers are given. It is frequently necessary to resort to reading
the required quantities from published graphs or tables [ 1, 21,
sometimes entering these (in reverse) with the quantity usually
taken as the dependent variable. Easily programmable numer-
ical solutions are advantageous in many instances. This is
especially true in view of the advent and proliferation of pro-
grammable calculators and microcomputers. Such solutions are
presented here.
2. CALCULATIONS FOR ERLANG LOSS SYSTEMS
The Erlang B formula gives the probability that a call or
customer is blocked when Poisson traffic is offered to a
group of .?I servers and blocked calls are cleared from the sys-
tem (i.e., M/ G/ h/ b queue). The blocking probability, 0, is given
by B(n, a) where
2 u k / k !
k = O
is the number of servers, and a is the traffic offered to the
server group. Values of B(b, a) can be conveniently calculated
Paper approved by the Editor for Communication Theory of the
IEEE Communications Society for publication without oral presenta-
tion. Manuscript received September 9, 1977; revised June 14, 1978.
This work was supported in part by the National Science Foundation
under Grant ENG-76-09001.
The author is with the Department of Electrical Engineering, State
University of New York, Stony Brook, NY 11794.
using the recursion formula [ 1 ]
with B( 0, U) 2 1.
number of busy servers and can be shown to be
The traffic carried by the system is defined as the average
It is often desired to find the offered traffic a and block-
ing probability when the carried traffic and number of servers
are given. Mathematically this is equivalent to solving the
implicit equation ( 3) for a. If one divides both sides of (3)
by [ l - B(h, a)] the equation can be written in the form
a =f(u), and one can attempt a solution using successive sub-
stitutions [ 3 ] . However, closer examination shows that itera-
tion in this form works well only for small values of a,.
Instead, using the recursion formula ( 2 ) one finds that (3)
can be written in the form u =f(u), where
Differentiation of (4) gives
and we define
The desired solution to (3) (i.e., u,), when U, and 6 are given,
can then be found using the iteration formulas
beginning with uo =a,/[ 1 - (u,/b)l. Rapid convergence is
usually achieved provided that f(a,) #1, and accuracy to
six significant figures is generally attained by stopping the
iteration when
I (ai+ 1 - Ui )/ ui + 1 I <1 0-6. (10)
A recursion formula for B(& a) which can be easily derived
from (2) is
0090-6778/79/OlOO-0249$00.75 O 1979 IEEE
250 I EEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-27, NO. 1 , JANUARY 1979
and the required values of B(6, ai) and B(6, ai) in the itera-
tion formulas (7)-(9) can be generated recursively using (2)
and (1 1) jointly, beginning with B(0, a) =1 and B(0, a) =0.
The blocking probability can then be determined using a,, the
solution to (3), obtained by this iteration procedure. Specifi-
cally, p =B(6, a,.). The iteration scheme described was tested
on a UNIVAC 11 10 computer at approximately 800 points in
the region, lop5 <( c L, / ~) <,999999, 1 <6 <500. Successive
iterates in the scheme were monitored. At several points
(where ~ ( c L, ) 1) for which these iterates were not converg-
ing, the magnitude of 0 was decremented from its previous
iterated value and fixed. The scheme was resumed with the
fixed 0 and this decrementing process was repeated if neces-
sary. In most cases tried the correct solution within an accu-
racy of six significant figures was obtained in less than fifteen
iterations. Often less than five or six iterations were required.
The second calculation described in section 1 is the deter-
mination of a, when 0 and 6 are given. Mathematically this is
equivalent to taking =aC/(l - 0) in (3) and solving the
resulting equation for a,. For A = 1 the solution is a, =0, and
for 6 =2 i t is a, =0 +doz 20( 1 - p), but for arbitrary 6
an explicit solution is unobtainable and a numerical technique
can be used.
The equation to be solved for a, (given 6 and 0) is
F ( u , ) ~ B 6,- -/3=0.
( 1 3 (12)
In this calculation the solution a, must lie in the interval
0 <U, <6. Also for 6 >1, F(u,) in (1 2) has an inflection
point somewhere in this interval. To obtain a scheme appli-
cable over a wide range of 0 and 6 the method of false position
was used [ 3 1 . To reduce the number of iterations required a
simple one dimensional search (requiring at most 20 evalua-
tions of F ) was first used to isolate the solution to within an
interval of b/lOO. Then beginning with c ~ ~ i - 1 and U,i as the
leftmost .and rightmost points of the interval, respectively, a
new value CL,~+~ was generated using
The process was continued until the absolute value of the
normalized difference between the two most recent iterates
was smaller than This results in a nominal accuracy of
six significant figures. The traffic offered to the system can
then be obtained using U =a,/( 1 - 0).
The scheme was programmed on a UNIVAC 11 10 com-
puter and tested over a grid of approximately 800 points
spanning the range lop5 <0 <.999999, 1 <b <500. The
explicit expressions were used for 6 =1 and 2. For larger 6,
six significant figures were obtained at all points tried in no
more than ten iterations after the initial search.
3. CALCULATIONS FOR ERLANG DELAY SYSTEMS
In dealing with Erlang delay systems a similar calculation is
encountered. The Erlang C formula gives the blocking proba-
bility for Poisson arrivals to the server group when blocked
arrivals are held until service is complete (i.e. M/ M/ 6 queue).
The blocking probability is given by [ 1 ]
.I
As before the number of servers is 6, but here all the traffic
a offered to the group is carried. Therefore 0 <d <6. For
arbitrary 6 and 0 it is in general not possible to find an ex-
plicit expression for the inverse function u =Cl ( 6, 0) although
it can be easily found that CI(1, 0) =0, and Cr( 2, 0) =
(0+4-0)/2. For >2 a numerical scheme was used.
C(6, a) can be easily calculated using the identity [ 11
together with the recursive formula (2). The equation to be
solved for u can be written in the form
F(u) P C(6, u) - p =0. (18)
Of aCipl and c(,i that value was saved which produced an F For 6 =1 and 2 the explicit expressions for CI ( 6, 0) were
differing in sign from F(c%,i+l). The smaller of the two values used. For larger values of 6, the method of false position was
saved was identified as u,.~-~ and the larger as aCi. Equation used preceded by an initial search as in the second calculation
(1 3) was then used again. This process was repeated until prox- described previously. The only differences were that (1 8)
imity to the solution was detected by the denominator of (1 3) defines the function used in the initial search, (1 3) should be
becoming smaller than .OOlfi. After this occurred (say on the replaced by
with the constant p given by
and (1 5) should be replaced by
( 15)
P =
u k - a k - 1
c(6, a h ) - c(& a h - 1
IEEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-27, NO. 1 , JANUARY 1979
25 1
Proximity to the solution was detected by the denominator
(1 5) becoming smaller than .OlP. As before the scheme was
tested over a grid of approximately 800 points spanning the
range loA5 <fl< ,999999, 1 < <500. In all cases six signif-
icant figures were obtained in less than ten iterations after the
initial search.
The formulas presented here have been found useful in
many instances (viz. [4] ) and the general approach can be
readily applied to other cases.
REFERENCES
Cooper, R. B. , Introduction t o Queueing Theory, Macmillan Co.:
New York, 1972.
Reference Data for Radio Engineers, 6th ed. Howard W. Sams &
Co.: Indianapolis, Ind. 1975.
Hildebrand, F. B. , Introduction t o Numerical Analysis, McGraw-
Hill: New York, 1956.
Rappaport, S . S., Traffic Capacity of DAMA Systems Using Col-
lision Type Request Channels, ho c . National Telecommunica-
tions Conference, vol. 3, December 1977.
Concerning the Recovery of a Bandlimited Signal or Its
Spectrum from a Finite Segment
RICHARD G. WILEY
Abstruct-The relationship between the iterative techniques of
Sandberg and Papoulis is clarified. The conditions of Sandbergs theorem
are not satisfied in general when the problem is that of extrapolating a
bandlimited signa outside of a known segment (a problem that is ill-
posed). For an interesting special case, the iteration may be applied
and the signal recovered exactly.
In [ 11, we showed that the iteration proposed by Papoulis
[2] for recovery of a bandlimited signal from a finite segment
was of the same form as the contraction mapping of Sandberg
[ 3] , and that the constant in this iteration could be taken as
unity. The key inequality which must be satisfied to apply
Sandbergs theorem when the constant in the iteration is unity
is1
m m
( Qx - Qv) ( x - Y> d t 2 k ( x - yI 2 d t
m
2 (PQx - P Q Y ) ~ d t (1)
Paper approved by the Editor for Communication Theory of the
IEEE Communications Society for publication without oral presenta-
tion. Manuscript received April 27, 1978; revised August 25,
1978.
The author is with the Syracuse Research Corporation, Syracuse,
NY 13210.
In [ 11, the center termof this inequality was not shown.
where:
Q is a distortion operator which transforms a bandlimited
x , y are any pair of (square-integrable) bandlimited signals
P is the projection operator which transforms a square
integrable signal into a bandlimited one
k is a positive constant which depends on Q and the
bandwidth of the subspace of bandlimited signals, but
not on x and y .
signal into a signal that is square integrable
When Q corresponds to multiplication in the time domain by
the function
it is not possible to determine a constant, k , such that the left
inequality of (1) holds for all x, y . We demonstrate this by
means of an example. Suppose x ( t ) is a sinc pulse centered at
+t o and y ( t ) is a sinc pulse centered at - t o. Then, since
m T
I-- L
( Ql x - Ql y ) ( x - Y ) d t = ( x - YI 2 dt , ( 3 )
it is clear that the left side of (1) can be made arbitrarily small
by choosing t o 9 T. At the same time,
is essentially equal to the combined energy of the two sinc
pulses. This shows that while
m m
Lm
( Ql x - Ql v >( x - u) d t 2 ( P Ql x - P QI Y ) ~ dt ,
m
( 5)
nevertheless, Sandbergs theorem does not apply when Q cor-
responds to multiplication by el, i.e., when trying to recover
a bandlimited signal from a finite segment.
Indeed, the failure of the left side of (1) is consistent with
the fact that this problem is ill-posed [4] in the sense of
Hadamard [ 5 1 . This means, roughly, that there are numerous
bandlimited functions which are arbitrarily close to the de-
sired function over the given segment, but which are. quite dif-
ferent from the desireh function outside of the given segment.
If our observation of the finite segment is slightly noisy or cor-
rupted slightly by tlie measurement or recording or sampling
process, the corresponding differences between the extrapolated
signal and the original signal outside the interval of observation
can be very large.
Note that this is due to the nature of the problem of trying
to extrapolate (or estimate the spectrum of) a bandlimited sig-
nal from a finite segment and is not due to the iterative ap-
proach. Any other solution (e.g., the maximum entropy
method [ 21 ) faces the same difficulty.
In practice, what one obtains by applying the iteration is
the (square-integrable) bandlimited signal which matches the
known segment and which has the minimum amouni of ad-
ditional energy (as allowed by the specified bandwidth) out-
side of the known segment. If the original bandlimited function
.0090-6778/79/0100-025 1$00.75 0 1979 IEEE
IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY, VOL. 56, NO. 4, JULY 2007 2375
Erlang Capacity Analysis of CDMA Systems Supporting
Voice and Delay-Tolerant Data Services
Under the Delay Constraint
Insoo Koo, Jeong Rok Yang, and Kiseon Kim
AbstractIn this paper, we analyze the Erlang capacity of a code-
division multiple-access system supporting voice and delay-tolerant data
services. The novelty here is to consider the characteristic of delay-tolerant
trafc in terms of a delay condence. The delay condence is dened as the
probability that a new data call is accepted within the maximum tolerable
delay without being blocked. In this case, the Erlang capacity is conned
not only by the required blocking probability of voice call but also by
the required delay condence of data call. For the performance analysis,
we develop a 2-D Markov model, which is based on the First-Come
First-Serve service discipline, and further present a numerical procedure
to analyze the Erlang capacity. According to the procedure, we can make
a balance between the Erlang capacities with respect to the blocking
probability of voice call and with respect to the delay condence of data
call, in order to accommodate extra Erlang capacity. We demonstrate the
balancing by properly selecting the size of the designated queue for data
trafc.
Index TermsCDMA, delay constraint, Erlang capacity, voice and data.
I. INTRODUCTION
Next-generation mobile communication systems are primarily de-
signed to provide users with multimedia services such as voice, inter-
active data, le transfer, Internet access, and image in the affordable
way as todays wired communication systems do. Multimedia trafc
will have different and multiple quality of service (QoS) requirements.
In terms of system operation, it is a very important task to analyze
the capacity of system that can be supportable while multiple ser-
vice requirements of multimedia trafc are being satised since the
capacity can be used as a measure of system resource. For these
reasons, much effort has been taken to analyze the capacity of a code-
division multiple-access (CDMA) system. Typically, the capacity of
a CDMA system has been dened as the maximum number of users
or the Erlang capacity [1][4]. The former and latter denitions of
the capacity are used for estimating a supportable size of the system
at a time and for measuring the economic usefulness of the system,
respectively [1]. In [1] and [2], the outage probability was presumed to
be the call blocking probability, and the call blocking probabilities of
different trafcs in the system were represented identically. By using a
multidimensional Markov loss model, which is based on the maximum
number of supportable current users, the call-blocking probabilities of
different trafcs were considered separately, and the Erlang capacity
was analyzed with respect to the required blocking probabilities of
different trafcs [8], [9].
Multimedia trafc can be roughly classied into delay-intolerant
and delay-tolerant trafc. To achieve higher capacity using the
delay-tolerant characteristic, delay-tolerant trafc can be queued until
the required resources are available in the system. The blocking
Manuscript received June 19, 2004; revised July 26, 2005 and January 15,
2006. This work was supported in part by the Ministry of Commerce, Industry,
and Energy and in part by Ulsan Metropolitan City through the Network-based
Automation Research Center at the University of Ulsan. The review of this
paper was coordinated by Prof. C. Lin.
I. Koo is with the University of Ulsan, Ulsan 680-749, Korea.
J. R. Yang is with the Korean Intellectual Property Ofce, Daejeon 302-701,
Korea.
K. Kim is with the Kwang-Ju Institute of Science and Technology, Gwangju
500-712, Korea.
Digital Object Identier 10.1109/TVT.2007.897655
probability and the average delay have been typically considered to be
performance measures for the delay-tolerant trafc [8], [10]. However,
a more meaningful measurement for delay-tolerant trafc is the delay
condence rather than the average delay, where the delay condence is
dened as the probability that a new data call gets a service within the
maximum tolerable delay requirement without being blocked. Noting
that the previous works [1], [8][10] have not considered the delay
condence when evaluating the Erlang capacity, in this paper, we adopt
the delay condence as a performance measure of delay-tolerant traf-
c. Further, we analyze Erlang capacity of a CDMA system supporting
voice and data services since voice and data calls are typical delay-
intolerant and delay-tolerant trafc, respectively. Here, the Erlang
capacity is dened as a set of average offered trafc loads of voice
and data calls that can be supported in the system while the required
blocking probability of voice call and the required delay condence
of data call are being satised simultaneously. To analyze the Erlang
capacity, we develop a 2-D Markov model, which is based on the First-
ComeFirst-Serve (FCFS) service discipline, where the queue with
nite size is exclusively allocated for delay-tolerant data calls. Based
on the Markov model, we present a numerical procedure to analyze the
call blocking probabilities of voice and data trafc and the delay distri-
bution and delay condence of data trafc, which are performance ma-
trices necessary to analyze the Erlang capacity. In addition, a procedure
selecting the proper size of queue length for data trafc is suggested,
in order to accommodate extra Erlang capacity in the system.
The remainder of this paper is organized as follows. In Section II,
we describe system model and a call admission control (CAC) scheme.
In Section III, we develop a 2-D Markov model and analyze the
blocking probabilities of voice and data calls. In Section IV, we present
an analytical approach to derive the cumulative distribution function
(CDF) of delay and delay condence of data calls in order to analyze
the Erlang capacity. In Section V, we consider a numerical example.
Finally, we draw conclusions in Section VI.
II. SYSTEM MODEL
In this paper, we are concerned with the trafc analysis of the
CDMA systems. In order to focus on the trafc analysis of the CDMA
systems, we consider the number-based admission control with pre-
determined CAC threshold, and as a reference for the predetermined
CAC threshold, we utilize the link capacity that details the numbers of
mobiles of each class in each cell that the system operator should allow
in order to maintain an acceptable QoS. The link capacity reects the
CDMA aspects and gives performance guarantees that overcome the
variability in the interference levels that are characteristic of CDMA
networks such that the CDMA mobile network, operating within the
admissible region described previously, has a very similar form to
a circuit-switched network. The similarity allows the existing trafc
modeling techniques and network management strategies for general
loss networks to be applied to CDMA systems.
Let us rst consider the link capacity. In CDMA systems, although
there is no hard limit on the number of concurrent users in the link
level, there is a practical limit on the number of supportable concurrent
users in order to control the interference among users having the same
pilot signal; otherwise, the system may fall into the outage state, where
QoS requirements of users cannot be guaranteed. In order to consider
the link capacity of a CDMA system, let Q
v
and Q
d
be the link
qualities such as frame error rate that individual voice and data users
experience in the CDMA, respectively, and let Q
v,m
and Q
d,m
be a
set of minimum link quality level of each service. Then, for a certain
set of system parameters, such as service quality requirements and
0018-9545/$25.00 2007 IEEE
2376 IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY, VOL. 56, NO. 4, JULY 2007
link propagation model, the admissible region of CDMA system with
respect to the simultaneous number of users satisfying service quality
requirements in the sense of statistic S can be dened as
S = (i, j)[P
r
(Q
v
Q
v,m
and Q
d
Q
d,m
) %
= (i, j)[0 f
n
(i, j) 1 and i, j Z
+
(1)
where i and j are admissible number of voice and data users in the
CDMA, respectively, % is the system reliability dened as minimum
requirement on the probability that the link quality of the current users
in the system is larger than the minimum link quality level, which
is usually given between 95% and 99%, and f
n
(i, j) is the normal-
ized capacity equation. In the single cell case, the f
n
(i, j) can be
given as [4]
f
n
(i, j) =
v
i +
d
j, i and j 0 (2)
where
v
= ((W/R
v
q
v
) + 1)
1
and
d
= ((W/R
d
q
d
) + 1)
1
,
which are the amount of system resources that are used by one
voice and one data user, respectively. W is the allocated frequency
bandwidth. q
v
and q
d
are the required bit energy to interference
power spectral density ratio for voice and data calls, respectively, to
achieve the target bit error rate at the base station (BS). R
v
and R
d
are the required information data rates of the voice and data calls,
respectively. For the link capacity of a multiple cell CDMA system,
the effect of other cell interference can be included in (1), and the
normalized capacity equation f
n
(i, j) is given as [4]
f
n
(i, j) =
v
i +
d
j +E(z) + 2.33
_
(var(z)) (3)
where the termE(z) + 2.33
_
(var(z)) includes the effect of intercell
interference on the link capacity. The link capacity of the multiple cell
case is conned by a lower hyperplane than that of the single cell case
due to the effect of the other cell interference. Such linear link capacity
bounds on the total number of users of each class, which can be sup-
ported simultaneously while maintaining adequate QoS requirements,
are commonly found in other literature on CDMA systems supporting
multiclass services [5], [6]. Further, provided the network state lines
within the admissible region, then the QoS requirement of each user
will be satised with % reliability. Here, we are concerned with the
trafc analysis of the CDMA systems operating with the specied
CAC algorithm such that we consider the single cell case in order to
focus on the trafc analysis of the CDMA systems.
Each user is classied by QoS requirements such as the required
information data rate and the required bit energy to interference
spectral density ratio, and all users in same service group have the same
QoS requirements. Equations (2) and (3) indicate that the calls with
different services take different amount of system resources according
to their QoS requirements. We also assume that the system employs
a circuit switching method to handle the transmission of voice and
data calls, which means that once a call request is accepted in the
system, the call occupies the required amount of system resources and
transmits the information without any delay throughout the duration of
the call.
In the aspects of network operation, it is of vital importance to
set up a suitable policy for the acceptance of an incoming call in
order to guarantee a certain QoS. In general, CAC policies can be
divided into two categories: 1) number-based CAC (NCAC) and
2) interference-based CAC (ICAC) [7]. NCAC implies that a call will
be accepted or not, depending on the condition whether the number of
concurrent users is greater than a threshold. In the case of ICAC, a BS
determines whether a new call is acceptable or not by monitoring the
interference level on a call-by-call basis, whereas the NCAC utilizes
a predetermined CAC threshold. In this paper, we adopt an NCAC-
type CAC due to its simplicity, although the NCAC generally suffers
from a slight performance degradation over the ICAC [7]. That is, the
CDMA mobile network, which is operated according to the NCAC
with the threshold stipulated by link capacity, has a very similar form
to a circuit-switched network. The similarity allows the existing trafc
modeling techniques and network management strategies for general
loss networks to be applied to CDMA systems. Here, we set the
capacity bound, which is stipulated by (2), as a predetermined CAC
threshold. However, it is noteworthy that the link capacity reects the
feature that CDMA is interference limited. Further, we consider the
queue with the nite length of K (K 1) for delay-tolerant data
trafc to exploit its delay-tolerant characteristic, where K denotes
the queue length, and assume that the service discipline is FCFS. In
the service model, a voice call experiences call blocking when there
is not enough resource to accommodate the voice call and receives
zero queuing delay. On the other hand, a data call experiences call
blocking when there is not enough resource and buffer space to
accommodate the data call and receives some queuing delay. Based on
these assumptions, the CAC, for the case
d
>
v
, can be summarized
as follows.
If
v
i +
d
j 1
d
, then both new voice and new data calls
are accepted.
If 1
d
<
v
i +
d
j 1
v
, then new voice calls are ac-
cepted, and new data calls are queued.
If 1
v
<
v
i +
d
j 1 + (K 1)
d
, then new voice calls
are blocked, and new data calls are queued.
If
v
i +
d
j > 1 + (K 1)
d
, then both new voice and new
data calls are blocked.
Here, we set a voice channel to occupy one basic channel. In this
case, the number of total basic channels in the system is 1/
v
, and
the number of basic channels required by one data call is
d
/
v
,
respectively. It is also noteworthy that the number of total basic
channels in the system and the number of basic channels required by
one data call are integer numbers in time-division multiple access or
frequency-division multiple access, whereas they are real numbers in
CDMA systems [9].
In order to analyze the performance of the system under the CAC
policy, the arrivals of voice and data calls are assumed to be distributed
according to independent Poisson processes with the average arrival
rate
v
and
d
, respectively. The service times of voice and data calls
are assumed to be exponentially distributed with the average service
time 1/
v
and 1/
d
, respectively. Then, the offered trafc loads of
voice and data calls are expressed as
v
=
v
/
v
and
d
=
d
/
d
,
respectively.
III. MARKOV CHAIN MODEL AND BLOCKING PROBABILITY
In this section, we develop an analytical model to determine the
blocking probabilities of voice and data calls. The model will also be
utilized to analyze the delay distribution of data call in Section IV.
According to the CAC rule based on the number of concurrent users
when the queue with the nite length of queue size K 1 for delay-
tolerant data trafc is employed, the set of possible admissible states is
given as follows at the medium access control (MAC) layer where call
admission process is performed:
S
=
_
(i, j)[0 i
1
v
, j 0,
v
i +
d
j 1 +
d
K
_
. (4)
Fig. 1. State transition diagram for the case that
d
>
v
.
Here, it is noteworthy that (2) corresponds to the link capacity of a
CDMA system supporting voice and data services in the link layer,
whereas (4) indicates the set of possible admissible states at the MAC
layer. For the case that the buffer is not considered in the BS (K = 0),
the link capacity corresponds to the threshold for the admission control
process at MAC layer directly. Here, we consider the case K 1.
Further, the possible admissible states can be divided into ve dis-
tinct regions, which are represented by the following sets, respectively:
A
=(i, j)[0
v
i +
d
j 1
d
B
=(i, j)[1
d
<
v
i +
d
j 1
v
C
=(i, j)[1
v
<
v
i +
d
j 1
D
=(i, j)[1 <
v
i +
d
j 1 +
d
(K 1)
E
=(i, j)[1 +
d
(K 1) <
v
i +
d
j 1 +
d
K .
(5)
Fig. 1 shows these ve distinct regions and a typical call-level state
transition example for each region. Noting that total rate of owing
into a state (i, j) is equal to that of owing out, we can get the steady-
state balance equation for each state as follows:
Rate-In =Rate-Out
Rate-In =a P
i+1,j
+
b P
i,j+1
+c P
i1,j
+

d P
i,j1
Rate-Out =(
i +
j +
k +
l) P
i,j
, for all states (6)
where the state transition rates a,
b, c,

d,
i,
j,

k, and
l involved in (6)
can be given by
a transition rate from state (i + 1, j) to state (i, j)
=
_
(i + 1)
v
, (i, j)
S
0, otherwise
(7)
b transition rate from state (i, j + 1) to state (i, j)

=
_
_
(j + 1)
d
, (i, j)
A
j
d
, (i, j)
B
,
C
(1
v
i)
1
d
|
d
, (i, j)
D
0, otherwise
(8)
Fig. 2. Steady-state balance equations corresponding to the voice/data CDMA
system for the case that
d
>
v
.
c transition rate from state (i 1, j) to state (i, j)
=
_
v
, (i, j)
A
,
B
,
C
0, otherwise
(9)
d transition rate from state (i, j 1) to state (i, j)

=
_
d
, (i, j)
S
0, otherwise
(10)
i transition rate from state (i, j) to state (i + 1, j)

=
_
v
, (i, j)
A
,
B
0, otherwise
(11)
j transition rate from state (i, j) to state (i, j + 1)

=
_
d
, (i, j)
A
,
B
,
C
,
D
0, otherwise
(12)
k transition rate from state (i, j) to state (i 1, j)

=
_
i
v
, (i, j)
S
0, otherwise
(13)
l transition rate from state (i, j) to state (i, j 1)

=
_
j
d
, (i, j)
A
,
B
,
C
(1
v
i)
1
d
|, otherwise.
(14)
Fig. 2 summarizes the steady-state balance equations for the state
transition diagramaccording to the region that the current state belongs
to. If the total number of all possible states is n
s
, the balance equations
become (n
s
1) linearly independent equations. With these (n
s
1)
equations and the normalized equation

(i,j)
S
P
i,j
= 1, a set
of n
s
linearly independent equations for the state diagram can be
formed as
Q = P (15)
where Q is the coefcient matrix of the n
s
linear equations, is the
vector of state probabilities, and P = [0, , 0, 1]
T
. The dimensions
of Q, , and P are n
s
n
s
, n
s
1, and n
s
1, respectively. By
solving = Q
1
P, we can obtain the steady-state probabilities of all
states [8].
Based on the CAC rule, a new voice call will be blocked if the chan-
nel resources are not enough to accept the call, and the corresponding
blocking probability for voice calls is given by
P
b
v
=
(i,j)
(nv,blo)
P
i,j
(16)
where
(nv,blo)
= (i, j)[
v
i +
d
j > 1
v
. (17)
(nv,blo)
is composed of the regions C, D, and E in Fig. 1. Similarly,
a new data call will be blocked if the queue is full, and the blocking
probability for data calls is given by
P
b
d
=
(i,j)
(nd,blo)
P
i,j
(18)
where
(nd,blo)
= (i, j)[
v
i +
d
j > 1 +
d
(K 1) . (19)
(nd,blo)
corresponds to the region E in Fig. 1.
IV. ERLANG CAPACITY ANALYSIS
In this section, we analyze the Erlang capacity of CDMA system
supporting voice and delay-tolerant services under the delay con-
straint. For this purpose, we will derive the CDF of delay and the delay
condence of data trafc. First, let us derive the CDF of delay , which
is based on the Markov chain model depicted in Fig. 1. The delay is
dened as the time that a data call waits in a queue until being accepted
in the system. For the convenience of analysis, we separate the CDF of
delay into two parts, corresponding to discrete and continuous parts of
the random variable , i.e.,
F
d
(t) = Pr t = F
d
(0) +G(t) (20)
where F
d
(0) = Pr 0, and G(t) represents the continuous part
of the delay. At rst, the discrete part F
d
(0) represents the case when
the delay is zero, and it can be calculated as follows:
F
d
(0) = Pr 0 = Pr = 0
=
(i,j)
(nd,acc)
P
i,j
(21)
Fig. 3. Set of states representing the admissible numbers of voice and data calls for the case that W = 1.25 MHz, q
v
= 7 dB, q
d
= 7 dB, R
v
= 9.6 kb/s,
R
d
= 19.2 kb/s, and K = 3.
where
(nd,acc)
is the acceptance region of new data calls, which is
given as
(nd,acc)
= (i, j)[
v
i +
d
j 1
d
(22)
and
P
i,j
=
P
i,j
1 P
b
d
. (23)
P
i,j
represents the probability that there are i voice and j data calls
in the system just before a new data call is admitted. If the state (i, j)
belongs to the blocking region of new data calls
(nd,blo)
, the call will
be blocked.
To investigate the continuous part of delay G(t), let (i
, j
) denote
the number of calls excluding the number of service-completed calls
within time from (i, j). Consider the case that (i, j) belongs to
the queueing region of new data calls just before a new data call is
admitted, where the queueing region of new data calls is given as
(nd,que)
= (i, j)[1
d
<
v
i +
d
j 1 + (K 1)
d
. (24)
In order for a new data call to be accepted within the time t according
to the FCFS service discipline, (i
, j
) should fall into the acceptance

region of new data calls within the time t. G(t) is the sum of the
probabilities of all cases that a state (i, j) in
(nd,que)
changes into
(i
, j
) in
(nd,acc)
within the time t, which can be expressed as
G(t) =
(i,j)
(nd,que)
Pr
_
(i
, j
)
(nd,acc)
within time t[
the system state is (i, j)
just before a new data call
is admitted
_
P
i,j
=
(i,j)
(nd,que)
t
_
0
w
(i,j)
()d P
i,j
(25)
where w
(i,j)
() is the delay distribution for the state (i, j), and it
represents the probability of a new data call being accepted within time
, given that the system state is (i, j) just before the call is admitted.
For example, Fig. 3 shows the set of states representing the admissible
numbers of voice and data calls for the case that W = 1.25 MHz,
q
v
= 7 dB, q
d
= 7 dB, R
v
= 9.6 kb/s, R
d
= 19.2 kb/s, and K = 3.
Consider the case that there are 17 voice calls and seven data calls in
the system just before a new data call is admitted. In this case, the
state (17, 7) can change into (i
, j
) in
(nd,acc)
through many paths
in order for a new data call to be accepted. For example, if (i
, j
) is
(17, 4), no voice call and three data calls are service completed during
the time , and if (i
, j
) is (16, 4), one voice call and three data calls

are service completed. For the more general case where k voice calls
Fig. 4. State transition paths for analyzing the delay distribution. (a) The case that no voice call and (j (1
v
i)/
d
+ 1) data calls are service completed
within time , given that the system state is (i, j) just before a new data call is attempted. (b) The case that one voice call and (j (1
v
(i 1))/
d
+ 1)
data calls are service completed within time . (c) The case that two voice calls and (j (1
v
(i 2))/
d
+ 1) data calls are service completed within
time . (d) The case that k voice calls and (j (1
v
(i k))/
d
+ 1) data calls are service completed within time .
get service completed during the time , the delay distribution for the
state (i, j) can be expressed as
w
(i,j)
() =
I
k=0
w
(i,j)
k
() (26)
where
I = min
_
i, i
_
1
d
(1 +j)
v
__
. (27)
w
(i,j)
k
() represents the delay distribution multiplied by the prob-
ability that k voice calls get service completed during the time ,
given that the system state is (i, j) just before a new data call is
admitted. I is the maximum number of service-completed voice calls
during the change, which happens when voice calls are only service
completed.
The paths where the state (i, j) in
(nd,que)
changes into (i
, j
)
in
(nd,acc)
can be generalized, as in Fig. 4. Since the service time
distribution is memoryless and the delay distribution is independent
of the current arrival, w
(i,j)
k
() is the convolution of k independent
exponential random variables, where k corresponds to the number
of service-completed voice calls [11]. Since the Laplace transforms
of w
(i,j)
k
() is equal to the product of the Laplace transforms of
exponential distributions, the Laplace transform of w
(i,j)
0
(), for the
case in Fig. 4(a), can be expressed as
W
(i,j)
0
(s) =
_
_
_
1
v
i
d
_
d
i
v
+
_
1
v
i
d
_
d
_
_
j
_
1
v
i
d
_
+1
_
_
_
1
v
i
d
_
d
s +
_
1
v
i
d
_
d
_
_
j
_
1
v
i
d
_
+1
. (28)
The rst term of W
(i,j)
0
(s) in (28) represents the probability for
k = 0, which corresponds to the probability that the state (i, j) in
(nd,que)
is changed into (i
, j
) in
(nd,acc)
, as in Fig. 4(a). In (28),
the exponent (j (1
v
i)/
d
| + 1) corresponds to the required
number of service-completed data calls for the new data call to be
accepted. The second term of W
(i,j)
0
(s) in (28) comes from the
product of the Laplace transforms of exponential distributions of
service time of the service-competed data calls.
For the case that k = 1 which corresponds to Fig. 4(b), there are
J
1
different paths, and w
(i,j)
1
() is expressed as the sum of delay
distributions multiplied by the probability that the path is selected out
of all paths. The Laplace transform of w
(i,j)
1
() can be expressed in
(29), shown at the bottom of the page, where
J
1
=
_
_
j
_
1
v
(i1)
d
_
, if
_
1
v
(i1)
d
_
=
_
1
v
i
d
_
j
_
1
v
(i1)
d
_
+ 1, otherwise.
(30)
In the case of one service-completed voice call, the number of service-
completed data calls should be (j (1
v
(i 1))/
d
| + 1) for a
new data call to be accepted. J
1
is selected to avoid the path for the
case of k = 0, and, for example, it takes the path (17, 7) (17, 4)
(16, 4) in Fig. 3.
By expanding the previous results to the general case of k service-
completed voice calls, W
(i,j)
k
(s) can be obtained in (31), shown at
the bottom of the page, where J
k
can be obtained in (32) and (33),
shown at the bottom of the page, and
D
a
(s) =
_
_
_
1
v
(ia)
d
_
d
(i a)
v
+
_
1
v
(ia)
d
_
d
_
_
_
_
_
1
v
(ia)
d
_
d
s +
_
1
v
(ia)
d
_
d
_
_
. (34)
W
(i,j)
1
(s) =
_
_
i
v
i
v
+
_
1
v
i
d
_
d
_
_
_
i
v
s +i
v
_
J
1
j
1
=0
_
_
_
_
_
1
v
i
d
_
d
i
v
+
_
1
v
i
d
_
d
_
_
j
1
_
_
_
1
v
i
d
_
d
s +
_
1
v
i
d
_
d
_
_
j
1
_
_
_
1
v
(i1)
d
_
d
(i 1)
v
+
_
1
v
(i1)
d
_
d
_
_
j
_
1
v
(i1)
d
_
+1j
1
_
_
_
1
v
(i1)
d
_
d
s +
_
1
v
(i1)
d
_
d
_
_
j
_
1
v
(i1)
d
_
+1j
1
_
_
(29)
W
(i,j)
k
(s) =
_
_
(D
0
(S))
j
_
1
v
i
d
_
+1
, k = 0
k1
a=0
V
a
(S)
J
k
j
1
=0
J
k
j
2
=j
1

J
k
j
k
=j
k
1
_
(D
0
(S))
j
1
(D
1
(S))
j
2
j
1
(D
2
(S))
j
3
j
2
. . .
(D
k1
(S))
j
k
j
k1
(D
k
(S))
j
_
1
v
(ik)
d
_
+1j
k
_
, otherwise
(31)
J
k
=
_
_
_
j
_
1
v
(ik)
d
_
, if
_
1
v
(ik)
d
_
=
_
1
v
(i(k1))
d
_
j
_
1
v
(ik)
d
_
+ 1, otherwise
(32)
V
a
(s) =
_
_
(i a)
v
(i a)
v
+
_
1
v
(ia)
d
_
d
_
_
_
(i a)
v
s + (i a)
v
_
(33)
The left term of V
a
(s) in (33) is the probability that one voice call
is service completed among (i a) voice and (1
v
(i a))/
d
|
data calls in the service state, and the right term is Laplace transform
of the time distribution for the voice call to be service completed.
On the other hand, the left term of D
a
(s) in (34) is the probabil-
ity that a data call is service completed among (i a) voice and
(1
v
(i a))/
d
| data calls in the service state, and the right term
is Laplace transform of the service time distribution for the data call
to be service completed. It is noteworthy that the probability of voice
or data call being service completed, and the time distribution for a
call to be service completed can be represented by the number of
voice calls for given average service times of voice and data calls. It
comes from the fact that the number of data calls in the service state
is determined by the number of voice calls. w
(i,j)
k
() is the sum of
delay distribution of all possible paths for k service-completed voice
calls multiplied by the probability that each path is selected. J
k
is a
parameter to prevent w
(i,j)
k
() from including the path for (k 1)
service-completed voice calls. Finally, we can get w
(i,j)
k
() from the
inverse Laplace transform of W
(i,j)
k
(s).
Substituting w
(i,j)
k
() into w
(i,j)
() and then successively substi-
tuting w
(i,j)
() into G(t), the CDF of delay can be calculated as
F
d
(t) =
(i,j)
(nd,acc)
P
i,j
+
(i,j)
(nd,que)
t
_
0
I
k=0
L
1
_
W
(i,j)
k
(s)
_
P
i,j
d (35)
where L
1
denotes the inverse Laplace transform.
For a delay-tolerant trafc, an important performance measure is
related with the delay requirement. Typically, the delay requirement
of data calls is that the system should provide the required services to
users within the maximum tolerable delay. Considering that the service
behavior is randomly characterized, we need to introduce the delay
condence, which is dened as the probability that new data calls are
accepted within the maximum tolerable delay without being blocked,
and further, we formulate the delay condence as follows:
P
c
(1 P
b
d
) F
d
(
max
) (36)
where
max
is the maximum tolerable delay requirement. Here, note
that the delay condence is related to not only the CDF but also the
blocking probability of data calls.
As a system level performance measure, we utilize Erlang capacity,
which is dened as a set of supportable offered trafc loads of voice
and data that can be supported while service requirements are satised,
where we consider the required call blocking probability for voice
calls and the required delay condence for data calls, as the service
requirements. Then, the Erlang capacity of CDMA system supporting
voice and data services can be formulated as follows:
C
Erlang

_
(
v
,
d
)[P
b
v
P
b
v,req
, P
c
P
c
req
_
(37)
where P
b
v,req
is the required blocking probability for voice calls,
and P
c
req
is the required delay condence for data trafc. We also
dene a set of supportable offered trafc loads of voice and data
that are conned by the required call blocking probability of voice
P
b
v,req
as the voice-limited Erlang capacity and that by the required
delay condence of data P
c
req
as the data-limited Erlang capacity,
respectively. Then, the Erlang capacity of the system is determined as
the overlapped region of the voice-limited Erlang and the data-limited
Erlang.
TABLE I
SYSTEM PARAMETERS FOR THE NUMERICAL EXAMPLE: A CDMA
SYSTEM SUPPORTING VOICE AND DELAY-TOLERABLE DATA SERVICES
Fig. 5. Voice-limited Erlang capacity and data-limited Erlang capacity with-
out allowing the delay in queue. (i) and (ii) represent the voice-limited Erlang
and the data-limited Erlang, and the Erlang capacity corresponds to the over-
lapped region limited by (i) and (ii), where P
b
v,req
= 1%, P
c
req
= 99%, and
K = 0.
V. NUMERICAL EXAMPLE
For a numerical example, we consider a CDMA system supporting
voice and delay-tolerant data services. The considered parameters are
summarized in Table I and the normalized delay
n
means that the
delay is normalized by average service time 1/
d
. Fig. 5 shows
the voice-limited Erlang capacity and the data-limited Erlang capacity
when K = 0. Lines (i) and (ii) represent the voice-limited Erlang
capacity and the data-limited Erlang capacity, respectively, where
P
b
v,req
= 1% and P
c
req
= 99%. The Erlang capacity is determined
as the overlapped region limited by lines (i) and (ii) to satisfy both
service requirements for voice and data calls at the same time. For
the case that there is no queue (K = 0), the CDF of delay at the
Fig. 6. Voice-limited Erlang capacity and data-limited Erlang capacity for
different values of the required delay condence P
c
req
, where P
b
v,req
= 1%,
n
max
= 0.1, and K = 1.
maximum tolerable delay F
d
(
max
) is given 1 such that the delay
condence P
c
becomes (1 P
b
d
), and the required delay condence
of 99% corresponds to the required blocking probability of 1% of
data. The Erlang capacity in Fig. 5 corresponds to that analyzed in [9]
for the blocking probabilities for voice and data trafc. Fig. 5 shows
that the Erlang capacity is mainly determined by the data-limited
Erlang capacity. The gap between the voice-limited Erlang capacity
and the data-limited Erlang capacity comes from the difference in the
service requirements for voice and data calls. In this case, the data-
limited Erlang capacity is lower than the voice-limited Erlang capacity
for the same blocking probability because a data call requires more
system resources than a voice call. In reality, the data calls have a
distinct characteristic that may allow some delay; however, it was
presented that the data calls behave like the voice calls, which are
eventually blocked when there is no instantly available resource.
In order to increase the Erlang capacity, a proper tradeoff is required
between the voice-limited Erlang capacity and the data-limited Erlang
capacity. One of the methods to get the tradeoff is to use queueing
for delay-tolerant data calls. Fig. 6 shows the voice-limited Erlang
capacity and the data-limited Erlang capacity for different values of the
required delay condence P
c
req
, where P
b
v,req
= 1%,
n
max
= 0.1,
and K = 1. In this case, a new data call can be queued into the
nite buffer of the size K until the required resources are available.
From Fig. 6, we know that the Erlang capacity is mainly determined
by the data-limited Erlang capacity, where P
c
req
= 99%. However,
the data-limited Erlang capacity gradually increases as the required
delay condence P
c
req
decreases such that the Erlang capacity is
limited by the voice-limited Erlang capacity when P
c
req
is given less
than 98%.
Fig. 7 shows the voice-limited Erlang capacity and the data-limited
Erlang capacity for different values of the maximum tolerable delay
n
max
, where P
b
v,req
= 1%, P
c
req
= 99%, and K = 1. The Erlang
capacity is mainly determined by the data-limited Erlang capacity
when the maximum tolerable delay
n
max
is less than 0.1. As
n
max
increases, i.e., when the delay allowance for data calls increases, the
data-limited Erlang capacity also increases. Fig. 7 shows that the
Erlang capacity is limited by the voice-limited Erlang capacity when
n
max
is more than 0.4.
Figs. 811 show the effect of the queue size on the Erlang capacity,
where P
b
v,req
= 1%, P
c
req
= 99%, and
n
max
= 0.1. In the gures,
Fig. 7. Voice-limited Erlang capacity and data-limited Erlang capacity for
different values of the maximum tolerable delay
n
max
, where P
b
v,req
= 1%,
P
c
req
= 99%, and K = 1.
Fig. 8. Effect of the queue size on the Erlang capacity, where P
b
v,req
=
1%, P
c
req
= 99%,
n
max
= 0.1, and K = 0, where the solid and dotted
lines represent the voice-limited Erlang capacity and the data-limited Erlang
capacity, respectively.
the solid and dotted lines represent the voice-limited Erlang capacity
and data-limited Erlang capacity, respectively. We know that the voice-
limited Erlang capacity decreases as the queue size increases, which
comes fromthe fact that the call blocking probability of voice increases
for a larger queue size. On the other hand, the data-limited Erlang
capacity increases until the queue size becomes 2, and after that, it
decreases for a larger queue size. It comes from the fact that the
call blocking probability of data and the CDF of delay, respectively,
decreases as the queue size increases. Noting that the delay condence
depends on not only the blocking probability of data call decreases but
also the CDF of delay, we know that the variation of the data-limited
Erlang capacity according to the queue size mainly comes from the
mutual effects between the improvement in the blocking probability of
data calls and the decrease of the CDF of delay. Figs. 811 also show
that the Erlang capacity when K = 3 is less than that when K = 0,
which means that the queue size should be properly selected to make
b
v,req
=
1%, P
c
req
= 99%,
n
max
b
v,req
=
1%, P
c
req
= 99%,
n
max
a balance between the voice-limited Erlang capacity and the data-
limited Erlang capacity and, further, to accommodate more Erlang
capacity. We observe that the optimum queue size is 1 in the case of
the numerical example, with respect to the Erlang capacity.
It will be useful to quantify the optimum size of the buffer for
a general case while taking all the necessary model parameters into
account. However, it is very complicated to nd out such general
solution for a general case, particularly with an analytical fashion. In
this paper, we only subsequently present an example study to show a
way to select the proper queue size for a specic case. The procedure,
however, can be applied to any case.
VI. CONCLUSION
In this paper, we have analyzed the Erlang capacity of a CDMA
system supporting voice and delay-tolerant data services by particu-
b
v,req
=
1%, P
c
req
= 99%,
n
max
larly considering a new performance metric of delay-tolerant trafc,
which is called delay condence. For the performance analysis, we
developed a 2-D Markov model, which is based on the FCFS service
discipline, and presented a numerical procedure to analyze the Erlang
capacity. As a result, for the case that there is no queue for data calls,
it was observed that the Erlang capacity is mainly determined by the
data-limited Erlang capacity since one data call requires more system
resources than one voice call. For the case that we consider nite size
buffer for data calls, the data-limited Erlang capacity increases as the
maximum tolerable delay increases or the required delay condence
decreases. Further, the Erlang capacity is mainly limited by the voice-
limited Erlang capacity if the required delay condence and the
maximum tolerable delay requirements go beyond certain values. By
observing the Erlang capacity according to the queue size, we showed
that the queue size should be properly selected to make a balance
between the voice-limited Erlang capacity and the data-limited Erlang
capacity. For the numerical example case, we demonstrated that a
proper queue size was selectable with respect to the Erlang capacity
under a given delay constraint.
REFERENCES
[1] A. M. Viterbi and A. J. Viterbi, Erlang capacity of a power-controlled
CDMA system, IEEE J. Sel. Areas Commun., vol. 11, no. 6, pp. 892
900, Aug. 1993.
[2] A. Sampath, N. B. Mandayam, and J. M. Holtzman, Erlang capacity of
a power controlled integrated voice and data CDMA system, in Proc.
IEEE Veh. Technol. Conf., 1997, pp. 15571561.
[3] A. Sampath, P. S. Kumar, and J. M. Holtzman, Power control and
resource management for a multimedia CDMA wireless system, in
Proc. IEEE Int. Symp. Pers., Indoor Mobile Radio Commun., 1995,
pp. 2125.
[4] J. R. Yang, Y. Y. Choi, J. H. Ahn, and K. Kim, Capacity plane of CDMA
system for multimedia trafc, Electron. Lett., vol. 33, no. 17, pp. 1432
1433, Aug. 1997.
[5] J. Evans and D. Everitt, On the teletrafc capacity of CDMA cellu-
lar networks, IEEE Trans. Veh. Technol., vol. 48, no. 1, pp. 153165,
Jan. 1999.
[6] J. Evans and D. Everitt, Effective bandwidth-based admission control
for multiservice CDMA cellular networks, IEEE Trans. Veh. Technol.,
vol. 48, no. 1, pp. 3646, Jan. 1999.
[7] Y. Ishikawa and N. Umeda, Capacity design and performance of call ad-
mission control in cellular CDMA systems, IEEE J. Sel. Areas Commun.,
vol. 15, no. 8, pp. 16271635, Oct. 1997.
[8] I. Koo, E. Kim, and K. Kim, Erlang capacity of voice/data DS-CDMA
systems with prioritized services, IEICE Trans. Commun., vol. E84-B,
no. 4, pp. 716726, Apr. 2001.
[9] F. A. Cruz-Perez, D. Lara-Rodriguez, and M. Lara, Performance analysis
of the fractional channel reservation in TDMA and CDMA integrated
services networks, in Proc. IEEE Veh. Technol. Conf.Spring, 2001,
pp. 10071011.
[10] B. S. Bae, K. T. Jin, and D. H. Cho, Performance analysis of an integrated
voice/data CDMA system with dynamic admission/access control, in
Proc. IEEE Veh. Technol. Conf.Spring, 2001, pp. 24402444.
[11] D. Gross and C. M. Harris, Fundamentals of Queueing Theory.
Hoboken, NJ: Wiley, 1998.
Pulse Distortion Caused by Cylinder Diffraction and
Its Impact on UWB Communications
Chenming Zhou, Student Member, IEEE, and
Robert C. Qiu, Senior Member, IEEE
AbstractOne of the characteristics of ultrawideband (UWB) signals is
pulse distortion, which is inherently determined by their huge bandwidth.
Using a cylinder model as an example, pulse distortion and its impacts
on UWB system performance have been investigated, based on the closed-
form impulse response rst derived in this paper. Although a lot of papers
have addressed the pulse-distortion issue, quantifying the impacts of pulse
distortion on system performance appears to be novel. The simulation
results show that the SNR loss caused by template mismatch could reach
as high as 4 dB. It is also found that the range error caused by pulse
distortion is much larger than the CramerRao lower bound; thus, it is
another fundamental source of errors limiting the accuracy of times of
arrival of a received signal. These results have direct applications in timing
synchronization and positioning.
Index TermsCylinder diffraction, pulse distortion, ray tracing, timing
and positioning, ultrawideband.
I. INTRODUCTION
Emerging applications of ultrawideband (UWB) are foreseen
for sensor networks that are critical to mobile computing [1],
[2]. Such networks, combining low medium-rate communications
(50 kb/s1 Mb/s) over distances of 100 m with positioning capabilities,
allow a new range of applications [1], including military applications,
medical applications (monitoring of patients), family communications/
supervision of children, search-and-rescue (communications with re
ghters, or avalanche/earthquake victims), control-of-home applica-
tions, logistics (package tracking), and security applications (localiz-
ing authorized persons in high-security areas).
When a sensor is placed in different environments, a nonline-
of-sight (NLOS) propagation is encountered very often, sometimes
in military communications [3]. Sometimes, the propagation path is
Manuscript received August 16, 2005; revised March 20, 2006, August 30,
2006, and September 11, 2006. This work was supported by the Ofce of
Naval Research under Grant N00014-07-1-0529, the National Science Foun-
dation under Grant ECS-0622125, the Army Research Laboratory, and the
Army Research Ofce under STIR Grant W911NF-06-1-0349, and DURIP
Grant W911NF-05-1-0111. The review of this paper was coordinated by
Prof. R. Janaswamy.
The authors are with the Center for Manufacturing Research, Department
of Electrical and Computer Engineering, Tennessee Technological University,
Cookeville, TN 38505 USA (e-mail: czhou21@tntech.edu; rqiu@tntech.edu).
Color versions of one or more of the gures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identier 10.1109/TVT.2007.897640
blocked by objects that can be modeled by a cylinder [4], [10]. For
example, when a hill is smooth and not covered by trees or houses, the
diffraction process is described more accurately in terms of creeping
rays [4]. The purpose of this paper is to model such an environment,
analyzing the possible pulse distortion and its impacts on system
performance.
To be mathematically tractable, a simple channel consisting of a
perfectly electrically conducting (PEC) cylinder is considered. The
transceivers are placed such that only diffracted rays are present at the
receiver. Noting that the research of UWB sensors is still in its early
stage, such a mathematically tractable physics-based channel model,
although simple, may still bring us a lot of insight.
The work in this paper is different from previous works [5],
[7][9] in several aspects. First, it is the rst time for us to use
the cylinder model, which is mathematically difcult to deal with.
Second, the study of pulse distortion from the point of ranging
is also new. Third, the results are different from other canonical
structures. Particularly, we found that the SNR loss and timing error
caused by pulse distortion in the correlation-based receivers could be
signicant and thus deserved special attention. It is thus shown that
physical mechanisms can be naturally connected with the system-level
parameters, such as SNR, timing errors, etc. This unique feature
motivates the research behind this paper.
The rest of this paper is organized as follows. Section II will
analyze the impacts of pulse distortion on the system performance. A
closed-form impulse response for a cylinder channel will be derived
in Section III, based on the well-known frequency-domain results.
Some numerical results on pulse distortion and its impact on system
performance will be shown in Section IV. Finally, Section V will
conclude this paper.
II. PERFORMANCE DEGRADATION DUE TO PULSE DISTORTION
A. Impact of Pulse Distortion on System Performance
We follow the general system model and its performance expression
in the studies in [5] and [14]. When zero and one are sent with
equal probability, the average error probability in the receiver can be
expressed as
P
e
= Q(
SNR) (1)
where Q(x) is dened as Q(x) =
_
x
(1)/(
2) exp(y
2
/2)dy,
and SNR is the signal-to-noise power ratio at the input to the threshold
device. The square root of SNR is given by
SNR =
(s
0
(t) q(t)) [
t=T
0
(s
1
(t) q(t))[
t=T
0
2N
0
|q|
(2)
where s
0
(t) and s
1
(t) are the received signals, which can be singular
but with limited energy, and denotes convolution operation. While
(2) is valid for any binary modulation, the antipodal modulation will
be assumed in this paper.
Let p
0
(t) and p
1
(t) denote the transmitted signals, and h(t) denote
the channel impulse response, then it follows that s
0
(t) = p
0
(t) h(t)
and that s
1
(t) = p
1
(t) h(t). In (2), q(t) is the local template used
in the correlation-based receiver, and |q| = [
_
q
2
(t)dt]
1/2
is the
normof q(t). It is known that, for the optimumreceiver, q(t) is selected
to be matched to the received signal s
i
(t). However, sometimes, q(t)
is matched to the transmitted waveform: p
0
(t) and p
1
(t). This implies
that the signal waveforms will not change as they pass through the
0018-9545/$25.00 2007 IEEE
INFORMATION AND COMMUNICATION TECHNOLOGIES AND SERVICES, VOL. 9, NO. 1, MARCH 2011 7
2011 ADVANCES IN ELECTRICAL AND ELECTRONIC ENGINEERING ISSN 1804-3119
ERLANG C FORMULA AND ITS USE IN THE CALL CENTERS
Erik CHROMY
.1
, Tibor MISUTH
.1
, Matej KAVACKY
.1

1
Department of Telecommunications, Faculty of Electrical Engineering and Information Technology, Slovak University of
Technology, Ilkoviova 3, 812 19 Bratislava, Slovak Republic
chromy@ktl.elf.stuba.sk, misuth@ktl.elf.stuba.sk, kavacky@ktl.elf.stuba.sk

Abstract. This paper deals with calculation of important
parameters of the Call Center using the Erlang C formula
and then results are verified through simulations. Erlang C
formula is defined as a function of two variables: the
number of agents N and the load A. On the base of their
values it is possible to determine the probability P
C
, that
the incoming call will not be served immediately, but it will
have to wait in the waiting queue. Simulations satisfy the
assumption of Markov models.
Keywords
Call Center, Erlang C Formula, Markov Models,
Quality of Service.
1. Introduction
Call Center is dynamical, technical system (package
of technical equipments hardware, software and human
sources) designed for effective connecting people with the
requirements for service with operator or with systems able
to satisfy their requirements. The core of the Call Center is
Automatic Call Distribution (ACD).
Each of the components of the ACD can be described
with some precision by means of mathematical tools and
causalities. Since the ACD systems process a large number
of incoming requests, the majority of models is based on
the principles of mathematical statistics. The right choice
of a statistical model is able to ensure the sufficient
accuracy of the results. It is essential to describe the
dependency of input variables and parameters that can
greatly affect the accuracy of the results. The modeling of
Call Center parameters is possible through Markov models,
but also through Erlang formulas.
This paper deals with calculation of important
parameters of the Call Center (which affect proper
operation of such queuing system) using the Erlang C
formula and then results are verified through simulations.
These simulations satisfy the assumption of Markov
models.
1.1 Erlang C formula and M/M/m/ model
Immediate rejection of call by reason of occupation of
all agents (as expected in Erlang B formula) is in terms of
provided services by Call Center inappropriate solution.
This shortness is eliminated in second Erlang formula
Erlang C. In the case that the call cannot be served
immediately, the call is placed into the waiting queue with
unlimited length. If the release of one of the agents
happens, it is automatically assigned to the following call
from the queue. If the waiting queue is empty, the agent is
free and he waits for next call.
Erlang C formula [5] is in the original form defined as
a function of two variables: the number of agents N and the
load A. On the base of their values it is possible to
determine the probability P
C
(1), that the incoming call will
not be served immediately, but it will have to wait in the
waiting queue.

( )
( )
( )
=

+
=
1
0
! !
!
,
N
i
N i
N
C
A N N
N A
i
A
A N N
N A
A N P
,
(1)
where

= A
.
(2)
Now, we use the relationship between load A (2), the
average number of calls per time and the average number
of requests processed per time . Next, we define the
variable , which represents the load of 1 agent as [2, 3, 7]:

q
N
=
.
(3)
By substituting (2) and (3) into equation (1) we have:
8 INFORMATION AND COMMUNICATION TECHNOLOGIES AND SERVICES, VOL. 9, NO. 1, MARCH 2011

( )
( )
( )
( ) ( )
( )
,
1 ! !
1 !
!
!
!
,
1
0
1
0
=
|
|
.
|
\
|
|
|
.
|
\
|
|
|
.
|
\
|
+
|
|
.
|
\
|
|
|
.
|
\
|
|
|
.
|
\
|
|
|
.
|
\
|
=
N
i
N i
N
N
i
N i
N
C
N
N
i
N
N
N
N N
N
i
N N
N
N P
q
q q
q
q
q
(4)
what corresponding with the relation for the probability
that the request in queuing system M/M/m/ will be placed
into the queue, i.e. in the system is more than m requests
[2, 3, 7]. Now it is analytically derived, that the Erlang C
model (1) and the Markov model M/M/m/ (4) are
identical.
It is possible by using of the basic form for the Erlang
C formula (1) to calculate the value of parameter A
(maximum load) at a known number of agents N and the
probability of waiting P
C
. Due to complexity of the
analytical expression of these unknown parameters,
numerical method for solving is used.
By adding the waiting queue into the system, many
other parameter variables that can be monitored and also
affected will appear. An important factor in terms of caller
is queue waiting time. This value is random variable
described by distribution function [7]:
( )
( )
0 , 1 > =

t t
t A N
C W
e P F
.
(5)
Then it is possible to calculate the average queue
waiting time W (average call waiting time in the queue
before assigning a call to agent):

( ) A N
P
W
C
.
(6)
and by applying of Little theorem [7] and formula (2) we
get the average number of requests in waiting queue as
follows:

( ) ( )
C C
P
A N
A
P
A N
W Q
= =

.
(7)
By using of general definition of distribution function
of any statistical distribution [4] and by applying its
properties on the distribution function (5), we can derive
the formula for calculation of the parameter GoS (Grade of
Service) (percentage of calls, that are answered, or
assigned to the agent before the defined threshold AWT
Acceptable Waiting Time) by known value of AWT [2, 6]:

( )AWT A N
C
e P GoS

=

1
.
(8)
The average number of requests in the system K (and
also the average number of occupied lines) is [7]:

( )
Q A P
A N
A
A P N K
C C
+ =
+ =
+ =
q
q
q
1
,
(9)
where we get from Little theorem a value of the average
time T, that the requirement spend in the system:

( ) A N
P
W
Q A K
T
C
+ = + =
+
= =

1 1
.
(10)
Other important parameter is the average utilization
of agents [2]:

( )
+
(
=
1
0
1
0
! !
!
1
N
i
N i
N
k
k
A N N
N A
i
A
A
k n
k N
q .
(11)
Erlang C formula is the basis for the analysis of
parameters and simulation of the call center. Its shortness is
the assumption of the unlimited waiting queue. This is not
a problem in terms of available storage capacity, and
therefore the waiting queue could be potentially unlimited,
but no real caller will wait too long. Therefore, the
limitation of waiting period represents other parameter that
is under consideration in the special Markov models.
1.2 M/M/m/M model with limited length of
waiting queue
A limited number of requirements placed in the queue
of queuing system can be described by Markov model
M/M/m/M, where the maximum number of requests in the
system M is greater or equal to the number of servers m=N
[2].
The probability p
0
[2], that in the system occurs
exactly 0 requirements (i.e. empty) is defined as:

1
1
1
0
0
1 !
1
!
|
.
|
\
|

(
(
|
.
|
\
|
+ =

N
A
N
N
A
A
k
A
p
N M
N
N
k
k
(12)
and then we can define the probability of the call rejection
P
B
[2] as:

0
!
p
N N
A
P
N M
M
B

=
(13)
and the probability, that the call will be assigned into the
waiting queue P
C
[2] as:

0
!
p
N
A
P
N
C
= .
(14)
Furthermore, for the values Q [2] and K [2] can be
calculated as

( )
|
.
|
\
|
|
.
|
\
|
+
|
.
|
\
|
+ +
N
A
N
A
N M
N
A
A N N
A p
Q
N M
N M N
1 1
1
) ( )! 1 (
1
2
1
0
,
(15)

|
|
.
|
\
|
+ =

0
!
1 p
N N
A
A Q K
N M
M
.
(16)
The time characteristics can be calculated on the basis
of Little theorem [7].
These relations are relatively complicated. It is
therefore possible a consideration, if the approximation
using standard Erlang C formula is not sufficient, or what
are the conditions, that such approximation is sufficiently
exact. If the average number of requests in the system
K will be far less than the limit M, it is possible to apply
Erlang C formula without thinking of the maximum
capacity. The more closely will be the value K to the limit,
the less accurate results Erlang C formula will provide.
2. The principle of realized simulations
The basic of the simulated algorithm consists of three
blocks:
- set up of inputs,
- traffic simulations,
- processing of measured values and their presentation.
2.1 Modeling of inputs
The basic parameters are:
- average number of incoming calls into the call center
per time ,
- average time of call processing by agent 1/,
- number of agents N.
The group of time parameters consists of:
- the total length of simulated traffic in seconds T
SIM
,
- time step of simulation T
STEP
(by default 1 second),
- time to steady state T
NAB
.
Vector with arrival times of each call based on the
average number of requests per time unit and length of
simulation is created by random generator. There is used
a property about exponential distribution of length of the
intervals between arrivals. Generated random variables are
then tested by the statistical chi-quadrate [4]. There is test
of vector consistency with exponential distribution on
significance level = 0,05. In the case of unsuccessful test,
the entire vector is randomly generated once again.
Number of generated calls is by 20 % higher than the
average number of calls that should enter into the call
center through the simulated period.
Furthermore, vector with service times of individual
calls that will be allocated during the simulation, is
generated for each agent based on his average service time.
All these operations are performed in advance by
reason of high rate and efficiency of MATLAB by working
with vectors and matrices. The generation of set of values
is then faster than a gradual generation of individual values
during the simulation.
2.2 Traffic simulation
The core of simulation is realized as a cycle, in which
the each iteration represents one time step in simulation. In
the each iteration is the vector of incoming calls compared
with actual time and if it is necessary, the call enters into
the simulated call center in proper timer. Concurrently the
time of incoming call is recorded for later calculations. Call
is placed into the waiting queue or directly assigned to a
free agent. The time index is stored for later calculations
(average waiting time) at the moment of call assignment.
Furthermore, in the each iteration the status and the
occupancy of all agents are checked. If any of the agents is
free, it will be assigned the first call from the waiting
queue. If more agents are free, the call is assigned to agent
that didnt work for the longest time. If there is not a call in
the waiting queue, the agent remains as free and expects
the arrival of another request into the system.
During the one iteration there is possibility of entry
into the system and also assignment to the agent of more
than one call simultaneously. The number of calls in the
system is stored in each step so then it is possible to
determine the number of calls in the waiting queue.
The mathematical model of queuing system is defined
for the steady state. It means, that in the moment of
parameters monitoring, the simulation runs infinitely long
time.
The run of the simulation is terminated after
predefined simulation time. Only the calls that are
terminated (and are thus served by agent) before expiration
of the simulated time are included into the statistics.
2.3 Processing of the simulation results
This phase of the algorithm ensures the processing of
all measured data during the simulation (time of events
generation, number of calls in the system, agent
occupancy, ). Based on these the particular parameters
monitored in the call center are calculated. Thus obtained
simulation results can be then compared with the expected
values obtained by calculations through the mathematical
model.
The general variables available from the results of the
simulated model are:
- real length of the simulation (calculation time),
- number of simulated calls,
- average service time per one call (1/),
- average number of calls in the system (K),
- average time spent by user in the system (T),
- number calls placed in the waiting queue,
- probability of blocking P
B
, respectively probability of
waiting P
C
,
- average utilization of agents .
3. Calculations using the Erlang C
model
Erlang C model works in the basic form with 3 input
parameters (A, N, P
C
). By use of relation (2) it is possible
to divide the value of load A (generated by incoming calls)
into 2 components: and 1/. Calculator thus always works
with 4 values, while 3 of them act as input parameters and
the last parameter is the output parameter. By adding of the
waiting queue we can obtain a set of new parameters that
can be calculated and then compared with the simulation
results:
- average number of requests in the waiting queue Q
(using (7)),
- average number of requests in the call center K (using
(9)),
- average waiting time in the waiting queue W (using
(6)),
- average time spent in the call center T (using (10)),
- value of GoS for AWT=20 s (using (8)),
- average utilization of agents (using (11)).
Probability of insertion into waiting queue P
C
can be
entered in three different variants:
- direct entry of value P
C
,
- entry of average waiting time W,
- entry of GoS.
When using a P
C
as input variable, this variable is
considered to the upper limit.
3.1 Calculation of the parameter Pc
The calculation of the unknown value of probability
of waiting can be easily realized by basic relation for
Erlang C model (1). The dividing of two large number
could lead to numerical errors and the obtained result
might not be accurate. Therefore we derived an alternative
relation, that is identical to the original (1) (in terms of
result):

( )
|
|
.
|
\
|

+
=
[
=
=
N
i
i
k
C
A
k N
N
A N
A N P
1
1
0
1
1
,
.
(17)
Similarly, we can use Horner scheme [8] also for
more efficient calculations.
3.2 Calculation of the parameter N
Analytical calculation according to (1) respectively
(17) would be very difficult. The solution is therefore
through a numerical method. The easiest technique is to
gradually increase of the number of agents and continuous
checking the stop conditions. This depends on the form, in
which the input value P
C
is inserted (thus direct, or as W,
or as GoS). At the moment, where the current value in the
calculation is less or equal to the required value, the
necessary number of agents is found. The implementation
uses this idea, but applying of the relation (17) in every
step of the calculation it is possible to use the current result
obtained from the previous iteration cycle for the lower
value of N. So, this is very quickly method to find the
unknown value N.
3.3 Calculation of the parameter or 1/
The calculation of one of these unknown quantities in
terms of Erlang C model means the find out of the load A,
that can the system process at the specified parameters N
and P
C
(respectively W and GoS parameters).
Consequently, we can calculate the second one by applying
the formula (2) and one of the value and 1/.
In terms of implementation, the finding of the unknown
value A is the most difficult of all three combinations. As
the solution we can use the feature, that allows to search
the value of unknown x, for which it holds f(x) = 0. In this
case, the function f(x) for input P
C
is:
0 ) , ( ) (
_
= =
INPUT C C
P N x P x f
.
(18)
If the input value is defined as the average waiting
time W, respectively GoS, we can use the following
substitutions by (6) and (10):
( ) A N W P
C
=
,
(19)

( ) AWT A N
C
e
GoS
P

=

1
.
(20)
In both cases we have the value . Therefore, if we
need the solution for the unknown value 1/ , we must find
the solution for the value and only then to calculate the
load A by (2). The functions f(x) for input W (by use of
substitution (19)) are:
( ) 0 ) , ( ) ( = = x N W N x P x f
C

,
(21)
respectively, if is unknown, then:

0
3600
,
3600
) ( = |
.
|
\
|
|
.
|
\
|
=
x
N
x
W
N
x
P x f
C

.
(22)
If the input parameter is GoS, the substitution (20) is
used and the function for parameter is:
( )
( )
0
1
, ) ( =
=
AWT x N
C
e
GoS
N x P x f

,
(23)
respectively, if is unknown, then:

0
1
,
3600
) (
3600
=
|
.
|
\
|
=
|
.
|
\
|
AWT
x
N x
C
e
GoS
N
x
P x f

.
(24)
4. Simulations
The whole process of simulation for Erlang C model
is shown in the following fig. 1:

Fig. 1. Process of simulation for Erlang C model
4.1 Simulation results
The parameters of simulated model are:
- 667 incoming calls per 1 hour,
- average call service time is 150 seconds,
- simulation time of the call center work is 30 hours
with 1 second step (run to steady state is 1 hour).
The calls that cannot be processed immediately are
placed into the waiting queue (according to Erlang C
formula). For the system stability it is necessary to satisfy
the condition A < N. Therefore the calculations and
simulations are realized for the number of agents in the
range of 28 to 37. In this range are the most notable
changes observed in output parameters. The table 1 and
table 2 shows the results obtained by calculation using the
mathematical model and results also obtained by
simulations.
Tab. 1. The results of calculations
N
Pc
[%]
K T [s] Q
W
[s]
GoS
[%]

[%]
28 95,4 155 836,6 127,2 686,6 7,2 99,3
29 75,3 45,1 243,5 17,3 93,5 35,9 95,8
30 58,7 35,2 189,9 7,4 39,9 56,3 92,6
31 45,1 31,7 171,1 3,9 21,1 70,6 89,7
32 34,1 30 162,1 2,2 12,1 80,6 86,8
33 25,3 29,1 157,3 1,4 7,3 87,3 84,2
34 18,5 28,6 154,5 0,8 4,5 91,9 81,7
35 13,3 28,3 152,8 0,5 2,8 94,9 79,4
36 9,4 28,1 151,7 0,3 1,7 96,8 77,2
37 6,5 28 151,1 0,2 1,1 98,1 75,1
Tab. 2. The results of Erlang C model simulations
N
Pc
[%]
K T [s] Q
W
[s]
GoS
[%]

[%]
28 91,9 88,5 478 60,9 328 12,3 98,8
29 74 43,2 233 15,4 83,4 38 95,6
30 59,3 35,1 189 7,2 38,8 55,9 93
31 46,2 32,4 174 4,5 24,1 69,3 90
32 33,7 30 162 2,2 11,9 81,3 86,9
33 25,1 29,2 158 1,4 7,3 87,5 84,4
34 19,2 28,8 155 0,9 4,7 91,5 82,1
35 13 28,3 153 0,5 2,5 95,5 79,4
36 8,7 28 151 0,3 1,5 97,4 77,1
37 6,4 28,1 151 0,2 1 98,2 75,3
Obtained simulation results and calculations are very
similar. The differences exist only in the case of the
minimum number of agents. However, it is probably that
any company will not carry on the call center with
extremely poor quality of service [9, 10, 11, 12] delivering
(mainly the very long waiting time).
From the caller point of view, the most important
parameters are the average waiting time in the waiting
queue W and parameter GoS (this case is evaluated for
AWT=20 seconds). The caller expects the lowest value of
W and also the value GoS, which is close to 100%. From
the call center point of view, the most important parameters
are the number of agents N and their utilization , because
these two variables significantly affect the financial
demands of service. The aim of the operator is to minimize
the number of agents and to maximize their utilization. The
aim of the analysis is therefore to find such a minimum
number of agents N, when the operation parameters are yet
on the sufficient level. As suitable we can consider the GoS
parameter on level 80% and the average waiting time about
10seconds.

Fig. 2. Simulation results
The fig. 2 shows the characteristic curve of the
important call center simulation results according to Erlang
C model assumptions in relation with the number of agents
N. The characteristic curve of the average waiting time has
a very strong exponential character and thus a slight
increase of the number of agent (about 1 to 2 agents) can
bring a significant improvement of this parameter.
A similar, even though less aggressive, is the characteristic
curve of the probability P
C
. GoS parameter also
exponentially converges to the level 100% and we can see
that a small change in the number of agents can bring
significant improvement. The characteristic curve of agents
load is in the displayed range almost linear.
According to the above mentioned requirements on
the provided quality of service by the call center is in this
case possible to consider 32 agents as sufficient. By adding
two agents it is possible to shorten the average waiting time
by half and to increase the value of GoS parameter at 10%
on very decent level (90%). The utilization of agents does
not decrease below 80% and therefore it does not create
unnecessarily long pauses, when the agents were
redundant.
5. Conclusion
Based on calculations and simulations it can be stated,
that in term of simplicity and accuracy of obtained results
Erlang C formula is applicable for call center simulations.
However, its shortness is the possibility of calculations
only for one service group, and also the need to define for
all agents the same service time.
Despite these limitations, it is possible to use the basic
Erlang C formula also for calculations for call center with
several service groups. In this case when it is possible to
determine the probability that calls are routed to the
individual service groups, then it is possible to use the
basic Erlang C formula for calculations for each service
group individually. The number of incoming calls per time
unit is the aliquot portion of the total number of incoming
calls into the call center. The Erlang C formula calculation
can be used in the case of different performance of agents.
It is possible to determine the value of the average call
processing time or average number of the processed calls
per time unit by dividing the total number of processed
calls per time unit of all agents and the total number of
agents in service group.
For the purpose of further study it would be
interesting to expand the simulations by more independent
service groups at the same time and the random
distribution of call between them according to defined
probabilities. Another interesting possibility could be
different average service time for individual agents. Finally
there is possibility to simulate the impact of unequal
performance of agents on the results of the whole call
center.
Acknowledgments
This work is a part of research activities conducted at
Slovak University of Technology Bratislava, Faculty
of Electrical Engineering and Information Technology,
Department of Telecommunications, within the scope of
the projects VEGA No. 1/0565/09 Modeling of traffic
parameters in NGN telecommunication networks and
services and ITMS 26240120029 Support for Building
of Centre of Excellence for SMART technologies, systems
and services II.
References
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Indianapolis (IN) (USA): Wiley Publishing, 2005, 80 p. ISBN 0-
471-75819-1. Available from:
<http://www.algotech.sk/files/ContactCentersforDummies.pdf>.
[2] UNOVSK, L. Stochastick modely operanej analzy. 1
st
ed.
Bratislava: ALFA, 1980, 416 p. ISBN 63-557-80.
[3] POLEC, J., KARLUBKOV, T. Stochastick modely v
telekomunikcich. 1
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ed. Bratislava: Fond Jozefa Murgaa pre
telekomunikcie n.f. vo vydavatestve FABER, 1999, 128 p. ISBN
80-968125-0-5.
[4] RIEANOV, Z. et al. Numerical methods and mathematical
statistics. 1
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559-87.
[5] Diagnostic Strategies. Traffic Modeling and Resource Allocation in
Call Centers. Needham, USA: Diagnostic Strategies, 2003.
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<http://www.fer.hr/_download/repository/A4_1Traffic_Modeling.pd
f>.
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Universiteit, 68 p. [cited 2008-01-26]. Available from:
<http://www.math.vu.nl/~koole/ccmath>.
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nd
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[9] POSOLDOV, A., BAROK, I.: AC Methods in ATM and IP
Networks. In New Information and Multimedia Technologies. NIMT
2010: Brno, Czech Republic, 16. - 17. 9. 2010, Brno University of
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in IP. In: New Information and Multimedia Technologies. NIMT
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About Authors ...
Erik CHROM was born in Vek Krt,
Slovakia, in 1981. He received the Master
degree in telecommunications in 2005 from
Faculty of Electrical Engineering and
Information Technology of Slovak
University of Technology (FEI STU)
Bratislava. In 2007 he submitted PhD work
from the field of Observation of statistical properties of
input flow of traffic sources on virtual paths dimensioning
and his scientific research is focused on optimizing of
processes in convergent networks. Nowadays he works as
assistant professor at the Department of
Telecommunications of FEI STU Bratislava.
Tibor MIUTH is a student of PhD.
study at Department of
Telecommunications, Faculty of
Electrical Engineering and Information
Technology of Slovak University of
Technology Bratislava. He focuses on
application of Erlangs' equations both in
classic telecommunication networks and modern IP
networks.
Matej KAVACK was born in Nitra,
Slovakia, in 1979. He received the Master
degree in telecommunications in 2004
from Faculty of Electrical Engineering
and Information Technology of Slovak
University of Technology (FEI STU)
Bratislava. In 2006 he submitted PhD
work Quality of Service in Broadband Networks.
Nowadays he works as assistant professor at the
Department of Telecommunications of FEI STU Bratislava
and his scientific research is focused on the field of quality
of service and private telecommunication networks.

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ESCUELA SUPERIOR POLITCNICA DE CHIMBORAZO

b transition rate from state (i, j + 1) to state (i, j)

d transition rate from state (i, j 1) to state (i, j)

i transition rate from state (i, j) to state (i + 1, j)

j transition rate from state (i, j) to state (i, j + 1)

k transition rate from state (i, j) to state (i 1, j)

l transition rate from state (i, j) to state (i, j 1)

) should fall into the acceptance

) is (16, 4), one voice call and three data calls

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