Performance Analysis of an Integrated Voice and Data CDMA

System

Takeshi Sato and Abbas Sandouk

Department of Information Electronics, School of Engineering, Nagoya University, Nagoya, Japan 464-8603

Takaya Yamazato

The Center for Information Media Studies, Nagoya University, Nagoya, Japan 464-8603

Masaaki Katayama and Akira Ogawa

Department of Information Electronics, School of Engineering, Nagoya University, Nagoya, Japan 464-8603

SUMMARY

This research is an analysis of the mixed system of

data packets and CDMA voice signals transmitted by the

channel load sensing protocol. The voice signal is a medium

requiring instantaneous action to complete the transmission

within a certain time whereas the data packet is a medium

requiring high reliability but allows temporal delay. The

blocking probability and Erlang capacity are derived for the

voice signal while the throughput is derived for the data

packet. The characteristics in the system using CDMA

strongly depend on the number of simultaneously transmit-

ting stations. In particular, in the present system, due to the

mixture of different media, the effect of traffic on each

medium needs to be considered. In regard to this point, this

paper analyzes the effect of traffic on each medium from

the relationship of the required quality of the media and the

number of simultaneously transmitting stations. Then the

optimization of the system with two kinds of media is

considered. Further, the system capacity is derived as a

guideline for the system optimization. The data offered load

is derived to make maximally efficient use of the channel.

© 2000 Scripta Technica, Electron Comm Jpn Pt 2, 83(4):

61�70, 2000

Key words: Channel load sensing protocol; voice

signal and data packet; system capacity.

1. Introduction

In the area of wireless communication, extensive

research has been carried out recently on the wireless

multimedia communication systems that provide wireless

service with various media such as voice, video, and data.

In the wireless multimedia communication system, the

transmission speeds of the media are different and so are

the required communication quality levels. Therefore, what

is required is establishing a system with a flexibility to

handle these media at the same time. The code division

multiple access (CDMA) using a spread spectrum has

drawn attention as a system suitable for such requirement

in multimedia communication [1�3].

© 2000 Scripta Technica

Electronics and Communications in Japan, Part 2, Vol. 83, No. 4, 2000Translated from Denshi Joho Tsushin Gakkai Ronbunshi, Vol. J81-B-II, No. 8, August 1998, pp. 733�741

61

In this paper, the characteristics are analyzed for the

asynchronous CDMA system in which the voice signals

and the data packet are mixed.

The voice signal is a medium that needs instantane-

ous action to complete transmission within a short time. On

the other hand, the data packet requires high reliability

although time delay can be tolerated [1]. In order to satisfy

the instantaneous nature, it is assumed that the voice users

make channel reservation at the time of transmission. When

a transmission request occurs, the user sends out the reser-

vation packet to the central station according to the instruc-

tion from them. If the number of secured channels reaches

a threshold at the central station, other voice users are

denied reservation, that is, they are blocked. The data packet

user sends out the packet according to the control signal

from the central station without making reservation.

In this analysis, the packet transmission uses CDMA

Unslotted ALOHA system [5, 6] while the transmission

control uses Channel Load Sensing Protocol (CLSP) [5�7].

In CLSP, the monitoring of the channel usage is carried out

at the base station. As the channel use condition changes,

the base station broadcasts the control signal permitting

packet transmission to all users at an arbitrary time if the

number of packets being transmitted is smaller than the

given threshold D, and the control signal forbidding packet

transmission if the number of packets exceeds D. Each user

transmits the packet according to the decision being re-

ceived. The CDMA Unslotted ALOHA system experiences

quality degradation by the increase of other interference

when the transmission requirement (offered load) is high.

However, since CLSP can prevent increase of the number

of other interference, a certain throughput can be main-

tained even if the offered load is high. Further, since the

number of simultaneously transmitting stations can be kept

below a certain preset threshold, the method can cope with

the required quality of each medium and with the priority

requirements [4].

The characteristics in the system using CDMA

strongly depend on the number of simultaneously transmit-

ting stations. In particular, in the present system, calls of

different media are mixed, so that the effect of each traffic

needs to be considered. This point is noted in the present

work. The effects of the traffic on each medium are ana-

lyzed from the relationship between the required quality of

the media and the number of simultaneously transmitting

stations so that the optimization of the system with two

kinds of media mixed is studied.

In this paper, the blocking probability for the voice

signal, Erlang capacity, and the throughput for the data

packet are derived. The evaluation of the characteristics

proceeds with a note on how much the throughput of the

data packet can be improved while the required quality of

the voice signal is maintained.

Basically, if either traffic is increased, the charac-

teristic of the other is expected to be degraded. Therefore,

it is important in the system evaluation to take into account

the trade-off between the voice traffic and the data packet

traffic. Hence, the system capacity is defined as the sum of

the Erlang capacity and the data throughput. From this

definition, the existence of the data offered load is derived

in order to make maximally efficient use of the channel.

Below, the system model is presented in Section 2,

the characteristic analysis in the case where the voice signal

and the data packet are mixed is given in Section 3, and the

analysis results are discussed in Section 4. Section 5 con-

cludes the paper.

2. System Model

Figure 1 shows the system model. In the present

analysis, only the uplink is considered. Several voice users

transmit spread spectrum signals to the central station at one

hop. The use with a transmission request sends out the

reservation packet (RP) if the control signal (CS) by CLSP

from the central station permits transmission (see Fig. 2).

This reservation is also assumed to have spread spectrum.

If the transmission is forbidden, the transmission permis-

sion is delayed. If the reservation packet is succeeded in

transmission, the user enters the transmission mode and

sends out the voice signal. The user directly spreads the

voice signal by means of the spreading code assigned

specifically to him. Then, the voice signal is transmitted to

the central station. In such an instance, it is not necessary

for the voice user to take into account the control signal by

CLSP while the instantaneous nature of the signal is se-

cured.

Fig. 1. System model.

62

The data packet user directly spreads the signal by

means of the spreading signal specifically assigned. Based

on the control signal, the signal is sent to the central station

in one hop.

With a view to carrying out the analysis with an

emphasis on two factors important in the CDMA system,

namely, the required qualities for the two types of calls and

the number of simultaneously transmitting stations, the

following assumptions are used.

x The voice conversation time length of each user

has an exponential length while the data packet

has a fixed length.

x The reservation packet and the data packet are

Poisson-generated at mean arrival rates of Ov (1/s)

and Od, respectively.

x Both the voice and data packet have perfect power

control. Each signal is received at an equal power.

x The spreading signal has a random signature.

x The spreading signal synchronization is perfect.

x The bit rates and the spreading rates of voice and

data packet are equal.

x There is no propagation delay in CLSP nor any

delay caused by other signal processing.

x No error correcting code is used. Hence, if there

is an error in even one bit, this packet cannot be

transmitted correctly.

x The bit error rate BER�k� for the random signature

in the asynchronous CDMA is given by the one

obtained by the standard Gaussian approximation

[8]:

where k is the number of simultaneously transmitting sta-

tions and N is the spreading rate derived from N W/ R.

Here, W is the operating frequency bandwidth and R is the

bit rate. Also, Q�x� 1/2 � er fc�x /�CC2 �.

3. Analysis for Voice and Data System

3.1. Voice-only system

3.1.1. Blocking probability

In the voice-only system, the reservation can defi-

nitely be obtained by one transmission. The interval from

the transmission of the reservation packet until the comple-

tion of the transmission of voice signals is assumed to

follow the exponential distribution of the service comple-

tion rate of Pv. Further, since the packets are sent out

according to the control signal from the central station, the

transition of the number of the voice signal existing in the

central station follows the queuing matrix system of

M/M/ D /D in CLSP [9]. Here, M/M/ D /D indicates the

(signal generation interval : exponential distribution)/(sig-

nal length : exponential distribution)/(number of output

lines)/(number of buffers at the central station). Then, the

blocking probability is given by the Erlang B formula as the

one in which the sending out of the reservation packet is

denied by the control signal [9]:

where m is the number of voice users and D indicates the

threshold of CLSP. Also, the value of D is derived from the

bit error rate the voice signal can tolerate. Hence, the

maximum number of simultaneously transmitting stations

k is used as the threshold:

BER�k� is the bit error rate (1) and the maximum allowable

error rate of the voice signal is 10�3.

3.1.2. Average number of voice signals

Since some transmissions may be denied in CLSP,

the number of transmission requests and the actual number

of signals transmitted are different. The offered load

Gv� Ov /Pv� of the voice user is defined as the average

number of voice users engaged in conversation during

average conversation time. For Gv, the average number of

voice signals Ev actually sent out is derived. Since this value

Fig. 2. Process of getting reservation for voice users.

(1)

(2)

63

corresponds to the average number of voice signals of the

queuing matrix system M/M/ D /D, this is given by

where m is the number of simultaneously transmitting

stations while U indicates the voice activation (VA) rate. VA

is the method in which no signal is transmitted in the

no-sound duration when the receiver is listening to the

transmitter or in the pause in conversation. Here, it is also

assumed that the central station is carrying out control

based on the number of voice users during conversations

even in CLSP. Figure 3 shows the mean number of estab-

lished voice users versus the offered load Gv of the voice

users. The used parameters as shown in the numerical

example in Section 4 are CLSP threshold D = 98, used

frequency bandwidth W = 20 MHz, bit rate of the voice

signal R = 32 kbps, and VA rate U = 0.4. Also, Eb /N0 o fso that only the multiple connection interference is taken

into account. It is found that the mean number of established

voice users decreases in comparison with the case without

VA. Hence, if VA is used, it is expected to accept more users

while the signal quality is maintained.

3.2. An integrated system

Next, the data packets that send out the CDMA data

packets to the voice system using CLSP are also subject to

control by CLSP. Hence, the base station is considered to

control the transmission of the signals by the number of data

packets and the number of established voice users.

3.2.1. Data throughput analysis

Let us derive the throughput S of the data. The

throughput S is the number of packets succeeding in trans-

mission within one packet time length. Also, the average

number of generated packets in one packet time length is

the offered load Gd. The number of simultaneously trans-

mitting stations of the data and the voice signals in CLSP

is changing in time.

In the case of the analysis with data only, the through-

put and the offered load of the data were related in one-to-

one correspondence [6]. Therefore, it was possible to use

the analysis method in which the transition of the number

of simultaneously transmitting stations can faithfully be

expressed as a Markov model. In the present system, how-

ever, the number of simultaneously transmitting stations of

the voice signal also affects the throughput. If the voice

signals with different characteristics are taken into account,

the transition of the simultaneously transmitting stations

becomes too complicated to be displayed by a simple

Markov model. On the other hand, the number of voice

users does not change within the packet being considered

(see Appendix 1). Hence, it is possible to use a technique

for fixing the number of engaged users kv of the voice

signal. By means of this technique, only the transition of

the number of simultaneously transmitting stations of the

data needs to be considered. Hence, the analysis can be

carried out in a manner similar to the case with data only.

By extending Eq. (1), the bit error rate considering the

number of voice users and the data packets is given by

where kd is the number of simultaneously transmitting

stations of the data packets, kv is the number of voice users

engaged, Nd is the diffusion rate of the data packet, and Nv

is that of the voice signal.

Note that the generation of the voice signals and that

of the data are not mutually independent. This is because

their generations are controlled by CLSP using the sum of

the numbers of simultaneously transmitting stations. The

fixed value kv depends on the offered load Gv of the voice

signal, and its probability density function can be given by

the steady probability of M / M /D /D as follows:

(3)

Fig. 3. Average number of voice transmission.

(4)

(5)

64

where m indicates the number of simultaneously transmit-

ting stations.

When the number of engaged voice users is fixed to

kv, the transition of the simultaneously transmitting stations

for the data follows M/ D / �D � kv� / �D � kv� in the queuing

matrix system (Fig. 4). Here, D indicates the (signal length :

fixed length).

Hence, by means of this steady probability, the num-

ber of simultaneously transmitting stations kd of the data

can be given as follows in terms of the conditional prob-

ability with kv:

Note that the number of transmission requests and the

number of signals actually arriving at the central station are

different in CLSP. This is because the transmission may be

denied. Therefore, the average number Ed of data packets

actually arriving at the central station is derived. The value

of Eq. (5) taking into account the VA rate is multiplied by

Eq. (6) and the result is averaged for all possible values of

kv and kd:

where m is the number of simultaneously transmitting

stations.

Figure 5 shows the state transition diagram on the

number of interference data packets within the noted data

packet when the number of engaged voice users is kv. Until

the number of interference data reaches D � kv, conven-

tional Poisson generation takes place and hence the genera-

tion rate of O and the completion rate P�kd1� are used [6].

Here, kd1 indicates the number of simultaneously transmit-

ted data packets at the first bit of the noted data packet. The

probability for the state larger than D � kv is 0. Hence, the

analysis is proceeded by considering that the number of

channels (number of servers) given for the data packets is

D � kv. The probability P�k, i, k1� is defined. This is the

probability such that when the number of interference data

for the first bit is k1, then the bits up to i � 1 are successful

and further the number of the interference data for the i-th

bit is k.

(i) For i = 1

For kd1 d D � kv � 1 and hence for the case where the

number of simultaneously transmitted data kd1 for the first

bit of the noted data packet is less than D � kv � 1, the results

are obtained from the steady probabili ty

M/D/ �D � kv��D � kv�:

Note that kd1 t D � kv � 1. Since the number of interference

data at the first bit of the noted packet is not larger than

D � kv � 1,

(ii) For i ! 1

The results are obtained from the state transition

diagram Fig. 5. For kd � D � kv � 1, the transition has the

generating rate of Od't and the completion rate of

Pd�kd1�'t since the number of interference signals is smaller

than D � kv � 1. Here, 't is one bit duration of the data

packet. For all cases from i � 1 to i bits, the bit success rate

and the transition rate are multiplied to obtainFig. 4. The arrival of data packets.

(6)

(7)

Fig. 5. State transition diagram.

(8)

(9)

65

When kd D � kv � 1, the number of the noted packet itself

is considered. Then, the generation is stopped as the number

of simultaneously transmitting stations is the threshold of

D � kv. Hence, in the first term of Eq. (10), let Od't 0.

Since the probability of the second term does not exist, we

can obtain

For kd ! D � kv � 1, the total number of simultaneously

transmitting stations never exceeds the threshold. There-

fore,

By using P�kd, i, kd1�, the packet success probability Q is

derived. Let i L in the probability P�kd, i, kd1� and multi-

ply it with the probability for the success of the L-th bit.

Further, by taking the average over all possible values of

the number of interference packets kd and kd1, Q becomes

Further, by taking the average over all possible numbers of

engaged voice users kv, the throughput S is derived as

3.2.2. Voice user blocking probability

The blocking probability derived in Section 2 is

extended to the case in which the number of simultaneously

transmitting stations for data is taken into account. As in

Section 3, the number of simultaneously transmitting sta-

tions for the data is fixed in the analysis (see Appendix 2).

The number of simultaneously transmitting stations kd for

the data can be obtained from the steady probability

PD�kd� of M/ D/D /D. Then the transition of the number of

simultaneously transmitting stations for the voice signals is

given by M/ M/ �D � kd� / �D � kd�. Hence, the blocking

probability of the voice signal when the number of simul-

taneously transmitting stations of the data is kd is given as

follows in terms of the conditional probability by means of

Erlang B equation (2):

When the steady probability for the number of simultane-

ously transmitting stations kd for the data is taken into

account, the blocking probability to be derived is as follows:

The Erlang capacity is defined to be the maximum allow-

able offered load that can maintain the allowable blocking

probability and is given by

where Gv is the maximum value sat isfying

Pblocking�Gv� d 0.01 while 0.01 indicates the maximum al-

lowable blocking probability for the voice signal.

Figure 7 shows the blocking probability for the voice

user. Also, when the offered load of the data is taken into

account, the average number of engaged voice users is

(10)

(11)

(12)

(13)

(14)

(15)

(16)

(17)

(18)

66

where m is the number of simultaneously transmitting

stations.

4. Numerical Examples

By means of the parameters shown in Table 1, the

characteristics are presented. The diffusion rate N for the

random signature is derived from N | W/R by using the

operating frequency bandwidth W = 20 MHz and the bit rate

R = 32 kbps.

Figure 6 shows the data throughput for the offered

load Gd of the data. Here, the offered load Gv of the voice

user is used as a parameter. As Gd is increased, the through-

put approaches the maximum. With a further increase, the

offered load of the entire system increases, so that the effect

of interference becomes stronger, making the throughput

decrease. Further, the characteristic degradation by the

increase of Gv is seen. Also, it is confirmed that the through-

put is improved by VA.

Figure 7 shows the blocking probability for a voice

user with the offered load Gd of the data as a parameter.

With Gd, the offered load of the entire system increases so

that the blocking probability also increases. Hence, the

voice user has more difficulty in making reservation.

Figure 8 shows the effect of the offered load of the

data packet for the Erlang capacity of the voice signal. This

is defined as the maximum allowable offered load of the

voice user when the maximum blocking probability of the

voice user is 1%. The characteristics with different bit rates

are shown for the data packet. As the offered load of the

data packet is increased, the occupancy rate of the data

packet on the channel increases so that the Erlang capacity

is decreased. As the bit rate of the data packet is reduced,

or as the diffusion rate of the data packet is increased, the

Erlang capacity is found to increase. This phenomenon can

be explained as follows. If the energy per bit is constant, the

signal power density becomes smaller as the diffusion rate

of the data packet is increased. Therefore, the offered load

becomes smaller for a larger diffusion rate when compared

at the same number of generation of the data. Then, the

amount of interference of the data on the voice signal

becomes smaller. Hence, the Erlang capacity of the voice

user increases as the bit rate of the data packet is reduced.

It is expected that the optimization of this system is

strongly dependent on the trade-off between the data

Table 1. Parameters

Parameter Value

Average conversation time of voice

user

Tr 60 s

Voice activation rate U = 0.4

Data packet length Ld = 500 bits

Hit rate (Voice, Data) Rv, Rd 32 kbps

Operating frequency bandwidth W = 20 MHz

Diffusion rate (Voice, Data) Nv, Nd = 312

CLSP threshold D = 98

Maximum allowable rate of voice

signal

10�3

Fig. 6. Data throughput.

Fig. 7. Blocking probability.

67

throughput and the Erlang capacity. Hence, the capacity of

the entire system is defined as follows:

Figure 9 shows the system capacity for the offered

load Gd of the data packet. The data rates for the voice signal

and the data packet are assumed identical. This can be

defined as the maximum channel capacity of this system of

the system. This quantity is a guideline as to how much

increase in the offered load of the data packet is needed in

sending out the data packet into the CDMA voice system

so that the system capacity is effectively used. The system

capacity increases if VA is taken into account. Although the

Erlang capacity of the voice user does not increase by VA,

the throughput is increased. When VA is considered, the

capacity of the system is considered to be used the most

efficiently in the range of Gd 0 to Gd 30. Also, the

maximally efficient use can still be maintained for up to

Gd 50 because the data throughput increases while the

Erlang capacity decreases. Subsequently, the data through-

put peaks and decreases. The characteristics of both become

equal slightly past Gd 80. At this point, generation by the

voice user is halted. Hence, the characteristic is based only

on the throughput of the data packet.

5. Conclusions

In this paper, a throughput analysis and an Erlang

capacity analysis were carried out for the mixed system of

the data packets and CDMA voice signals transmitted under

the channel load sensing protocol. The effect of each on the

other was studied. It was found that the number of simulta-

neously transmitting stations becomes an important factor

in the characteristic analysis for the system using CDMA.

The relationship of the required qualities (bit error rates) of

both the voice and data traffics with the number of simul-

taneously transmitting stations is recognized so that the

analysis for this system is carried out by means of the

queuing matrix system. Further, an optimization of this

system was discussed.

As the evaluation criterion of the entire system, the

system capacity is defined as a reference indicating the

trade-off of the evaluation standard of both the voice user

and the data packet user. This quantity indicates one of the

guidelines as to how the offered load of the data packet

should be increased in sending out the data packets into the

CDMA voice system so as to make the system capacity

efficiently. As a result, the offered load of the data that uses

the channel capacity efficiently is derived. The generation

of the data packets is further controlled by means of traffic

control so that the efficient use of the system capacity is

possible while the system optimization, or the throughput

of the data packet and the Erlang capacity of the voice user,

is maintained to a certain degree.

REFERENCES

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Fig. 8. Erlang capacity.

Fig. 9. System capacity.

68

2. Ganesh R, Joseph K, Wilson ND, Raychaudhuri D.

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grated voice/data network. Int J Wireless Information

Networks 1994;1.

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lengths. IEEE J Sel Areas Commun 1990;8:529�541.

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A. Throughput analysis of DS/SSMA unslotted aloha

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APPENDIX

1. Reason Why the Number of Simultaneously

Transmitting Stations of the Voice Signal

Can Be Fixed in Derivation of the Data

Throughput

The voice conversation time is longer than the data

packet length. In this analysis, a length of about 5000 times

is assumed. Further, for both the data and the voice, the

offered load is defined as the product of the number of

signals generated in an average signal time (the number of

reservation packets for the voice) and the average packet

length (average conversation time for the voice). Hence, if

for instance the comparison is by means of the same offered

load, the voice signals with a longer average conversation

time have smaller generation rate and service completion

rate than the data packets. Hence, the probability of the

number of voice channels varying in one data packet length

is quite small so that the number of voice channels can be

considered constant. In actual numerical terms, the follow-

ing is given. Let the offered load of the voice users be Gv

and the average voice engaged time be Tv. When a data

packet is generated, the number of existing voice users is

kv. The average number of the new voice users kv_birth in the

data packet length Td is the product of the generation rate

Ov of the voice user and Td:

Further, the average number kv_birth of the existing kv voice

users to complete within the data packet length Td is given

as follows in terms of the voice conversation completion

rate Pv:

If Gv = 50 and kv = 50, then each value above is kv_birth =

0.0125 and kv_death = 0.0125. Here, the data packet length is

500 bits, the average voice conversation time Tv = 60 s, and

the bit rates are 32 kbps for both. From this, it is considered

that generation and completion of the voice user are almost

nonexistent within one data packet length. Of course, this

value changes with the condition. If the offered load of the

voice signal becomes very large to the extent that the

number of voice channels changes within one data packet

length, this assumption can no longer be used. However, as

a practical problem, if the offered load of the number of

voice users becomes very large, the system no longer holds.

Therefore, the offered load of the voice users in the range

shown in this paper is considered sufficient. Hence, this

assumption is considered sufficiently effective.

2. Reason Why the Number of Simultaneously

Transmitting Stations for the Data Can

Be Fixed When the Blocking Probability

of the Voice User Is Derived

Since the generation of the reservation packets of the

voice signal is controlled by CLSP, it is dependent on the

instantaneous number of signals (data packets and number

of engaged voice users) at the central station. Hence, what

is important at the generation of the new voice user is only

the instantaneous number of signals at the central station.

Whether the voice user can be accepted for reservation,

namely, the blocking probability, depends only on the in-

stantaneous number of signals. What was different from the

data analysis presented above is that it was important that

the number of simultaneously transmitting stations not

change during a certain time period (data packet time) so

that the number of voice signals was fixed. However, at the

present time, it is important how the number of signals from

one instance to another is given. Since the steady-state

conditions are considered in this paper, the number of

simultaneously transmitting stations for the data packets

obtained from the steady probability given by M/ D/D /Dis used for the value. Hence, the desired blocking prob-

ability is obtained from the coupling probability of Eq. (16).

69

AUTHORS (from left to right)

Takeshi Sato (student member) graduated from the Department of Electrical Engineering, Nagoya University, in 1994

and completed his doctoral course in 1998. He holds a D.Eng. degree. He has been engaged in research on packet communication

using spread spectrum communication system. He is a member of IEEE.

Abbas Sandouk (student member) graduated from the Department of Electrical Engineering, University of Damaskus,

in 1991 and completed his master�s course in 1998. He is presently a doctoral student. He has been engaged in research on

packet communication using spread spectrum communication system. He is an IEEE member.

Takaya Yamazato (member) graduated from the Department of Electrical Engineering, Shinshu University, in 1988 and

completed his master�s course in 1990. In 1993, he completed his doctoral course at Keio University. He then became a research

associate at Nagoya University. In 1998, he became an associate professor in the Information and Media Education Center of

the university. He holds a D.Eng. degree. He has been engaged in research on coded modulation methods, satellite communi-

cation measurement, and communication theory. He is a member of the Information Theory and Application Society and IEEE.

Masaaki Katayama (member) graduated from the Department of Communication Engineering, Osaka University, in

1981 and completed his doctoral course in 1986. He then became a research associate at Toyohashi University of Technology.

In 1989, he became a lecturer at the Computation Center, Osaka University. In 1992, he became a lecturer in the Department

of Electrical and Computer Engineering, and was promoted to an associate professor in 1993. He holds a D.Eng. degree. He

has been engaged in research on satellite communication and measurement, spread spectrum communication, modulation/de-

modulation theory, noise theory, traffic theory, and computer network. In 1986, he received a Shinohara Memorial Award. He

is a member of the Information Theory and Application Society and IEEE.

Akira Ogawa (member) graduated from the Department of Electrical Engineering, Nagoya University, in 1960 and joined

Kokusai Denshin Denwa. After serving at its Research Laboratory, he joined Nagoya University in 1988. He has been engaged

in research and development of digital communication, satellite communication, and mobile communication. Presently, he is a

professor. He holds a D.Eng. degree. He coauthored Satellite Communication Technologies. He is a member of IEEE.

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