Wireless Netw (2008) 14: 17–28

DOI 10.1007/s11276-006-7522-1

Performance analysis of a multiple access protocol for voiceand data support in multiuser broadband wireless LANsR. Fantacci · G. Vannuccini · G. Vestri

Published online: 9 June 2006C© Springer Science + Business Media, LLC 2006

Abstract Wireless Local Area Networks (WLANs) is one

of the most promising access technologies for the upcom-

ing fourth-generation wireless communication systems. In

the last few years, several research efforts have been de-

voted to investigate possible multiple access schemes capa-

ble of supporting real-time traffic as well as best-effort data

transmissions. In particular, the use of suitable transmission

schemes allows not only to achieve higher data-rates, but also

to perform resource allocation in order to guarantee specific

service requirements. In this paper we propose a medium

access control (MAC) scheme for a WLAN supporting real-

time (voice) and best-effort (data) services, based on Orthog-

onal Frequency Division Multiplexing (OFDM) technique.

A suitable analytical approach is proposed in order to derive

the performance of the proposed MAC scheme. In particular,

it is shown in the paper that a high overall network capac-

ity in terms of simultaneously active users is achieved by

effectively exploiting the multiuser capabilities offered by

OFDM, together with a proper service differentiation.

Keywords Wireless networks · Multiple access · OFDM

1. Introduction

In the last few years, the use of more efficient transmission

techniques has allowed to obtain higher data rates in Wireless

Local Area Networks (WLANs), thus making their use more

effective in broadband communication systems and more

Work partially supported by MIUR within the WOMEN project.

R. Fantacci · G. Vannuccini ( ) · G. VestriElectronics and Telecommunications Department, University ofFirenze Via di Santa Marta 3, 50139 Firenze, Italye-mail: g.vannuccini@ieee.org

competitive as to the Fast and Gigabit Ethernet wired-LAN

technologies. Several standardization efforts were made on

the optimal transmission and medium-access techniques for

WLAN applications. The two main high data-rate WLAN

standards, namely IEEE 802.11a [1, 2] and HIPERLAN type

2 [3], essentially introduce the same innovation in the phys-

ical layer, i.e., Orthogonal Frequency Division Multiplexing

(OFDM), both being based on the 5 GHz frequency band.

Another recent standard, discussed in more detail in Sec-

tion 3, which is going to be significantly widespread in the

Wireless Metropolitan Area Network scenario is the IEEE

802.16 − 2004, that is also based on OFDM. The service

offered by current WLAN systems is basically a best-effort

type, which means that no guarantees are provided to the end

user with what concerns the service performance. Significant

QoS enhancements to the WLAN standards have been devel-

oped within the HIPERLAN type 2, IEEE 802.11 Point Coor-

dination Function, and the new IEEE 802.11e MAC standard

[4]. The performance of such solutions have been previously

analyzed in the literature [5, 6]. However, to our knowledge,

none of these techniques is currently widely deployed on the

market. As pointed out in [7, 8], the use of specific trans-

mission techniques at the physical layer would allow a more

efficient use of the available bandwidth, thus providing end

users with a suitable QoS. In this work, we propose a MAC

protocol which takes advantage of the sub-channel division

operated by the OFDM technique. As we will show in the fol-

lowing, the considered protocol adopts a Frequency Division

Duplex (FDD) structure together with a packet reservation

scheme, thus allocating a large number of voice users within

a broadband WLAN endowed with the OFDM physical layer,

while voice and data QoS requirements are properly satisfied.

In Section 2 the considered OFDM signal processing scheme

for the proposed access protocol is presented. In Section 3 the

access scheme for both voice and data terminals is described,

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18 Wireless Netw (2008) 14: 17–28

and its analytical model is derived in Section 4. Finally, in

Section 5 numerical results are shown for the considered sys-

tem, while comparing analytical predictions with computer

simulation results.

2. OFDM signal processing for the proposedaccess scheme

OFDM is a well-known transmission technique that allows

a single signal to be transmitted over different parallel sub-

carriers. It is more resistant to frequency selective fading than

single carrier systems, due to the segmentation of the chan-

nel into narrowband flat fading sub-channels. It also avoids

Inter Symbol Interference (ISI) through the use of a cyclic

prefix. Among the main drawbacks of OFDM, there are time

and frequency synchronization requirements, due to a strong

performance variation with frequency and time offsets. Such

issues can be partially mitigated by effectively employing

frequency, time and power control techniques, that bring to

the possibility to assign different sub-carriers to different

users [7, 8]. For the system under consideration we assume

an FDD frequency structure, where 20 MHz-wide uplink and

downlink bandwidth are divided into Nch = 4 sub-channels

of Nc = 16 comb-spread sub-carriers each. The total number

of available sub-carriers is then 64 for both the uplink and the

downlink. A comb-spread allocation scheme can be assumed

to cope with selective fading of the radio channel in an effi-

cient way [7–9]. The system proposed is modelled with an

infrastructure-WLAN with an Access Point (AP), Nd data

terminals (DTs) and Nv terminals (VTs). In the proposed

access scheme, available resources for uplink or downlink

transmissions can be depicted as a time-channel (T-C) ma-

trix. The rows of the T-C matrix are the 4 available frequency

sub-channels, each sub-channel containing 16 comb-spread

carriers. Columns of the T-C matrix are the time slots. In par-

ticular, 50 slots form a frame, whose duration T f is 20 ms.

Each element of the T-C matrix corresponds to a voice or data

packet allocated to a specific uplink or downlink sub-channel

(i.e., a specific group of carriers), via the reservation-based

process that will be discussed in the following. Every VT

can transmit its traffic on a group of Nc carriers which the

AP has allocated for it. A DT can transmit on one or more

sub-channels not reserved by VTs.

A time slot contains 100 minislots. Each minislot corre-

sponds to an OFDM symbol time of 4 μs. An OFDM symbol

time is composed of an useful interval Tu(3.2 μs) and a guard

interval Tg(0.8 μs) to cope with multipath fading.

Each voice or data message is finally fragmented into a

164 byte-long packet, which composes the payload of a MAC

frame, as shown in Fig. 1. After the channel encoding (with

rate 1/2), such a payload is mapped into 82 minislots, i.e.,

82 OFDM symbols. The remaining 18 OFDM symbols that

Fig. 1 MAC and PHY packet format

compose a time slot are used, as shown in Fig. 1, for MAC

signalling, physical layer synchronization and channel esti-

mation. The overall MAC protocol efficiency ratio for the

considered system is therefore 82%.

In particular, 4 OFDM symbols are used at the MAC layer

for addressing, signalling of the available subchannels to

DTs and VTs, and generic control signalling for the WLAN

management.

Then, as shown in Fig. 1, 14 OFDM symbols at the physi-

cal layer are assumed to be sufficient to achieve symbol syn-

chronization among stations and channel estimation. Such

a synchronization at the OFDM symbol, coupled with a

proper WLAN-wide clock broadcasting system, may allow

to develop an FDD time-frequency division multiple access

scheme such as the one proposed in this paper.

The QPSK modulation scheme was assumed in transmis-

sion for its simplicity and robustness, despite it is certainly not

always the optimum solution for such a broadband wireless

system. However, more efficient solutions as those based on

link adaptation algorithms, that adapt the modulation scheme

used in transmission to the channel conditions, are beyond

the scope of this paper.

In general, a terminal could avoid transmitting on some

of the carriers by zeroing some of the inputs of the IFFT and

using only the remaining inputs, corresponding to the carriers

allocated to it by the AP, to transmit its information. The AP

is able to separate the uplink information flows of different

users thanks to the linearity of IFFT processing, and to the

assumption of mini-slot synchronization among users. The

centralized structure of the proposed protocol allows the AP

to know at each time to which user a certain sub-channel is

assigned.

The receiver basically performs the reverse operations of

the transmitter. Once the received signal has been trans-

lated into the base band, cyclic prefix is extracted and the

useful samples are serial to parallel converted. These data

are then processed by a FFT block. As it will be shown in

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Wireless Netw (2008) 14: 17–28 19

the next sections, every receiver knows which sub-channels

are used for packets addressed to it. By properly tuning on

these sub-channels, QPSK demodulators recover the infor-

mation out of the OFDM symbols.

3. Description of the access scheme

3.1. VT and DT access scheme

Voice signals are known to have a two-state ON/OFF be-

havior, where talkspurts (ON state) are followed by silence

periods (OFF state) [10]. The ITU-T G.711 speech codec

[11] has been selected to model good-quality voice calls,

with a 64 kbit/s bit rate, 160 byte-long packets generated ev-

ery 20 ms during a talkspurt, and no packets generated in

the silence periods. The RTP/UDP/IP header overhead was

considered, but as the packet length is very limited, a header

compression was assumed, which allows to compress the 40

bytes of the RTP/UDP/IP header into 4 bytes [12]. The talk-

spurt and silence duration times are exponentially distributed

with a mean value of 1 s and 1.35 s, respectively [10]. It is

assumed here that the maximum tolerable delay for a voice

packet is equal to a frame time T f , which is also the vocoder

inter-packet generation time. If a voice request packet cannot

be sent to the AP within this delay, it is dropped. Before the

end of each slot, the available OFDM sub-channels which

can be used for both contention and data transmission in

the following slot are broadcasted by the AP in downlink

to all terminals via a specific field of the MAC packet. In

particular, the AP exploits the 8-bit-long subcarrier alloca-

tion signalling field in the downlink in order to broadcast

which sub-channels are free and which are busy, in this case

also the terminals using them. This is done by exploiting the

information in the time-channel allocation matrix which is

managed by the AP. In particular, the AP will send on the i-th

downlink channel the address of the station which will use

the corresponding i-th uplink channel, on which the station

required to transmit by sending its transmission request.

Each VT or DT waits for the AP to acknowledge the suc-

cessful transmission of the channel request, via a grant mes-

sage on the required downlink channel. Whenever a talkspurt

begins at a particular VT, the access phase to the shared chan-

nel resources is started, by sending a preemption symbol to

the AP at the beginning of the next slot. The AP will then

preempt possible active DT transmissions in favor of the VT

transmissions. In the following slot, if any free channels are

available, the VT transmits its request packet with permis-

sion probability pv over one of the free channels, chosen

with uniform probability Fig. 2. Otherwise, the VT sends

another preemption symbol at the beginning of the current

slot. If the request packet is successfully received by the AP,

as shown in Fig. 2, the AP will then assign to the specific

VT that sub-channel every frame time for the transmission

of voice packets. If the request packet collides with other re-

quest packets, the terminals that caused the collision, having

not received any grant from the AP, will try to transmit their

request packets again in the following slots. The DT on ser-

vice, whose transmission suffered preemption, is enabled to

transmit on the first slot following the slot in which collisions

and preemption requests are absent, as shown in Fig. 2.

Once the VT has gained the required resources for a spe-

cific talkspurt, the channel access is guaranteed to that VT,

which will be able to send its packets avoiding collisions as

in Fig. 2. When a VT talkspurt ends, the VT will set a flag in

its last voice packet. As a consequence, the allocated channel

resources are released by the AP.

Every DT has a FIFO-served messages buffer. Data mes-

sages are segmented into fixed length (164 byte) packets be-

fore being forwarded. In order to transmit a message, a DT

has to send a data request packet to the AP, following the

same scheme as outlined for VTs. This contains the sender

address, the receiver address, and the first part of the infor-

mation message.

A data request will be valid for all messages generated

by the DT since the arrival of the first message to the instant of

the complete depletion of the data buffer (exhaustive policy).

The AP contains a FIFO-served Request Queue (APRQ). If

a data request is successfully received, it will be put at the

end of the APRQ (FIFO policy). When a DT request reaches

the head of the APRQ, that DT will begin to be served, i.e.,

it will be able to send its data packets in the sub-channels not

reserved by VTs (up to Nch per slot). Before the end of each

slot, the available sub-channels for the next slot are broadcast

by the AP. In the above scheme, it is possible that a DT

request collides with other DT requests and/or VT requests.

Preemption is also adopted for DTs over the currently active

DT, in order to avoid starvation for data requests.

The proposed algorithm was initially conceived while

an important standardization activity was carried out within

the IEEE Working Group 802.16, which brought to the 2004

IEEE 802.16–2004 MAC and PHY specifications [13]. The

IEEE 802.16 specification was first thought for point-to-

multipoint broadband wireless networks, in a metropolitan

scenario [14]. The IEEE 802.16 standard specifies both PHY

and MAC layer, even if scheduling and resource management

details are intentionally left to different vendor implementa-

tions. The scheme considered in this paper is intended to

propose a scheduling technique for the allocation of network

resources with regards to QoS demands of voice and data

services. The proposed scheme is also analyzed by means

of computer simulations and a theoretical model of both the

protocol and the voice and data sources. The algorithm pre-

sented in this paper offers several interesting similarities with

the IEEE 802.16 standard. In particular, we deal with a time-

slotted/FDD multiple access protocol, based on the OFDM

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20 Wireless Netw (2008) 14: 17–28

uplink

downlink

frequency

timeFrame time: 20ms

Slot time: 400 s

WSTA5 voice req+ first voicepacket

WSTA 3 datarequest

WSTA 4 secondvoice packet

WSTA 2 and 3:Grant notreceived

WSTA 2 Voicereq+ first voicepacket+

WSTA 3 data req

WSTA2 voicereq+ first voicepacket

WSTA 1 Voicereq + first voicepacket

WSTA5 voice req+ first voicepacket

WSTA 3 datarequest

WSTA 4 secondvoice packet

WSTA 2 and 3:Grant notreceived

WSTA 2 Voicereq+ first voicepacket+

WSTA 3 data req

WSTA2 voicereq+ first voicepacket

WSTA 1 Voicereq + first voicepacket

Grant to WSTA 3Channel3

Grant to WSTA 2Channel1

Grant to WSTA 5channel4

Grant to WSTA 1Channel1

Grant to WSTA 3Channel3

Grant to WSTA 2Channel1

Grant to WSTA 5channel4

Grant to WSTA 1Channel1

WSTA 5 Secondvoice packet

WSTA 4 Thirdvoice packet

WSTA 1

SecondVoicepacket

WSTA3 datapacket

WSTA2 secondvoice packet

WSTA 5 Secondvoice packet

WSTA 4 Thirdvoice packet

WSTA 1

SecondVoicepacket

WSTA3 datapacket

WSTA2 secondvoice packet

Fig. 2 Uplink and Downlink access scheme for the considered system

capabilities of allocating different channels to different ser-

vices and users. This is a similar scheme to the one included

in the Wireless MAN-OFDMA option of the IEEE 802.16–

2004 standard [13]. However, although the basic concepts

are similar, the algorithm presented in this paper is consid-

erably simpler than the IEEE 802.16 MAC and PHY spec-

ifications, having here focused more on the implementation

of a computer simulation and analytical approach to the per-

formance evaluation of a proposed OFDM-based multiple

access scheme for broadband wireless networks. Therefore,

the Uplink and Downlink MAP, used in the IEEE 802.16

PHY to allow the Base Station (BS) and the Service Stations

(SSs) to select the proper physical transmission parameters

(e.g., FEC and modulation), is much simpler in this paper,

where a lookup-table processed by the AP is conceived to

optimize and manage the channels allocation to the different

terminals. The transmitted data burst-size profile, which is

dynamically negotiated in the IEEE 802.16 specifications,

is here reduced to a simpler allocation scheme, where the

AP decides to allocate more channels to DT services thus

obtaining a parallel transmission of more data units, which

can still be thought of as a dynamic burst. Like the IEEE

802.16 MAC, the MAC scheme under consideration is basi-

cally connection-oriented, since each service is mapped on

a connection, which is granted with specific QoS charac-

teristics, depending on the considered service. Again, this

connection-oriented resource management is implemented

in a complex and efficient way in the IEEE 802.16 standard,

by using the Connection Identifier (CID) field in the MAC

header, which is used to address each connection between the

BS and the SS in the IEEE 802.16 WMAN. This addressing

is extremely simplified in the proposed algorithm, where a

MAC field (the subcarrier allocation signalling field) is used

for the same aim of addressing and communicating with ter-

minals allowed to transmit. The IEEE 802.16 MAC header

is endowed with a Bandwidth request field, which again is

simplified in the considered protocol in a proper request mes-

sage encoded within the signalling field. Several similarities

can also be found in the concept of resource allocation ser-

vices of the IEEE 802.16 MAC. In particular, the considered

algorithm allocation scheme could be compared to the Real-

Time Polling Service of the IEEE standard, where an SS can

send an explicit request-to-transmit to the BS. As pointed out

in [14], when compared to the other allocation service pro-

posed in the IEEE 802.16, i.e., the Unsolicited Grant Service,

where no explicit request is allowed to SSs, the Real-Time

Polling Service can increase the latency and protocol over-

head, but it allows the AP to allocate as much bandwidth

as required by terminals, thus optimizing resource allocation

in the network. For what concerns the grant mechanism, the

considered algorithm is similar to the Grant per Connection

(GPC) scheme of the IEEE 802.16 MAC, since each channel

allocation sent by the AP is granted to a specific DT or VT

transmission, which can be thought of as a connection as

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Wireless Netw (2008) 14: 17–28 21

meant by the IEEE 802.16 standard, since we assume that

each VT or DT is represented by a single service.

4. Voice and data analytical model

In deriving the performance of the proposed scheme, a the-

oretical model was developed for the overall system. We

focused on the uplink channel, since it is the most critical

channel due to the possibility of collisions among terminals

accessing the shared resources. The principal aim of this pa-

per was to analyze the performance of the MAC protocol,

and, hence, the effects of the radio channel were neglected

as a first approximation. This may be further justified by

assuming such effects to be partially mitigated by using suit-

able coding, interleaving, windowing and comb spreading

techniques [7, 9].

The system parameters considered in this paper are given

in Table 1.

The voice subsystem, shown in Fig. 3, was modelled as in

[10]. Voice packet dropping probability, Pdrop, was consid-

ered as the target QoS parameter for voice services.

Each VT can be in one of Ns + 2 states. All transitions

are assumed to occur at the end of a slot. A VT remains

in the silent state SIL unless it has no speech packets to be

transmitted on the channel. This happens during the silent

time of the voice model. As soon as the talkspurt period

begins, and the first packet is generated within this talkspurt,

the VT leaves the SIL state and moves to the contending

state CON. After that a VT has successfully transmitted its

request packet, it can be in one of the Ns reservation states

RESi (i = 0, . . . , Ns − 1).RESi denotes the state occupied by

a VT which has to wait i slots before transmitting a voice

packet on its reserved sub-channel.

From state RESi , i �= 0, a VT always moves to state

RESi−1. From state RES0, it returns to state RESNs−1 if it

continues to generate voice packets, otherwise it goes to state

SIL. We assume that no transitions from state CON to SILstate can occur, i.e., a talkspurt never ends before a reserva-

tion attempt becomes successful.

For the DT subsystem, a finite-state chain model was

adopted [15] Fig. 3. The data message delay, Tmes, is the

QoS parameter to be evaluated. A DT can be in one of the

following states:� SIL: a DT has no data messages to transmit in its buffer,

this implies that the DT remains in the silent state SIL;� CON j : a DT stays in this state if it is in the contention

phase and its buffer contains j messages;� WAIT j : a DT remains in this state if its address is stored

in the APRQ and its buffer contains j messages.

From state SIL, a DT moves to state CON1 if, in a specific

slot, a message is generated. From state CONi , a DT passes

Table 1 System parameters

Default

Symbol Definition value

System parametersNch number of sub-channels per slot 4

Nc number of carriers per sub-channel 16

Ns number of slots per frame 50

TOFDM OFDM symbol time 4 μs

Ts slot time 400 μs

T f frame time 20 ms

sv mean # of VTs in Silence at eq.

cv mean # of VTs in Contention at eq.

rv,i mean # of VTs in Res - i at eq.

sd mean # of DTs in Silence at eq.

cd mean # of DTs in Contention at eq.

wd mean # of DTs in Wait at eq.

Voice terminal parametersσv prob. that talksp. starts in a slot, expon. distr.

γv prob. that talksp. ends in a slot, expon. distr.

γ fv prob. that talksp. ends in a frame, expon. distr.

voice packet time-out 20 ms

Nv number of active VTs

pv permission probability for VT

Data terminal parametersλd mean message interarrival time, exp. distr.

Ld mean message length, mod. geom. distr.

σd prob. that message starts in a slot, exp.

distr.

γd prob. that message ends in a slot, exp. distr.

Nd number of active DTs

pd permission probability for DT

into the waiting state WAITi if no messages are generated

and the data request is successfully forwarded. The DT may

also go from state CONi to state WAITi+1 if there is a new

message arrival and the DT request is successfully sent to

the AP. Finally, the DT can move from state CONi to state

CONi+1 if a new message is generated and the request is

not successfully sent. From state WAITi , a DT can move to

WAITi+1 if a new message is generated and the DT request

is not at the end of the APRQ, or if a new message arrives,

the DT request is at the end of the APRQ and the currently

served message transmission is not terminated. From state

WAITi+1, a DT can pass to state WAITi and from state WAIT1

it can pass to state SIL if no message is generated, the DT

request is at the end of the APRQ and the served message

transmission is completed. A DT remains in state WAITi if

one of the following cases applies: (a) its request is not at the

end of APRQ and no new messages are generated, (b) no new

messages are generated, the request is at the end of APRQ

and the served message transmission is not terminated, (c) a

new message arrives, the request is at the end of APRQ and

the served message transmission is completed.

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22 Wireless Netw (2008) 14: 17–28

v

fv

Pcrv

1-Pcrv

SIL CON

RESN -1sRESN -2sRES0

fv

v

CON1 CON2

SIL

WAIT1 WAIT2

(1-Pod) d(1-Pod) d

(1-P )(1- )od d (1-P )(1- )od d

d odP

(1- ) Pd od

d

d1-

(1- ) Pd ow d

(1-P )(1- )+(1- ) P (1- Pow d d ow d)+ d ow d (1-P )(1- )+(1- ) P (1- Pow d d ow d)+ d ow d

(1-P ) + P (1-ow d d ow d)

(1- ) Pd ow d

(1- ) Pd od

(1- ) Pd ow d

(1-P ) + P (1-ow d d ow d)

(1-Pod) d

d odP

Fig. 3 Voice and data subsystem models

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Wireless Netw (2008) 14: 17–28 23

4.1. system analysis

By using the Equilibrium Point Analysis (EPA) method

[16], it is possible to determine the equilibrium values

of voice and data subsystem state variables, described in

Table 1, respectively as {Sv, Cv, Rv,0, Rv,1, . . . , Rv,Ns−1} and

{Sd , Cd , Wd}. According to Table 1, the equilibrium values

of these variables are hereafter referred with the following

lowercase letters notation: {sv, cv, rv,0, rv,1, . . . , rv,Ns−1} and

{sd , cd , wd}.1The system considered in this paper is assumed to be un-

der non-heavy load traffic situation (stability region). At the

equilibrium, the expected rate at which terminals leave a

given state is equal to the expected rate at which terminals

enter this state. Let us now write the equations necessary to

determine the equilibrium values of state variables. First of

all we will concentrate on voice subsystem. The probability,

for a VT, to move from state CON to state RESNs−1, i.e, to

successfully forward its request packet is shown in 1 (see

table 4 for parameters description):

Pcrv(cv, cd , rv) =

⎧⎪⎪⎪⎪⎨⎪⎪⎪⎪⎩Nch∑j=1

pv

(1 − pv

j

)cv−1 (1 − pd

j

)cd(

Nch

j

) (rv

Nch

)(Nch− j) (1 − rv

Nch

) j

if cv ≥ 1

Nch∑j=1

pv

(1 − pd

j

)cd(

Nch

j

) (rv

Nch

)(Nch− j) (1 − rv

Nch

) j

if cv < 1

(1)

The parameters used are detailed in Table 4.

This is obtained by considering that:� each VT has a permission probability pv;� each DT has a permission probability pd ;� in general there may be j free sub-channels per slot, among

which a VT chooses only one with a uniform probability1j ;� if cv ≥ 1, a VT reservation will be successful if none of

the other (cv − 1) contending VTs and none of the other cd

contending DTs chooses the same sub-channel to transmit

its own request;� the probability that there are j free sub-channels in a certain

slot is binomial and equal to(Nch

j

) (rv

Nch

)(Nch− j) (1 − rv

Nch

) j

; (2)

� if cv < 1, a VT reservation will be successful if none of the

other cd contending DTs chooses the same sub-channel to

transmit its own request.

1 Note that, being mean values, the equilibrium state variables can alsotake values less than 1.

By equating the outflow and the inflow at states

RES0, RES1, . . . , RESNs−2, we have:

rv,0 = rv,1 = . . . = rv,Ns−1 = rv. (3)

Equilibrium equations at states RESNs−1 and SIL can thus

be written as follows:

rv(1 − γ fv ) + cv Pcrv = rv, (4)

rvγ fv = svσv (5)

Recalling that the total number of VTs in the network is

Nv , we have, at equilibrium

sv + cv + Nvrv = Nv. (6)

Let us now consider data subsystem. The term

Pod (cv, cd , rv), which denotes the probability for a DT to

successfully send a request packet, is formally identical to

(1), except that the terms pv and pd , and the terms cv and

cd in (1) are mutually exchanged. For state CONi , through

some calculations, we have

cdi = sd

1 − Pod

[σd (1 − Pod )

σd + Pod + σd Pod

]i

. (7)

Finally, we can write

cd =∞∑

i=1

cdi = σd · sd

Pod. (8)

Recalling that the total number of DTs in the network is

Nd , we have, at equilibrium,

sd + cd + wd = Nd (9)

where wd = bNd , with b indicating the probability for a DT

to have its request in the APRQ.

At equilibrium, the average number of packets arriving at

the DTs within a slot time is equal to the average number

of data packets sent within a slot time, given by the number

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24 Wireless Netw (2008) 14: 17–28

of packets sent by the active DT not blocked by the VTs

preemption, plus the sum of the packets sent by the DTs being

in Contention state, cd Pod . The above statement is expressed

in the last equation of the following nonlinear system, which

is used to derive equilibrium values (cv, rv, sv, cd , sd , b) of

system state variables:

⎧⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎨⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎩

rv(1 − γ fv ) + cv Pcrv = rv

rvγ fv = svσv

sv + cv + Nsrv = Nv

cd = σd ·sdPod

cd + sd + bNd = Nd

λd Nd Ld Ts = [1 − (1 − b)Nd ][1 − PITD].Nch∑j=1

j

(Nch

j

) (1 − rv

Nch

) j (rv

Nch

)Nch− j

+ cd Pod .

(10)

The term PITD represents the probability for the current active

DT to be preempted, and is given by:

PITD = 1 − (1 − σv)sv (1 − σd )sd Pcv

crv Pcdod . (11)

From the solution of the nonlinear system (10), equilib-

rium values of state variables are derived. These are taken as

average values of corresponding state variables and used to

build the system state probability �(Cv, Rv, Cd , Wd ).

By applying the chain rule several times [15], we have:

�(Cv, Cd , Wd , Rv)

=(

Nd − Wd

Cd

) (cd

Nd

)Cd(

1 − cd

Nd

)Nd−Wd−Cd

×(

M − Rv Ns

Cv

) (cv

Nv

)Cv(

1 − cv

Nv

)M−Rv Ns−Cv

×(

Nch

Rv

) (rv

Nch

)Rv(

1 − rv

Nch

)Nch−Rv

×(

Nd

Wd

)bWd (1 − b)Nd−Wd . (12)

where M = max(Nv, Nch Ns).

4.2. QoS parameters evaluation

Let v(Cv, Rv) be the probability for a VT to remain in the

CON state for a given slot time, when the number of con-

tending VTs Cv and voice-reserved sub-channels Rv are

fixed:

v(Cv, Rv) =

⎧⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎨⎪⎪⎪⎪⎪⎪⎪⎪⎪⎪⎩

1−Nch∑j=1

pv

(1− pv

j

)Cv(

Nch

j

)·(

Rv

Nch

)Nch− j (1 − Rv

Nch

) j

if Cv ≥ 1,

1 − pv

[1 −

(Rv

Nch

)Nch]

if Cv < 1.

(13)

We have that vm , the average probability that a VT is not

able to successfully send a transmission request packet to the

AP within a slot time, results to be:

vm =Nch∑

Rv=0

M−Rv Ns∑Cv=0

v(Cv, Rv) · �(Cv, Rv). (14)

At the beginning of a talkspurt a VT waits j slots before

obtaining a reservation. This waiting probability conforms to

the geometric distribution:

PW ( j) = (1 − vm) v j−1m , j = 1, 2, . . . . (15)

Let us assume that a talkspurt is L packets long. No packets

are dropped if j ≤ Ns . After waiting Ns slots without a suc-

cessful request transmission, the VT drops the initial packet

of the talkspurt. The VT will drop the following packet if it

cannot be successfully forwarded within the next Ns slots. A

similar treatment is applied to the following packets of the

talkspurt. All the packets of the talkspurt will be dropped if

j ≥ L Ns + 1. The number of dropped packets as a function

of waiting time j can be expressed as follows:

ndrop( j) = 0 if 1 ≤ j ≤ Ns

ndrop( j) = k (0 < k < L) , if k Ns + 1

≤ j ≤ (k + 1)Nsndrop( j) = L if L Ns + 1 ≤ j

(16)

From (15) and (16) it is possible to derive the probability

that, in a talkspurt of L packets, k packets are dropped:

Pr (ndrop = 0|L) =Ns∑j=1

PW ( j) = 1 − vNsm

Pr (ndrop = k|L) =(k+1)Ns∑

j = k Ns+1

PW ( j) = vk Nsm − v(k+1)Ns

m

Pr (ndrop = L|L) =∞∑

j = L Ns+1

PW ( j) = vL Nsm

, if 1 ≤ k ≤ Ns − 1 (17)

Springer

Wireless Netw (2008) 14: 17–28 25

The mean number of dropped packets for a talkspurt of

length L is given by:

E[ndrop|L] =L∑

k = 0

k Pr (ndrop = k|L) = vNsm

1 − vNs Lm

1 − vNsm

. (18)

By assuming that the probability Pr(L) of a talkspurt

length is geometrically distributed:

Pr (L) = γ fv (1 − γ fv )L−1, (19)

we calculate the average number of dropped packets per

talkspurt:

E[ndrop] =∞∑

L=1

E[ndrop|L]Pr (L) = vNsm

1−(1−γ fv )vNsm

. (20)

Finally, Pdrop is obtained as the ratio of E[ndrop] over the

average number of packets per talkspurt, i.e., 1/γ fv :

Pdrop = γ fvvNs

m

1 − (1 − γ fv )vNsm

. (21)

As for data subsystem, the performance evaluation is car-

ried out by evaluating the total message transmission delay,

within the uplink from a DT to the AP. The data message

transmission delay is most probably higher in the uplink than

in the downlink, due to the negative effect of the contention

mechanism. The total message delay Tmes can be calculated

as the sum of four terms:� the time spent to successfully forward the data request

to the AP, which is nonzero only if a message arrives find-

ing the buffer empty;� the time Tch necessary to transmit a message over the chan-

nel;� the time TqueueAP needed for the request to reach the top of

the APRQ;� the time TqueueDT necessary for a message to arrive at the

top of DT messages buffer.

First of all we note that a message arrived at a DT finding

the buffer empty has a different service time statistic with

respect to messages arrived when the buffer is not empty.

In the former case the service time is given by the sum of

the contention time (i.e., the time spent to successfully send

the request packet) plus the time necessary to complete the

message transmission. In the latter case the service time is

equal to the time needed to transmit the message over the

channel. Hence, we can model each DT as an M/G/1 system

with different service time statistics [17]. The mean message

delay for the considered M/G/1 system is

Tmes � [2 ˙A(1) + ¨A(1) − A(1)] + A(1)

2λd [1 − A(1) + A(1)](22)

where A(z) and A(z) are the message arrivals generating

functions in the cases of, respectively, non-empty and empty

buffer. Under the assumption that a message is composed of

k packets, by means of standard queuing results, we have

A(z|k) =∞∑

i = 0

(λdkTseq)i 1

i!e−λd Tseq zi (23)

where Tseq = Tsν

is the equivalent slot time for a DT, which

is the real slot time perceived by the active DT which is

attempting to transmit its packets. The term v is the average

number of channels per slot available for data transmission:

ν =Nch∑

Rv=0

M−Rv Ns∑Cv=0

Nd∑Cd=0

Nd−Cd∑Wd=0

(Nch − Rv)(1 − σv)M−Cv−Rv Ns

× (1 − σd )Nd−Cd−Wd PCdcrv PCd

od �(Cv, Cd , Wd , Rv). (24)

If the message length is geometrically distributed with

mean Ld , then A(z) is obtained by averaging A(z|k) over k.

Let us now evaluate A(z). The service time for a message

finding an empty buffer is given by the sum of the time spent

for the contention phase plus the time needed for the message

transmission over the channel. One slot is spent for sending

the preemption symbol. If i slots are necessary for success-

fully sending the request and if the message is composed of

n packets, we have:

A(z|i, n) = e−λd (i+1+ n−1ν

)T s(1−z). (25)

The probability for a DT to remain in the contending state

for i slots and the probability for a message to be n packets

long are geometric and equal to

Pcon(i) = Pod (1 − Pod )i−1 (26)

PL (n) = 1

Ld

(1 − 1

Ld

)n−1

(27)

The joint probability for a DT to remain in the contending

state for i slots and for a message to be n packets long is

given by the product of the two previous terms. Finally we

can calculate

Springer

26 Wireless Netw (2008) 14: 17–28

A(z) =∞∑

i=1

∞∑n=1

A(z|i, n)Pcon(i)PL (n). (28)

5. Numerical results

In order to make the performance evaluation for the proposed

scheme analytically and computationally feasible, a scaled

version of the whole system was considered in which Ts =2 ms, Ns = 10, T f = 20 ms, Nch = 4.

Figure 4 shows the behavior of the packet dropping prob-

ability (Pdrop) as a function of Nv , when no DTs are present,

for three different values of pv . If the total number of VTs

is comparable with the total number of channels per frame

time (Ns Nch = 40), then Pdrop minimization will be achieved

with higher pv (close to 1). On the contrary, if the VTs num-

ber is considerably larger than the number of voice channels

per frame time, then a smaller pv will achieve better per-

formance. This shows how smaller permission probabilities

allow us to allocate larger number of active terminals in the

network.

Figure 5 shows Pdrop as a function of pv , when Nv is

fixed. The dropping probability Pdrop shows now a minimum

within the pv interval [0,1]. The Pdrop dependence on pv can

be explained by considering that if permission probability

is too small, voice packets drop is most probable, because

their transmission is more difficult, due to the low pv , while

if it is too large, packets drop is caused most probably by

collisions. An intermediate value for pv of 0.7 has proven to

be the optimum value that minimizes Pdrop in this case. An

important consequence coming from the proposed protocol is

the independence of voice service quality from data traffic,

due to the DTs preemption in favor of VTs. Accordingly,

Fig. 6 shows the variation of Pdrop with Nv in the cases for

Nd = 0 and Nd = 10.

10 20 30 40 50 60 70 8010

–7

10–6

10–5

10–4

10–3

10–2

10–1

Nv

Pdro

p

simulation pv=0.3

theory pv=0.3

simulation pv=0.5

theory pv=0.5

simulation pv=0.7

theory pv=0.7

Fig. 4 Pdrop vs Nv with pv = 0.3, 0.5, 0.7

0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 110

–4

10–3

10–2

pv

Pdro

p

simulationtheory

Fig. 5 Pdrop vs pv with Nv = 60

10 15 20 25 30 35 40 45 50 55 6010

–5

10–4

10–3

Nv

Pdro

p

Nd=0

Nd=10

Fig. 6 Pdrop vs Nv with pv = pd = 0.5, Ld = 100, λd = 0.2 msg/s

Figure 7 shows the performance of the proposed sys-

tem, in terms of message delay for data packets, when

Nv = 40, λd = 0.2msg/s, Nd = 2 and pv = pd = 0.5. Two

DTs are able to share the network in the scaled version of the

proposed system with 40 VTs, while a good overall message

delay is achieved. A good approximation for the average bit

rate available for the active DT is Rb = Rb · νNch

, where ν is

the mean number of available channels per slot. The above

figure also shows the observed confidence interval of simu-

lation tests.

An accurate comparison of the proposed system results

with other existing works is made difficult by the fact that

performance are extremely influenced by several parameters,

like the adopted voice codec, the channel coding scheme, and

channel transmission capability. From our literature analy-

sis, a lack of systems capable of allocating large numbers of

voice services with quality guarantees is outlined. In [5] and

[18], where results are shown for the case of an ideal channel,

Springer

Wireless Netw (2008) 14: 17–28 27

50 100 150 200 250 300 350 4000.05

0.1

0.15

0.2

0.25

0.3

0.35

0.4

0.45

Tm

es

(s)

Ld

theoreticalsimulation

Fig. 7 Tmes vs Ld with λd = 0.2 msg/s, Nv = 40, Nd = 2, pv = pd =0.5

like in the present paper, a relatively smaller number of voice

terminals, with respect to available channel bandwidth, can

be supported in the network, even compared to the scaled ver-

sion of the system proposed in this paper. Moreover, Pdrop per-

formance are extremely influenced by the presence of other

traffic types [18]. Differently from [19], the proposed pro-

tocol allows only collisions among transmission requests,

and not among information messages. This lets the actual

throughput remain substantially close to the offered network

load. Moreover, when the system is sufficiently far from be-

ing congested, the data message delay is not heavily influ-

enced by the number of data terminals, since this parameter

is theoretically not affected by collisions.

6. Conclusions

In this paper we have discussed the performance evaluation

of a reservation-based MAC scheme, which effectively ex-

ploits the OFDM sub-channel structure and permits to meet

the service quality constraints of real-time applications such

as voice communications. A suitable analytical approach was

proposed and the obtained analytical predictions were com-

pared with simulation results. Even in case of a scaled-down

version of the system, it has been demonstrated that a large

number of voice terminals can be supported by the WLAN

under consideration, while data services still have good per-

formance in terms of delay message.

References

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Metropolitan area Networks-specific Requirements-part 11: Wire-less Lan Medium Access Control (MAC) and Physical Layer (PHY)Specifications (1999).

2. IEEE Std 802.11a-1999 Supplement to IEEE standard for in-formation technology—telecommunications and information ex-change between systems—local and metropolitan area networks—specific requirements. Part 11: wireless LAN Medium Access Con-trol (MAC) and Physical Layer (PHY) (1999).

3. ETSI, TR 101 031, Broadband Radio Access Networks (BRAN);High Performance Radio Local Area Network (HIPERLAN) Type2; Requirements and architectures for wireless broadband access(1999).

4. IEEE Standard for Information Technology—Telecommunicationsand Information Exchange Between Systems–LAN/MAN SpecificRequirements—Part 11 Wireless Medium Access Control (MAC)and Physical Layer (PHY) specifications: Medium Access Control(MAC) Quality of Service (QoS) Enhancements (2005).

5. A. Banchs, X. Perez, M. Radimirsch and H.J. Stuttgen, “Servicedifferentiation extensions for elastic and real-time traffic in 802.11wireless LAN,” High Performance Switching and Routing, 2001IEEE Workshop on, 2001 (2001) pp. 245–249.

6. S. Mangold, S.Choi, P. May, O. Klein, G. Hiertz and L. Stibor,“IEEE 802.11e wireless LAN for quality of service,” Proc. ofEW2002, Florence, Italy (Feb. 2002) pp. 32–39.

7. B. Mc Farland, G. Chesson, C. Temme and T. Meng, “The 5-upprotocol for unified multiservice wireless networks,” IEEE Comm.Mag., Vol. 39, No. 11 (Nov. 2001) pp. 74–80.

8. E. Lawrey, “Multiuser OFDM,” Signal Processing and its Applica-tions, 1999 5th IEEE International Symposium on (Aug. 1999).

9. R. van Nee, “New high-rate wireless LAN standards,” IEEE Comm.Mag. (Dec. 1999) pp. 82–88.

10. H. Xie and D.J. Goodman, “Prma system performance with vari-able number of calls,” Technical Report, WINLAB-TR-36 (June1992).

11. ITU-T Recommendation G.711 (11/88): “Pulse code modulation(PCM) of voice frequencies.”

12. K. Svanbro, “Lower layer guidelines for robust RTP/UDP/IPheader compression,” draft-ietf-rohc-rtp-lower-layer-guidelines-03.txt (Dec. 2001).

13. IEEE Standard for Local and Metropolitan Area Networks. Part.16: Air Interface for Fixed Broadband Wireless Access Sytems—IEEE Std 802.16-2004 (2004).

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16. S. Nanda, D.J. Goodman and U. Timor, “Performance of PRMA: apacket voice protocol for cellular systems,” IEEE Trans.on Comm.(Aug. 1991) pp. 584–598.

17. L. Kleinrock, Queuing Systems, Prentice Hall (1992).18. T. Suzuki and S. Tasaka, “Performance evaluation of priority-based

multimedia transmission with the PCF in an IEEE 802.11 standardwireless LAN,” Personal, Indoor and Mobile Radio Communica-tions, 2001 12th IEEE International Symposium on, Vol. 2 (Sept.2001) pp. 70–77.

19. M. Natkaniec and A.R. Pach, “An analysis of the backoff mecha-nism used in IEEE 802.11 networks,” Computers and Communi-cations, 2000. Proceedings. ISCC 2000. Fifth IEEE Symposium on(2000) pp. 444–449.

Springer

28 Wireless Netw (2008) 14: 17–28

Romano Fantacci, (M’87,SM’91, F’05) bornin Pistoia, Italy, graduated from the Engineer-ing School of the Università di Firenze, Flo-rence, Italy, with a degree in electronics in1982. He received his Ph.D. degree in telecom-munications in 1987. After joining the Dipar-timento di Elettronica e Telecomunicazioni asan assistant professor, he was appointed asso-ciate professor in 1991 and full professor in1999. His current research interests are digital

communications, computer communications, queuing theory, satellitecommunication systems, wireless broadband communication networks,ad-hoc and sensor networks. He has been involved in several Euro-pean Space Agency (ESA) and INTELSAT advanced research projects.He is the author of numerous articles published in prestigious com-munication science journals. He guest edited special issues in IEEEJournals and magazines and served as symposium chair of severalIEEE conferences, including VTC, ICC and Globecom. Professor Fan-tacci received the IEE IERE Benefactor premium in 1990 and IEEECOMSOC Award Distinguished Contributions to Satellite Communica-tions in 2002. He is currently serving as Editor for TelecommunicationSystems, IEEE Trans. Commun. and IEEE Transactions on WirelessCommunications.

Gianluca Vannuccini born in Florence, Italy,graduated in Electronics Engineering in 1999.He received his Ph.D. degree in Telematics andInformation Society in 2003 from the Elec-tronics and Telecommunications Departmentof the University of Florence. During 2002 hewas for six months in IBM Zurich ResearchLaboratory, Zurich, Switzerland, working ina research project on IEEE 802.11e perfor-mance evaluation. During the Ph.D. course,

he has published on several IEEE conferences and served as reviewerfor several journals on the telecommunications and telematics researcharea. He has been IEEE student member since 1999 and IEEE membersince 2004. He is now with the IT department of the Florence localgovernment organization, where he works as a program manager ondata quality and integration and e-government IT projects.

Gabriele Vestri was born in Florence (Italy)in November 1976. He received his degreein telecommunications from the Università diFirenze, Florence, Italy, in March 2002.He has been research scientist for CSO Oph-thalmic (Florence) since September 2002. Hiscurrent research interests include ophthalmicinstruments, contact lens design, image pro-cessing, the effects of optical aberrations andof retinal architecture of the eye on visual

performance.

Springer