Management of Low and Variable Bit Rate ATM … › files › 3214883 › Voo...CONTENTS MANAGEMENT...

Western Australian Telecommunications Research Institute

and

The University of Western Australia

Management of Low and Variable Bit Rate

ATM Adaptation Layer Type 2 Traffic

Charles Voo

This thesis is presented for the

Degree of Doctor of Philosophy

of

The University of Western Australia

School of Electrical, Electronic and Computer Engineering

October 2003

MANAGEMENT OF LOW AND VARIABLE BIT RATE ATM ADAPTATION LAYER TYPE 2 TRAFFIC

i

Acknowledgments

I wish to express my sincere thanks to my supervisors, Associate Professor John

Siliquini and Professor Zigmantas Budrikis for their guidance and assistance throughout

my studies towards the Ph.D. degree.

I would also like to acknowledge the support of the Western Australia

Telecommunications Research Institute (WATRI) throughout my Ph.D.

In addition, I would like to acknowledge the financial support provided to me for my

Ph.D. studies by an Australian Postgraduate Award and an Australian

Telecommunications Cooperative Research Centre Award.

More thanks are due to my parents for their encouragement and support throughout my

studies. Special thanks to Tarith Devadason for his valuable comments.

Finally, I would like to thank Jin for her continual love and support.

ABSTRACT


ii

Abstract

Asynchronous Transfer Mode (ATM) Adaptation Layer Type 2 (AAL2) has been developed to carry low and variable bit rate traffic. It provides high bandwidth efficiency with low packing delay by allowing voice traffic from different AAL2 channels to be multiplexed onto a single ATM virtual channel connection. Examples of where AAL2 are used include the Code Division Multiple Access and the Third Generation mobile telephony networks. The main objective of this thesis is to study traditional and novel AAL2 multiplexing methods and to characterise their performance when carrying low and variable bit rate (VBR) voice traffic.

This work develops a comprehensive QoS framework which is used as a basis to study the performance of the AAL2 multiplexer system. In this QoS framework the effects of packet delay, delay variation, subjective voice quality and bandwidth utilisation are all used to determine the overall performance of the end-to-end system for the support of real time voice communications.

Extensions to existing AAL2 voice multiplexers are proposed and characterised. In the case where different types of voice applications are presented to the AAL2 multiplexer, it was observed that increased efficiency gains are possible when a priority queuing scheme is introduced into the traditional AAL2 multiplexer system.

Studies of the voice traffic characteristics and their effects on the performance of the AAL2 multiplexer are also investigated. It is shown that particular source behaviours can have deleterious effect on the performance of the AAL2 multiplexer. Methods of isolating these voice sources are examined and the performance of the AAL2 multiplexer re-evaluated to show the beneficial effects of a particular source isolation technique.

The extent to which statistical multiplexing is possible for real time variable VBR sources is theoretically examined. These calculations highlight the difficulties in multiplexing VBR real time traffic while maintaining guaranteed delay bounds for these sources. Based on these calculations, multiplexing schemes that incorporate data transfers within the real time traffic transfer are proposed as alternatives for utilising unused bandwidth caused by the VBR nature of the voice traffic.

CONTENTS


iii

Table of Contents

Chapter 1 Introduction 1

1.1 Broadband Integrated Services Digital Network (B-ISDN) 1

1.2 Asynchronous Transfer Mode (ATM) 2

1.2.1 ATM Cell Structure 3

1.2.2 ATM Switching Principles 5

1.2.3 ATM Transfer Capabilities 6

1.2.3.1 Deterministic Bit Rate (DBR) 7

1.2.3.1.1 Using the DBR ATC to Transport CBR Traffic 8

1.2.3.1.2 Using the DBR ATC to Transport VBR Traffic 9

1.2.3.2 Statistical Bit Rate (SBR) 11

1.3 ATM Adaptation Layer (AAL) 13

1.3.1 AAL1 14

1.3.2 Original AAL2 16

1.3.3 AAL3/4 16

1.3.4 AAL5 19

CONTENTS


iv

1.3.5 AAL Summary 20

1.3.6 The New AAL2 22

1.3.6.1 Comparisons 25

1.3.6.1.1 AAL1 and AAL2 25

1.3.6.1.2 AAL5 and AAL2 30

1.3.6.2 AAL2 Work 33

1.4 Objectives 33

1.5 Thesis Contents 34

Chapter 2 Establishing Real Time Connections in ATM Networks Using AAL2 36

2.1 AAL2 Network Structure 37

2.2 Real Time Communications 38

2.2.1 Delay Constancy 39

2.2.1.1 Source 40

2.2.1.2 Network 44

2.2.1.3 Destination 46

2.2.1.4 Continuity of Data Flow 47

2.2.1.4.1 Spacer buffer 47

2.2.1.4.2 Play-out buffer overflow 50

CONTENTS


v

2.2.1.4.3 Sink starvation 51

2.2.1.5 Establishing the real time connection 54

2.2.2 Quality of Service (QoS) Framework 57

Chapter 3 AAL2 Multiplexer Model 59

3.1. AAL2 Multiplexer System Model 60

3.1.1. Voice Sources 60

3.1.2. AAL2 Multiplexer Model 66

3.2 Simulation in OPNET 69

Chapter 4 Priority Queuing 76

4.1 Delay Budget 76

4.2 Scenario Example 79

4.2.1 General AAL2 Multiplexer Performance 81

4.2.2 Performance of Prioritised AAL2 Multiplexer 83

Chapter 5 Source Sensitivity 89

5.1 Simulation Example 89

5.2 Performance Sensitivity of AAL2 Multiplexer 91

5.3 Usage Parameter Control (UPC) 93

5.3.1 Token Bucket Policer 94

CONTENTS


vi

5.3.2 Selection of Token Bucket Parameters 97

5.4 Conclusion 103

Chapter 6 Alternative Multiplexing Method 104

6.1 Simulation Example 104

6.2 Worst Case Behaviour of the Token Bucket Parameters 107

6.3 Future Work - Integrated Multiplexing Scheme 111

6.4 Conclusion 114

Chapter 7 Conclusion 115

References 118

Appendix A Implementation of the DBR and SBR Cell Dispatch Processes 127

Appendix B An Analysis Establishing the Equivalence between DBR and SBR ATCs 139

Appendix C UPC – Software Implementation 148

ACRONYMS


vii

Acronyms

3G Third Generation

AAL1 ATM Adaptation Layer Type 1



ATM Asynchronous Transfer Mode

CDMA Code Division Multiple Access

CDV Cell Delay Variation

CID Channel Identifier

CPS Common Part Sublayer

DBR Deterministic Bit Rate

E-ADPCM Embedded-Adaptive Differential Pulse Code Modulation

ETSI European Telecommunications Standards Institute

FCFS First Come First Serve

HEC Header Error Control

IP Internet Protocol

ITU-T International Telecommunications Union – Telecommunications

ACRONYMS


viii

LI Length Indicator

MBS Maximum Burst Size

MOS Mean Opinion Score

OSF Offset Field

P Parity

PCM Pulse Code Modulation

PDU Protocol Data Unit

PPR Peak Packet Rate

PSTN Public Switched Telephone Network

QoS Quality of Service

SAP Service Access Point

SBR Sustainable Bit Rate

SMG Statistical Multiplexing Gain

SN Sequence Number

SNR Signal to Noise Ratio

SPR Sustainable Packet Rate

SSCS Service Specific Convergence Sublayer

UPC Usage Parameter Control

UTRAN Universal Mobile Telecommunications System Terrestrial Radio

ACRONYMS


ix

Access Network

UUI User to User Indication

VAD Voice Activity Factor

VBR Variable Bit Rate

AUTHOR‘S PUBLICATIONS LIST


x

Author’s Publications List

[1]. C. Voo, J. F. Siliquini, and G. Mercankosk, “Service differentiation of variable

bit rate voice in AAL2 multiplexers”, in Proceedings of IEEE Region 10

International Conference on Electrical and Electronic Technology

(TENCON’01), vol. 2, pp. 631-635, 2001.

[2]. C. Voo, J. F. Siliquini and G. Mercankosk, “Performance of AAL Type 2 Voice

Multiplexers”, Proceedings of the 9th IEEE International Conference on

Telecommunications (ICT’02), vol. 1, pp. 1045-1049, June, 2002.

[3]. C. Voo and J. F. Siliquini, “Performance Comparison of Multiplexing Methods

for Voice over ATM using AAL2”, in Proceedings of the 9th IEEE

International Conference on Telecommunications (ICT’02), vol. 1, pp. 593-597,

June, 2002.

[4]. C. Voo, “A Review of the New Adaptation Layer Type 2”, Inter-University

Postgraduate Electrical Engineering Symposium (IUPEES’99), pp. 17-18, July

1999.

[5]. C. Voo, “Performance of Statistical Multiplexed Voice over ATM using AAL2

and Deterministic Bit Dropping”, Inter-University Postgraduate Electrical

Engineering Symposium (IUPEES’00), pp. 71-74, July 2000.

[6]. J. F. Siliquini, G. Mercankosk, S. Ivandich, C. Voo, Z. L. Budrikis, and A.

Cantoni, “On Statistical Multiplexing Gain for Variable Bit Rate Voice

Sources”, in Proceedings of the 8th IEEE International Conference on


CHAPTER 1 INTRODUCTION


1

Chapter 1

Introduction

The Broadband Integrated Services Digital Network (B-ISDN) has been defined for

integrating the transport of different traffic types onto a single network infrastructure.

The underlying technology chosen for the B-ISDN is the Asynchronous Transfer Mode

(ATM). In this chapter, some important characteristics of the ATM such as cell

structure, switching principles and transfer capabilities are described. Also, descriptions

for the roles of ATM adaptation layers (AAL) are given. Comparisons between existing

AAL protocols will highlight the need for the recently defined AAL type 2. Finally, the

AAL2 is described and the thesis’ aims listed.

1.1 Broadband Integrated Services Digital Network (B-ISDN)

Historically, land based telecommunications systems started with the integration of data

communications equipment into the existing Public Switched Telephone Network

(PSTN). The connection of remote computer equipment across different countries was

financially economical as the network infrastructure was already installed

internationally. However, the PSTN was designed for low bandwidth voice traffic and

therefore not suitable for transmitting data, especially when speed requirements

increased. As a result, these networks were later modified with the addition of high

speed cabling but still requiring Modem (Modulator Demodulators using analogue

signalling) connections. These networks had become known as Public Switched Data

Networks (PSDN).



2

As the demands on bandwidth increased, the need for a true digital media became

apparent and this led to the development of the Integrated Service Digital Network

(ISDN). The ISDN supports telephony and a wide range of data applications such as

teletext and facsimile in the same network at connection speeds of 64kbit/s. This rate

was chosen because, at that time, it was the standard rate for digitised voice.

With the development of ISDN, the possibility of new services such as video

conferencing, video telephony and other multimedia type applications were being

investigated. However the introduction of these new services into the ISDN was

hampered by the limitation that ISDN can only support applications compatible with the

64kbit/s switched digital connections. Therefore, the Broadband Integrated Service

Digital Network (B-ISDN) was developed and standardised by the International

Telecommunications Union – Telecommunications (ITU-T) to support multimedia

services with different bandwidths and delay requirements.

At the time B-ISDN was developed, there were two existing technologies that could be

used to support the B-ISDN. These were the Synchronous Digital Hierarchy (SDH) and

the Asynchronous Transfer Mode (ATM) technologies. ATM was chosen as a candidate

to be the transport mechanism for B-ISDN due to its simplicity and its capability to

support a variety of both delay and loss sensitive traffic types.

1.2 Asynchronous Transfer Mode (ATM)

ATM is a cell based networking and switching technology which can support a variety

of both delay and loss sensitive traffic. As a cell based transmission technology, ATM

packs data from various sources attached to the B-ISDN into a standard ATM cell

format and the network transports these cells across the network. This uniform cell

structure standardises the processing of cells and simplifies the integration of network

components. ATM has been standardised by both the International Telecommunications

Union (ITU) [1] [2] and the ATM Forum for use in the planned public network of the

future.



3

ITU was established on May 17, 1865. It was formed to allow the interconnection of

telegraph networks between countries. Now there are three general ITU sectors:

Telecommunications (ITU-T), Radiocommunications (ITU-R) and Development (ITU-

D). The main objective of the ITU is to define international standards that are to be

adopted by all countries. Because the standardisation body is large, much time is

required before a standard is adopted. Therefore the ATM Forum was established in

October 1991 and was meant to accelerate the development of ATM products and

services through a rapid convergence of interoperability specifications. In addition, it

was supposed to promote industry cooperation and market awareness. For the work

presented in this thesis, descriptions of any ATM terms will be based on the ITU-T

standards as ITU is internationally recognised.

1.2.1 ATM Cell Structure

With ATM, information for all services is conveyed and switched in fixed sized

segments called cells. Each cell is 53 octets in length, consisting of a 5 octet header and

a 48 octet payload field. There are two different types of ATM cells as shown in Figure

1. ATM cells transferred between a terminal and the local ATM switch follow the User-

Network Interface (UNI) cell structure, which includes a Generic Flow Control (GFC)

field. ATM cells transferred within the network between ATM switches follow the

Network-Network Interface (NNI) cell structure, which has an expanded Virtual Path

field in place of the GFC field.



4

UNI Octet NNI

GFC VPI

VPI VCI

VCI

VCI PTI CLP

HEC

1ST octet of payload

2nd octet of payload

48 octet of payload

7

6

5

4

3

2

1

VPI

VCI PTI

48 octet of payload

2nd octet of payload

1ST octet of payload

HEC

CLP

VCI

VCI

VPI

53

0 77 bit bit 0

Figure 1: UNI and NNI ATM cell structure.

The following gives a description of the various ATM fields:

• Generic Flow Control (GFC): Consists of 4 bits and is optionally used to

regulate the entry of cells into the ATM network.

• Virtual Path Identifier (VPI): Consists of 12 bits in the NNI and 8 bits in the

UNI, and is used for the identification and routing of cells.

• Virtual Channel Identifier (VCI): Consists of 16 bits and is also used for the

identification and routing of cells.

• Payload Type Identifier (PTI): Consists of 3 bits and is used to identify the type

of information contained in the ATM cell.

• Cell Loss Priority (CLP): Consists of 1 bit and is used to identify the priority of

the cell with regards to its discard potential. Cells with CLP = ‘1’ are considered



5

low priority and discarded first when a network element experiences congestion.

• Header Error Control (HEC): Consists of 8 bits and is used for error checking on

the first 4 octets of the header.

1.2.2 ATM Switching Principles

The VPI and VCI fields within each ATM cell are used to identify and switch cells

across the ATM network. The size of the fields are minimised by only providing

switching information between the current and next switching elements and not an end-

to-end global address as in the Internet Protocol (IP) address. The VPI/VCI fields within

the cells are updated as they are passed from switch to switch.

VPI_1 VPI_2

VPI_3

VCI_1 VCI_1

VCI_2

VCI_1

ATM Switch

Figure 2: VPI/VCI Translation within an ATM switch.

Figure 2 shows the relationship between VPIs and VCIs and how they are translated

when processed in an ATM switch. From Figure 2, there are two connections both

having a VPI of 1. Within a single virtual path, there can be theoretically up to 216

virtual connections. In the above example, there are only 2 VCI labelled 1 and 2.



6

1.2.3 ATM Transfer Capabilities

Applications with different service requirements are supported in ATM by different

transfer schemes [2] [3] [4]. Delay-sensitive applications, such as telephony and video

require a timing relation between source and destination and there are transfer

capabilities which exist for these. These real-time applications require a limit on the

variation in the end-to-end delay to allow the communications to be real-time in nature.

It also allows for the relevant transmit and receive buffers to be dimensioned. On the

other hand, for applications such as data transfers, which are non-delay sensitive but

loss-sensitive, no timing relation is required between source and destination. There are

also transfer capabilities to support this type of communications.

ATM supports four different transfer capabilities, these being the Deterministic Bit Rate

(DBR), Statistical Bit Rate (SBR), Available Bit Rate (ABR) and Unspecified Bit Rate

(UBR) transfer capabilities.

In Table 1 we summarise the characteristics of each of the ATM transfer capabilities in

terms of their suitability for the transport of real time and non-real time traffic. Since

this thesis is primarily concerned with the transfer of real time traffic, only DBR and

SBR ATM transfer capabilities will be described in more detail.

SBR Service

Characteristics

DBR

Type 1 Type 2 and 3

ABR UBR

Bandwidth Guarantee Yes Yes Yes Optional No

Real time traffic Yes Yes No No No

Bursty data traffic No No Yes Yes Yes

Table 1: Summary of the ATM service characteristics.



7

The difference in SBR types (excluding SBR Type 1 (SBR1)) is in the handling of cells

based on the value of the CLP field in the ATM header (described in Section 1.2.3). For

cells with CLP=1, selective cell discard applies to both SBR Type 2 (SBR2) and SBR3

Type 3 (SBR3). However SBR3 also allow cells with CLP=1 to be tagged. For voice

and other real time applications, only DBR and SBR Type 1 (SBR1) transfer

capabilities are suitable.

1.2.3.1 Deterministic Bit Rate (DBR)

Deterministic Bit Rate (DBR) ATM transfer capability (ATC) is used for the transport

of real time traffic with guaranteed bandwidth for delay sensitive applications. Quality

of Service commitments provided by DBR ATC are guaranteed for each connection.

The traffic characteristic of the DBR ATC is modelled by a single traffic descriptor,

namely the Peak Cell Rate (PCRDBR). Under ITU standardisation, the DBR ATC can be

used to support both constant bit rate (CBR) traffic and variable bit rate (VBR) traffic.

The general case of transporting ATM cells using DBR ATC is shown in Figure 3.

Incoming cells

tk+1Spacer

DBR

dspacer,k , 'k k spacer k p kt d Dζ τ= + + +

tk tk-1 ζk+1 ζk ζk-1

timetime

ATMτκ’

Tk,k+1

Source Network Destination

Figure 3: General model showing distribution of cells through an ATM network

using the DBR ATM transfer capability.

Source

Referring to Figure 3, tk denotes the time at which the last bit of the kth cell is presented

to the spacer. The interarrival time between the kth and the k+1th cell is defined as Tk,k+1.



8

Spacer

The function of the spacer is to limit the presentation of cells into the ATM network to a

rate less than or equal to the peak cell rate of the DBR connection i.e. PCRDBR. Let

dspacer,k denote the waiting time experienced by the kth cell in the spacer buffer. This

value is assumed to be statistically bounded by τspacer (i.e. 0 ≤ dspacer,k < τspacer).

Network

There are two components of delay associated with the transport of cells through the

ATM network. The first component delay is the fixed propagation delay and is the time

it would take a cell to traverse the ATM network if the cell experienced no queuing

delay along its path. It also includes the packet transmission and processing delay

within the switches in the network. This component of delay is denoted as Dp. The

second component of delay is the queuing delay. Let τk’ denote the total queuing delay

experience by the kth cell along its path. The value of τk’ is dependent on the amount of

jitter or queuing delay experienced by the cells travelling through the ATM network.

According to DBR traffic contract, the value of τk’ is statistically bounded by the cell

delay variation tolerance, τCDV [5].

Destination

At the destination, the last bit of the kth cell arrives at time ζk defined as

, 'k k spacer k p kt d Dζ τ= + + + (1.1)

1.2.3.1.1 Using the DBR ATC to Transport CBR Traffic

Constant bit rate (CBR) sources are sources that produce traffic at fixed rates and are

characterised by a single traffic parameter, their peak cell rate. Figure 4 shows the

transport of these CBR cells through the ATM network using the DBR ATC.



9

, 11

k kDBR

T PCR+ ≥

Incoming cells

Spacer

dspacer,k

DBR

'k k p kt Dζ τ= + +PCRDBR

tk+1 tk tk-1

time

ζk+1 ζk ζk-1

time

ATMτκ’

Network DestinationSource

Figure 4: Transport of CBR cells through an ATM network using the DBR ATM

transfer capability.

Referring to Figure 4, to conform to the DBR traffic contract the CBR input traffic rate

must be less than or equal to the peak cell rate of the DBR connection (i.e. Tk,k+1 ≥

1/PCRDBR). In this case, the waiting time for each cell in the spacer, dspacer,k is 0. Note

that the CBR input traffic rates cannot be greater than the PCR of the DBR connection

because incoming cells will be discarded when the spacer inevitably overflows. Using

(1.1) and τCDV as the bound for τk’, we can write for ζk:

k p k k p CDt D t D Vζ τ+ ≤ ≤ + + (1.2)

1.2.3.1.2 Using the DBR ATC to Transport VBR Traffic

The DBR ATC can also be used to transport Variable Bit Rate (VBR) sources. VBR

sources are characterised by variable inter-cell arrival times. We assume that the VBR

sources generate packets that conform to the Generic Cell Rate Algorithm (GCRA) [6]

that has three parameters; Peak Cell Rate (PCRsource), Sustainable Cell Rate (SCRsource)

and Intrinsic Burst Tolerance (τIBTsource). In this case, the GCRA is a policer that

discards non-conforming cells before sending them to the spacer buffer to prevent the

possibility of spacer buffer overflow. Therefore the sustainable cell rate of the source

(i.e. SCRsource) must be smaller or equal to the peak cell rate of the DBR ATC (i.e.

PCRDBR). The characteristics of the cell transport for a VBR source using the DBR ATC

is shown in Figure 5.



10

ATMτκ’

Incoming cells

Spacer

dspacer,k

DBR

, 'k k spacer k p kt d Dζ τ= + + +PCRDBR

Cells conform to GCRA(PCRsource, SCRsource, τIBTsource)

tk+1 tk tk-1

time

ζk+1 ζk ζk-1

time

Source Network Destination

Figure 5: Distribution of VBR packets through an ATM network with DBR

transfer capabilities.

Referring to Figure 5, the VBR input traffic rate can be greater than the PCR of the

DBR connection (i.e. Tk,k+1 ≤ 1/PCRDBR) when the burst of cells conform to GCRA

(PCRsource, SCRsource and τIBTsource). The maximum burst size of the traffic source is

given by

11 1

IBTsource

source source

MBS

SCR PCR

τ= +

⎛ ⎞−⎜ ⎟⎝ ⎠

(1.3)

For VBR traffic, dspacer,k is statistically bounded by the intrinsic burst tolerance, τIBTsource

(i.e. 0 ≤ dspacer,k < τIBTsource). The maximum network delay, τk’ experienced by each cell

is again statistically bounded by τCDV. At the destination, each cell arrives at time ζk that

has bounds in the range given by (1.4).

, k p k k spacer k p CDVt D t d Dζ τ+ ≤ ≤ + + + (1.4)



11

1.2.3.2 Statistical Bit Rate (SBR)

Statistical Bit Rate (SBR Type 1) ATM transfer capability (ATC) has 3 traffic

parameters associated with it, namely the PCRSBR, SCRSBR and τIBT SBR. The input rate

into the network is limited to the PCRSBR, which with the specified τIBT SBR provides a

limit on the volume of traffic that may be input above the SCRSBR. Therefore, VBR

sources must first be shaped according to the GCRA with its PCRSBR, SCRSBR and

τIBT SBR parameters before they are carried by the SBR ATC. The SBR service

guarantees that the actual transfer will be a rate at least equal to the SCRSBR. However, at

times the VBR traffic can be serviced at a higher rate than the SCRSBR. But that is not

guaranteed. The model for VBR cell transport through an ATM network with SBR

ATM transfer capability is shown in Figure 6. Note that no spacer is required in this

case.

ATMτκ’

Incoming cells

Policer

SBR 'k k p kt Dζ τ= + +PCRSBR, SCRSBR, τIBT SBRtime

tk+1 tk tk-1time

ζk+1 ζk ζk-1

Source Network DestinationCells conform to GCRA(PCRsource, SCRsource, τIBTsource)

Figure 6: Distribution of VBR packets through an ATM network with SBR

transfer capabilities.

Referring to Figure 6, incoming cells pass through a policer according to the GCRA

(PCRSBR, SCRSBR and τIBT SBR). Conforming cells are passed to the ATM network with

no delay incurred by the policer. Note that any cells found non-conforming by the

GCRA (PCRSBR, SCRSBR and τIBT SBR) are unconditionally discarded. Within the ATM

network, each cell will be subject to a variable queuing delay. The composition of the

queuing delay includes not only the constant delay Dp and the delay due to phase

coincidences with other traffic (i.e. τCDV) but also a possible smoothing delay whenever

the service rate within the network is less than the rate at which the cells enter the ATM

network (i.e bounded by the intrinsic burst tolerance, τIBT SBR). The bounds on these



12

various delay components for a number of scheduling disciplines used in the ATM

switches are summarised in Table 2.

Scheduling disciplines Variable delay bounds

Parekh-Gallager [7]

PGPSi MUX IBT SBR

SBR

KD SCRτ τ ⎛ ⎞< + + ⎜ ⎟⎝ ⎠

Golestani [8] 1 ...SCFQ

i MUX IBT SBR KSBR

KD NSCR N Kτ τ δ⎛ ⎞< + + + + + −⎜ ⎟⎝ ⎠

δ δ

Stiliadis-Varma [9] ( ) ( )( )1 ... KLR

i MUX IBT SBR i iD τ τ θ θ< + + + +

Goyal-Vin-Cheng [10] ( ) 1 ...SFQi MUX IBT SBR KD N N Kτ τ δ δ< + + + + − δ

Table 2: Variable transfer delay bounds for various scheduling disciplines.

Referring to Table 2, the bracketed term in each of the bounds corresponds to the delay

due to phase coincidences (i.e. τCDV) with other traffic across the network. The number

of hops a cell goes through is denoted by K and δ denotes one cell transmission time in

seconds. Nj denotes the number of connections competing for access at hop j. It can be

seen in Table 2 that for any of the scheduling disciplines, the smoothing delay is

bounded by τIBT SBR.

Therefore, when using the SBR ATC for transporting VBR traffic, cells arrive at the

destination at time ζk that has bounds in the range given by

' k p k k pt D t D kζ τ+ ≤ < + + (1.5)

Where the term τ’k is given by

' k CDV IBT SBτ τ τ= + R (1.6)



13

1.3 ATM Adaptation Layer (AAL)

Transfer capabilities are supported by ATM Adaptation Layers (AAL). AALs provide a

mapping protocol between higher layers and the ATM layer. In terms of hierarchy

within the B-ISDN Protocol Reference Model, it rests on top of the ATM layer as

shown in Figure 7.

ATM Adaptation

Layer (AAL) (3)

Convergence Sublayer (CS)

Segmentation & Reassembly

Sublayer (SAR)

ATM Layer (2)

Physical Layer (1)

Higher Layers (4+)

Figure 7: B-ISDN Protocol Reference Model.

Referring to Figure 7, the lowest layer is the Physical layer. This layer is concerned with

the transmission of data as well as other low level functions such as bit timing,

transmission of frames and error checking. The second layer is the ATM layer, which

provides VPI/VCI translation as described in Section 1.2, cell header creation/retrieval

and Generic Flow Control.

Currently, there are four AAL protocols specified to cover the transfer capabilities

described in Section 1.2.3. The AAL chosen for use is one that best suits the

characteristics of the higher layer applications. The characteristics of each AAL is

summarised in Table 3.



14

ATM Adaptation Layer Type Characteristics

AAL1 AAL2 (old)

AAL3/4 AAL5

Timing Relationship

Yes Not Required

Bit Rate Constant Variable

Mode Connection Oriented

Connectionless / Connection Oriented

Connection Oriented

Table 3: ATM Adaptation Layer Characteristics.

The AAL layer consists of two sublayers known as the Convergence Sublayer (CS) and

the Segmentation and Reassembly sublayer (SAR). The functionality of the CS is to

receive/send packets from/to higher layers. For each AAL, the packet structure is

different and as such, each AAL supports a different type of CS packet. The

functionality of the SAR sublayer is to segment packets into sizes equivalent to the

length of the ATM cell payload before sending these onto the ATM layer.

1.3.1 AAL1

AAL1 has been designed to provide a DBR, connection oriented service wherein the

timing relationship between the source and the destination is required. This timing

relationship is obtained by the use of the Source Clock Frequency Recovery [11] [12].

The use of AAL1 is suitable for delay sensitive applications such as telephony.

The Segmentation and Reassembly Protocol Data Unit (SAR-PDU) packet structure of

AAL1 is shown in Figure 8. The size of the packet structure is exactly 48 octets in

length (i.e. corresponding to the length of an ATM cell payload). A SAR-PDU is

formed by prepending a Segmentation and Reassembly Service Data Unit (SAR-SDU)

with an octet header when it leaves for the ATM layer. Note that the header is extended

to 2 octets when the CSI field is set to 1 and the SCF field is of even count (i.e. 0, 2, 4



15

or 6). The offset field in the additional octet is used as a pointer to indicate the end of

the payload and the parity bit is used to provide protection over the offset field. For this

SAR-PDU, the maximum payload is 46 octets.

1 3 3 1 1 Bits7

CSI SCF CRC SAR-SDU

1 Octet

SAR-SDU=47 Octets when CSI=0

PAR PAR Offset Field

Present when CSI=1 and

SCF=0, 2, 4, or 6

Figure 8: AAL1 SAR-PDU Packet Structure.

The following gives a description of the various AAL1 fields:

• Convergence Sublayer Indication (CSI): Consists of 1 bit. This indicates the

presence of the convergence sublayer function. Some examples of CS functions

include the handling of SAR-PDU for partially filled SAR-PDU payloads, the

handling of cell delay variation for delivery of AAL-SDUs to an AAL user at a

constant bit rate, and timing information transfers.

• Sequence Count Field (SCF): Consists of 3 bits. This is provided by the CS layer

and is used for the detection of lost or mis-inserted SAR-SDUs at the receiver.

• Cyclic Redundancy Checksum (CRC): Consists of 3 bits. This is used for bit

error detection and correction over the SAR-PDU header.

• Parity (PAR): Consists of 1 bit. This is set such that the 1 octet SAR-PDU

header has even parity and is used to protect the CRC.

The one octet header is checked and removed by the AAL1 SAR sublayer on reception

and the payload sent to the higher layer from the CS layer.



16

As previously mentioned, partially filled SAR-PDU payloads can be handled by the CS

sublayer using the Structure Data Transfer (SDT) method where an additional octet in

the SAR-PDU header (i.e. extended to 2 octets) is used as a pointer to indicate the end

of the payload [13]. This cell format can only be used when the sequence count value

(i.e. SCF) in the SAR-PDU header is 0, 2, 4, or 6. For delay-sensitive applications

where the SAR-SDUs are always less than 47 octets, [13] defines a CS procedure for

partially filling the payload of a SAR-PDU. This method (known as the partial fill

procedure) requires the receiving AAL CS to know when the payload contains

overhead, the number of overhead octets and the position of these octets in the payload.

Using this method, the number and position of AAL user information octets and CS

generated dummy value octets in the remaining payload octets can be determined.

However, [13] does not specify how the receiver will be able to distinguish AAL user

information from padding (i.e. dummy octets) using information obtained from the

AAL header. This has yet to be implemented for the AAL1.

1.3.2 Original AAL2

Referring to Table 3, the original AAL2 was intended to provide real time services that

have variable bit rates. However, due to the standard having many undefined properties,

it is no longer under development. Note that this AAL bears no structural relation to the

new AAL2 that will be described later in Section 1.3.6.

1.3.3 AAL3/4

AAL3 and AAL4 merged to become AAL3/4 and provide both connection and

connectionless data service for variable bit rate (VBR) traffic. However, the AAL itself

does not perform all functions required by a connectionless service, since functions such

as routing and network addressing are performed at the network layer. There are two

modes in which the AAL can operate; Stream Mode and Message Mode. If the

preservation of message boundaries is required, then Message Mode must be used.



17

The size of an AAL3/4 packet can be as large as 65,535 octets. The large packet size

introduces transmission latencies, thus making it unsuitable for real time traffic. Figure

9 shows both the AAL3/4 CS and SAR PDU packet structures. Note that the SAR-SDU

can be smaller than 44 octets.

1 1 2 Octets =<65535 0-3 1 1 2

CPI BETag BA AAL-SDU Pad AL BETag

ST SN MID 44 Octets of CS-PDU LI

SAR-PDU

CS-PDU

6 10 bits 4 102

CRC

Length

Figure 9: AAL3/4 CS and SAR packet structures.

The following gives a description of the various AAL3/4 CS-PDU fields:

• Common Part Indicator (CPI): Consist of 1 octet. This is used to interpret

subsequent fields for the CS functions in the CS-PDU header. Examples include

identifying related AAL layer management messages such as performance and

fault monitoring, and the transfer of Operation and Management (OAM)

messages.

• Begin End Tag (BETag): Consist of 1 octet. This is a sequence number used for

checking packet synchronisation. Made redundant for connectionless services.

Note that it is also repeated in the tail.

• Buffer Allocation (BA): Consists of 2 octets. This allows the receiving CS to

allocate the appropriate amount of memory resources for incoming data. When

AAL3/4 operates in message mode, the BA value is encoded equal to the CS-



18

PDU payload length. In streaming mode, the BA value is encoded equal to or

greater than the CS-PDU payload length.

• AAL-Service Data Unit (AAL-SDU): Allows up to 65,535 octets of data to pass

between the higher layer application and the CS.

• Pad: Consists of up to 3 octets. Size of padding required must be such that the

total packet length is a multiple of 4.

• Alignment (AL): Consists of 1 octet. This is similar to Pad but ensuring that the

trailer is 4 octets long.

• Length: Consists of 2 octets. This indicates the length of the CS-PDU payload

field in octets and is also used by the receiver to detect loss or gain of

information.

For transmission, the CS-PDUs are segmented by the SAR sublayer into 44 octet blocks

and prepended 2 octet headers and appended 2 octet trailers. Note that the SAR-SDUs

can be smaller than 44 octets. The resultant 48 octet SAR-PDUs are sent to the ATM

layer where they are encapsulated into ATM cells through the prepending of 5 octet

headers.

The following gives a description of the various AAL3/4 SAR-PDU fields:

• Segment Type (ST): This identifies a SAR-PDU as containing a Beginning of

Message (BOM), a Continuation of Message (COM), an End of Message (EOM)

or a Single Segment Message (SSM).

• Sequence Number (SN): Consists of 4 bits and can be used for the detection of

missing SAR-SDUs.

• Multiplexing Identifier (MID): Consists of 10 bits used to identify which CS-

PDU the received SAR-PDU relates to. This allows AAL3/4 to multiplex data



19

from different AAL3/4 connections.

• Length Indicator (LI): Consists of 6 bits to indicate the length of SAR-SDU

information in the SAR-PDU payload.

• Cyclic Redundancy Checksum (CRC): Consists of 10 bits used for bit error

detection on transfer of SAR-PDU.

1.3.4 AAL5

Due to the merging of AAL3 and AAL4, AAL3/4 has large overheads. AAL5 was then

developed to replace AAL3/4. It provides similar connectionless services support as

AAL3/4 but with less transmission overheads and better error detection. However,

unlike AAL3/4, AAL5 does not support multiplexing of different AAL5 connections

onto a single VCC.

The CS for AAL5 is further divided into two parts, the Common Part Convergence

Sublayer (CPCS) and the Service Specific Convergence Sublayer (SSCS) [11]. The

SSCS provides signalling functions as required by the higher layers, and may

sometimes be null. Similar to AAL3/4, AAL5 has two modes of operations; Streaming

Mode and Message Mode. This has been discussed in Section 1.3.3. The AAL5 CS-

PDU packet structure is shown in Figure 10.

octets 0-47 1 1 2 4

=< 65535 octets payload Pad UU CPI Length CRC

CS-PDU

Figure 10: AAL5 CS-PDU packet structure.



20

The following gives a description of the various AAL5 CS-PDU fields:

• Pad: Padding of 0 - 47 octets added to ensure that the CS-PDU length is an

integral number of 48 octet segments. This helps to simplify SAR segmentation

process and receiver packet field decoding.

• User to User identifier (UU): Consists of 1 octet. It enables the AAL user layers

to identify the associated Service Access Point (SAP).

• Common Part Indicator (CPI): Consist of 1 octet. This is not used but is included

to ensure trailer without padding is 8 octets in length.

• Length: Consists of 2 octets. It indicates the length of the CS-PDU payload field

and is also used by the receiver to detect loss or gain of information.

• Cyclic Redundancy Checksum (CRC): Consists of 4 octets used for error

detection on transfer of CS-PDU.

The SAR-PDUs are created by segmenting CS-PDUs into 48 octet blocks. There are no

prepended header or appended trailer fields. AAL5 uses the ATM User to User (AUU)

parameter in the ATM cell PTI field to indicate the existence of the end of a CS-PDU in

a SAR-PDU payload. Note a SAR-PDU where the value of AUU is ‘1’ indicates the

end of a CS-PDU; the value of ‘0’ indicates the beginning or continuation of a CS-PDU.

1.3.5 AAL Summary

From the above descriptions of AALs, it is found that each different AAL is used for a

different service class.

AAL1 supports DBR real time, connection-oriented services and is therefore suitable

for the transportation of voice band signals (e.g. One 64kbit/s A-law or µ-law coded

G711 signal), video and high quality audio traffic.



21

AAL3/4 is suitable for services that utilise long packets. This is because the longer the

packet, the smaller the percentage of the transmission overheads. However, having a

long packet may result in a received packet being discarded when only a bit is in error.

AAL3/4 supports multiplexing of different AAL3/4 data streams unto a virtual channel

through the use of the MID field.

AAL5 is the improved version of AAL3/4. It provides the same functions but with less

transmission overheads and better error detection than the AAL3/4. Hence, AAL5 has

been adopted as the standard for the ATM signalling protocol. It does not support

multiplexing of packets on a virtual channel.

In summary, any real time services such as multimedia require the use of AAL1 and

non real time services require the use of either AAL3/4 or AAL5.

Since the mid 1990s, there has been an increasing need for the support of low bit rate

time sensitive traffic. Although AAL1 is able to support this type of traffic, it is at the

expense of inefficient use of bandwidth since the cell rate must be high to maintain real

performance but for low bit rate traffic, only a small portion of the ATM cell payload is

utilised. Here it has been assumed that AAL1 has the capability to support partially

filled payloads using the partial fill method described in Section 1.3.1.

An example of such an application is the transfer of voice data using AAL1. Audio

sources are usually sampled at 8kHz and quantified into an 8-bit word, therefore

requiring a 64kbits/s channel. Using an ATM cell to transfer just one voice data word is

very inefficient use of available bandwidth (i.e. 1 octet out of a possible 47-octet

payload space). Packing more voice data words into the ATM cell payload can increase

the bandwidth efficiency, but can only be done at the expense of delaying transmission

of some data until the cell is partly or completely full.

If ATM cells can only be transmitted when the payload is completely filled, then the

first cell in the payload would have to wait for a total of 47 octets. Byte arrivals occur

once every 125µs. Once the first byte in an ATM cell payload has been received, it



22

would have to wait for another 46 octets. In terms of time to accumulate 47 octets, this

corresponds to a total time of 47×125µs=5. 875ms at an effective rate of 64kbits/s.

At this point, it should be noted that most voice sources compress their voice data prior

to transmission. An operational characteristic of compressors is to buffer the voice data

until sufficient data is obtained before transmitting them. This introduces additional

delay especially when the system is already waiting for enough data to fill an entire

ATM cell payload. So in the above example, if voice was compressed to an effective

data rate of 8kbits/s, the total packetisation delay would increase to 8×5.875ms=48ms.

This is unacceptable because for a voice connection, the delay budget that consists of

delay components such as queuing delay, propagation delay, network delay and

equalisation delay is tight (i.e. around 100ms one way). Hence with such a large

packetisation delay, it is difficult to meet this delay budget, given the existence of the

other delay components (refer to Section 4.1 for more details).

The solution then is to provide an AAL that can multiplex data packets from multiple

sources into a single ATM cell payload. This requires a small but variable length

packet. It should also allow for processing of concurrent packet arrivals from different

higher layer applications since a number of packets of different sizes can be multiplexed

into the same ATM cell. This can greatly increase bandwidth efficiency and reduce

transmission latencies due to the reduced time in filling the ATM payloads. The

recently defined AAL2 supports such requirements.

1.3.6 The New AAL2

The new AAL2 standardised by the ITU-T in November 2000 [14] [15] [16] and the

ATM Forum in [17] was developed specifically to support the transfer of low and

variable bit rate traffic across the ATM network. It does this efficiently by supporting

the multiplexing of AAL2 packets from different higher layer applications into the same

ATM cell for transmission. The AAL2 SAR-PDU packet structure is shown in Figure

11.



23

8 6 5 5 bits

CID LI UUI HEC

3 octets =< 44 octets

Payload

Figure 11: AAL2 SAR-PDU packet structure.

Descriptions of the AAL2 packet fields are as follows:

• Channel Identifier (CID): Denotes the corresponding Convergence Part Sublayer

(CPS) connection using 8 bits. Note that there are 8 reserved values.

• Length Indicator (LI): Indicates the number of valid octets in the payload within

the range of 1 to 45 using 6 bits in the header.

• User to User Indication (UUI): Consists of 5 bits and is used by the Service

Specific Convergence Sublayer (SSCS) for traffic management such as

Operation Administration and Maintenance (OAM), long packet segmentation

and carrying audio encoding format profiles for code points 0 to 15. Code points

between 16 and 22 are reserved.

• Header Error Control (HEC): Consists of 5 bits and is used for error detection in

the packet header.

The AAL2 packet is variable in size and can be much smaller than an ATM cell payload

of 48 octets. Due to its variable size, many AAL2 packets from different connections in

the application layers can be packed into the same ATM cell for transfer. As mentioned

previously, this reduces the network latency and improves bandwidth efficiency at high

rates.

The AAL2 Common Part Sublayer (CPS) provides the multiplexing capability for

AAL2 packets when multiple packets are packed into a single ATM cell payload. To do

this, the first byte in the ATM cell payload is used to carry the Start Field byte, shown

in Figure 12. The first six bits of the Start Field define the location of the first new



24

AAL2 packet. From there, the AAL2 packet lengths are used to maintain packet

alignment within the 47-octet payload. The remaining bits in the Start Field are used for

cell sequence checking and bad parity detection within the Start Field.

6 1 1 bits

Hdr Offset SEQ PAR Payload = AAL2 SAR-PDUs Start Field 47 octets

ATM Cell Payload

Figure 12: AAL2 CPS-PDU packet structure.

The multiplexing process for AAL2 is illustrated in Figure 13.

1 2 3

Common Part Sublayer (CPS)

1 2

Sources

1 a b 3Service Specific Convergence Sublayer (SSCS)

AAL2 SAR-PDU

AAL2 CPS-PDU

3-byte header

Start Field header

Source Packets

3

ATM Cells

ATM header

48 Octet

47 Octet

Figure 13: Voice transportation using AAL2.



25

Referring to Figure 13, source packets (labelled 1 to 3) are collected by the SSCS and

segmented into 44 octet segments. Note that source packets can be of variable sizes.

Segmentation of packets is illustrated by Packet 2 whereby it is segmented into 2

smaller segments, labelled ‘a’ (full 44 octets) and ‘b’ (remaining octets). Each segment

is AAL2 encapsulated (i.e. prepended a 3 byte header) to form AAL2 SAR-PDU

packets. These AAL2 SAR-PDUs are then segmented by the CPS to form AAL2 CPS

PDUs each with a Start Field header. Note that the CPS-PDU payload is fully utilised.

1.3.6.1 Comparisons

The performance of the AAL2 is now compared with the performance of AAL1 and

AAL5 in terms of packing efficiency and the transmission delay.

1.3.6.1.1 AAL1 and AAL2

As mentioned previously, even though AAL1 is suitable for real time traffic, it is not

able to efficiently carry low and variable bit rate real time traffic such as voice due to

bandwidth inefficiency. The example of Figure 13 used to show the transportation of

source packets via AAL2 will be used to illustrate the transportation of the same packets

using AAL1. This is shown in Figure 14.



26

1 2

1 2 Source Packets

1 octet header

AAL1 Layer AAL1 packets

3 Padding

VCC1 VCC2 VCC2 VCC3

Sources (Applications)

3

ATM Layer ATM packets

5 octet header

Figure 14: Voice transportation using AAL1.

Referring to Figure 14, each packet (whole or segmented) is encapsulated to form

AAL1 packets. Note that partial fill procedure described in [13] is used. Due to the

fixed length AAL1 payload, padding is required to fill the remaining spaces, thus

resulting in poor bandwidth utilisation. The problem of low bandwidth utilisation is

solved for the case of the AAL2, in which packets from different connections are

multiplexed onto the same ATM payload, thereby utilizing the whole payload space. A

case scenario shown in Figure 15 is used to compare the performance of AAL1 and

AAL2.



27

8 kb/s Codec

Figure 15: Case scenario for comparisons between AAL1 and AAL2.

Referring to Figure 15, a number of 8kbits/s codecs are used. The size of the packets is

dependent on the allowed packetisation delay. For AAL1, source packets are

encapsulated into a fixed size payload of 47 octets. The remaining unused payload not

filled by the SAR-SDU is padded and an AAL1 header is prepended. Each AAL1

packet is then further encapsulated into an ATM cell. Also packets from each source

require a different Virtual Channel Connection (VCC).

For the AAL2, source packets are encapsulated and prepended 3 octets header to form

an AAL2 packet. These are then further segmented into fixed size AAL2 CPS PDU

packets before being ATM encapsulated. AAL2 packets from different sources are

multiplexed onto an AAL2 CPS-PDU packet. The remaining AAL2 packet that cannot

AAL2 Packet

1 n 1 n

Source Packet

AAL1 Packet

ATM Packet

1 octet header

5 octet header

Padding

VCC1 VCCn VCC1 VCC1

3 octet header

Source Packet

AAL2 CPS PDU

Packetisation Delay

AAL1 AAL2

ATM Packet

1 octet header



28

be filled into the first CPS-PDU payload is filled into the second CPS-PDU payload,

thus resulting in little to no bandwidth wastage (see Section 1.3.6). Note that packets

from different sources share a common VCC.

The bandwidth efficiency for the AAL1 can be calculated by dividing the number of

useful octets over the total size (i.e. the total number of ATM cells used) as given by

(1.7). The size of an ATM cell is 53 octets (i.e. 5 octet header and 48 octet payload).

The size of an AAL1 PDU is 48 octets, which includes the one octet header.

No. of useful octetsEfficiency total No. of ATM packets × 53

= (1.7)

The maximum efficiency is obtained when number of useful octets equal 47 resulting in

an efficiency of 88.7%.

The bandwidth efficiency for the AAL2 requires obtaining the total number of AAL2

packets that result from the source packet, and then dividing the useful octets over the

total size (i.e. the total number of ATM cells used). AAL2 packets are variable in length

with a maximum of 47 octets (i.e. 3-octet header and 44-octet payload). The AAL2

CPS-PDU is 48 octets in length including the one octet header. Using (1.8), the total

number of AAL2 packets that is required by the source packet can be determined. The

bandwidth efficiency is calculated using the resultant value through the application of

(1.7).

( )Size of source packet in bitsNo. of AAL2 packets

44 8 bits⎡ ⎤

= ⎢ ⎥×⎢ ⎥

(1.8)

Where , 1x R n n x x n x∀ ∈ ∈ = ⇔ ≤ < +⎡ ⎤⎢ ⎥¢



29

The performance comparison between AAL1 and AAL2 is shown in Figure 16. Values

in this figure is obtained using (1.7) and (1.8). The horizontal scale is obtained with

reference to an 8kbit/sec sampling rate.

0 10 20 30 40 500

20

40

60

80

1008kb/sec voice codec

Typical codec generation times

AAL Type 1 AAL Type 2

Band

wid

th E

ffic

ienc

y (%

)

Packetisation Delay (ms)

Figure 16: Bandwidth efficiency vs codec delay.

Referring to Figure 16, it can be observed that AAL2 performs better than AAL1 over a

wide packetisation range. The performance of AAL1 is comparable to AAL2 only when

the packet length reaches a certain size. The bandwidth efficiency for the AAL1 is

directly proportional to the size of the packet whereas for the AAL2, it reaches seven-

eighths of its maximum bandwidth efficiency (i.e. 70%) for a packetisation delay of

10ms.



30

1.3.6.1.2 AAL5 and AAL2

Although this thesis is primarily concerned with the transport of real time data, it is

interesting to examine the use of AAL2 for the transport of non-real time data. As was

mentioned earlier, AAL5 is most suitable for carrying time insensitive traffic. The

transportation of data traffic using AAL5 is shown in Figure 17.

1 Source Packets

AAL5 packet length a multiple of 48 octets

Padding AAL5 trailer

AAL5 Layer AAL5 Packets

ATM Layer ATM Packets

Source

1

Figure 17: Data transportation using AAL5.

Referring to Figure 17, the data packet is first padded to ensure that its overall length is

a multiple of 48 octets, given that a trailer of 8 octets is appended. In this example, the

disadvantages of using the AAL5 are:

• Padding the length of AAL5 packets so that they map directly into 48 octet

ATM cell payload results in bandwidth wastage, especially if packets are short.

• For short packets, the 8-octet trailer takes up a relatively large percentage of the

overall payload capacity.

• Each separate data channel requires its own Virtual Channel Connection (VCC).



31

As the AAL5 packet size increase, the trailer and padding required become a lesser

percentage of the overall payload being transferred, thus resulting in better bandwidth

efficiency.

The use of AAL2 for transporting data packets has the advantage that for short packet

lengths, it introduces less overhead than AAL5, thus better bandwidth efficiency. For

larger packet lengths, the bandwidth efficiencies of AAL5 and AAL2 are comparable.

A case scenario shown in Figure 18 is used to compare the performance of AAL5 and

AAL2 for the transport of non-real time data. The performance criterion considered is

bandwidth efficiency.

1 1 Packet Source

AAL5 AAL2

Figure 18: Case scenario for comparisons between AAL5 and AAL2.

Referring to Figure 18, for the AAL5, each source packet is padded and appended with

an AAL5 trailer to ensure that the AAL5 packet length is a multiple of 48 octets. The

AAL5 packet is then segmented into 48 octets before being ATM encapsulated. Note

Source Packet

AAL5 Packet

ATM Packet

5 octet header Padding

VCC1

VCC1

AAL2 Packet

3 octet header

Source Packet

AAL2 CPS PDU

AAL5 trailer

1 octet header

ATM Packet

Contains another partial AAL2 packet



32

that a different VCC is used for packets from each different source.

The performance comparison between the AAL5 and AAL2 for the case scenario of

Figure 18 is shown in Figure 19. Figure 19 shows the corresponding bandwidth

efficiency for a range of data packet lengths.

0 200 400 600 800 10000

10

20

30

40

50

60

70

80

90

100

AAL2 AAL5

Band

wid

th E

ffici

ency

(%)

Data Packet Length (octets)

Figure 19: Bandwidth efficiency vs packet length.

The values of Figure 19 can be calculated or obtained from [18]. Before calculating the

bandwidth efficiency of AAL5, the total size of the AAL5 packet (including 8 octet

trailer and padding) and the number of ATM cells required to transmit the whole data

packet have to be determined. Once these values are obtained, the bandwidth efficiency

can be calculated by dividing the number of useful octets over the total size (i.e. the

total number of ATM cells required).

Referring to Figure 19, the bandwidth efficiency of AAL2 is much better than AAL5 for

packet lengths smaller than 100 octets. For larger packet lengths, the bandwidth



33

efficiency of both AAL2 and AAL5 are comparable. Note that the saw-tooth effect of

AAL5 is due to the padding required to ensure the whole AAL5 packet is a multiple of

48 octets.

1.3.6.2 AAL2 Work

Work published incorporating the use of this relatively new AAL2 are as follows: in

[18] [19], the authors compare the performance of AAL2 against existing AALs. From

these papers, AAL2 outperforms AAL1 when carrying low bit rate traffic by being able

to multiplex across many virtual channels when packing an AAL2 payload seen in

Figure 16. However performance of AAL2 for large and bursty data is only comparable

to AAL5 for large packets seen in Figure 19. This is largely due to the many headers

required for these AAL2 packets resulting in low bandwidth efficiency.

The use of AAL2 has found its place in many applications. These applications include

Code Division Multiple Access (CDMA) in [20] [21] [22], 3G in

[23] [24] [25] [26] [27] [28], Universal Mobile Telecommunications System Terrestrial

Radio Access Network (UTRAN) in [29] [30] [31] [32] and Internet protocol (IP) [33].

Much of the work on AAL2 is based on traffic management issues. These include work

on congestion control involving AAL2 [34] [35] [36] [37], Quality of Service (QoS)

requirements [38], packing efficiency [39] [40] [41], trunking efficiency [42], bandwidth

management issues [43], performance issues of AAL2 [44] [45], AAL2 implementation

[46] and various other issues [47]. This thesis focuses on the traffic management issues

incorporating AAL2 for low and variable bit rate traffic.

1.4 Objectives

The main objective of this thesis is to examine the performance of AAL2 multiplexers

and in particular the traffic management issues associated with the multiplexer. The

specific objectives of this thesis are:



34

• To develop a QoS framework (i.e. a set of QoS parameters) for describing the

performance of AAL2 voice multiplexers.

• To examine the performance limitation of the single-queued AAL2 multiplexer for

the transport of Variable Bit Rate (VBR) voice.

• To propose and examine an extension to the single-queued AAL2 multiplexer for

achieving a higher trunking efficiency.

• To examine performance sensitivity of the AAL2 multiplexer with respect to input

voice traffic.

• To examine source policing of VBR voice sources to improve the delay

performance of the AAL2 multiplexer in the presence of misbehaving voice sources.

• To examine statistical multiplexing and the extent to which it is possible for real

time VBR voice.

• To propose and examine an alternative multiplexing method to statistical

multiplexing for utilising available unused bandwidth.

1.5 Thesis Contents

The content of this thesis provides a step by step approach to investigating the traffic

management issues associated with AAL2 multiplexer design and is organised as

follows:

Chapter 2 outlines an ATM network used to study the transport of real time

communications using AAL2. This includes a study on the requirements for

establishing a real time voice connection. Once a connection has been established, the

resulting voice quality is then examined and described using a set of Quality of Service

(QoS) parameters. Also in this chapter, the use of DBR and SBR ATM Transfer

Capabilities (ATC) for the transportation of voice traffic in AAL2 connections are



35

described and compared. Based on the comparisons, a suitable ATC is then chosen and

adopted in the chapters that follow.

Chapter 3 describes the general AAL2 multiplexer system that consists of input sources

and the AAL2 multiplexer itself. The input sources are modelled as voice codecs with

silence suppression and are characterised in terms of bandwidth and a subjective quality

score. The AAL2 multiplexer performance is described via the statistical multiplexing

gain that can be achieved while maintaining QoS requirements. The study of the AAL2

multiplexer and its performance is investigated using the network simulation tool called

OPNET.

Chapter 4 extends the single-queued AAL2 model to two prioritised queues where

priority levels are assigned based on their input traffics’ multiplexing delay

requirements. A high priority is assigned to traffic that have tighter multiplexing delay

requirements and is determined via comparing their associated delay budgets. The delay

performance of the prioritised AAL2 multiplexer is compared to the single-queued

AAL2 multiplexer in terms of the statistical multiplexing gain that they achieve.

Chapter 5 examines the effects of source sensitivity on the performance of the AAL2

multiplexer. The desired behaviour of a source can be obtained by enforcing its traffic

through some form of usage parameter control. A common UPC and the selection of its

parameters are described in this chapter.

Chapter 6 examines again the delay performance of the AAL2 multiplexer by

considering the worst case traffic behaviour that can pass the UPC for statistically

multiplexed sources. This is achieved by obtaining the number of sources that can be

accommodated using the derived worst case delay within the multiplexer. Based on

these results alternative multiplexing schemes are proposed to utilise unused bandwidth

caused by the VBR nature of the voice traffic.

Finally, conclusions are drawn in Chapter 7.

CHAPTER 2 ESTABLISHING REAL TIME CONNECTIONS IN ATM NETWORKS USING AAL2


36

Chapter 2

Establishing Real Time Connections in ATM Networks Using AAL2

In Chapter 1, AAL2 was defined and described as the most suitable adaptation layer for

the transportation of low and variable bit rate traffic such as voice. With its multiplexing

capability, it is able to achieve high packing efficiency at low transmission delays.

Hence, AAL2 has found its place in many applications. An example where AAL2 has

been used is in third generation (3G) mobile technology.

In this chapter, a general network incorporating the use of AAL2 is described. Using

this network, we proceed to examine the requirements for establishing a real time voice

connection. Once this has been established, it is important then to quantify the resulting

voice quality. This is described using a set of Quality of Service (QoS) parameters.

As previously mentioned in Chapter 1, real time traffic can be transported using either a

DBR or SBR transfer capability. In this chapter, an analysis is presented to illustrate

that the use of either ATM transfer capability will give equal performance under certain

conditions.



37

2.1 AAL2 Network Structure

ATM Cloud

Workstation

Fax

Telephone

ATM Switch

ATM Switch

ATM Switch

ATM Switch

AAL2

DeM

ultiplexer

AAL2

Mul

tiple

xer

Workstation

Fax

Telephone

Virtual ChannelConnection

Figure 20: Transport of data traffic in an ATM network via AAL2.

Referring to Figure 20, AAL2 supports a number of diverse traffic sources. The network

structure shown in Figure 20 consists of sources, an AAL2 multiplexer (at the source),

ATM network and an AAL2 demultiplexer (at the destination). Packets sent to the

AAL2 multiplexer are segmented and encapsulated. These AAL2 packets then enter the

ATM network using a particular ATM transfer capability (see Section 1.2.3). Within the

ATM network, cells are switched from one switch to another via the VPI/VCI fields in

the ATM cell header until they arrive at the destination. Reassembly of packets is

performed in the AAL2 demultiplexer and the reassembled packets are sent to the

appropriate destination (see Section 1.3.6).

As packets transverse through the network of Figure 20, they experience both fixed and

variable delays. An end-to-end packet delay is defined as the time interval from when

the packet arrives at the source AAL2 multiplexer to it being played out at the receiver.

In general, this value is not constant.



38

2.2 Real Time Communications

For real time communications, a voice signal at the transmitter must be reproduced at

the receiver with the same timing. This means that voice packets should be played out

by the receiver using the same timing structure as created by the source transmitter. At

the destination, packets that have arrived are decoded into voice samples which are then

played out.

The quality of real time communications is affected by packet loss when packets are

discarded due to buffer overflows or when at the sink, a packet is unavailable at its play-

out interval. The case where a packet is unavailable to be played out at the sink can arise

as follows. Given that the end-to-end packet delay as described previously is variable,

when the receiver plays out the voice samples decoded from the first packet

immediately upon its arrival, it may not be possible for the receiver to play the next set

of voice samples (decoded from the next packet) after the designated time interval due

to the packet arriving too late. Noise samples are played out in place of these voice

samples and the late packet is discarded resulting in poor quality. This effect is known

as sink starvation. A method of overcoming this problem is to delay playing out the first

packet such that after the designated time interval, the receiver is able to play out the

subsequent set of voice samples from the next packet. Introducing such a fixed delay is

known as equalisation. Generally, the required equalisation delay is dependent on the

variable delays experienced by the packets.

The quality of real time communications is also affected by packet loss when the buffer

overflows (spacer/policer buffer at the source and playout buffers at the receivers). This

happens when packet arrivals into the buffer are greater than the packet departures. A

method of eliminating this effect is to dimension the buffers such that overflows will

not occur.



39

In the following sub-section, the requirements for establishing a real time connection

are described. The equalisation process and the dimensioning of buffers are also

described.

2.2.1 Delay Constancy

As outlined in the previous section, to provide quality of service to real time connection,

it is necessary to achieve constant end-to-end delay across the network. Constancy of

end-to-end delay can be achieved by maintaining continuity of data flow for the

duration of the call. Such continuity breaks down either when buffers overflow or

underflow. To minimise these problems, dimensioning of these buffers can avoid the

possibility of buffer overflow while equalisation can prevent buffer underflow (or sink

starvation). Note that it is important not to use a large equalisation delay as it will cause

large delay with only slight improvement in the possibility of buffer underflow.

In Figure 20, there is a delay associated with each element in the model, which is

illustrated in Figure 21. The end-to-end packet delay is the sum of these delay

components experienced by the packet as it travels from the transmitter to the receiver.

In the following sub-sections, the end-to-end packet delay is analysed. From this, the

equalisation delay as well as the required buffer sizes (such that overflows will not

occur) can be obtained [38].



40

Figure 21: Delay components of the AAL2 network.

2.2.1.1 Source

Referring to Figure 21, at the source, each voice transmitter (L=1, 2,…, n) generates a

packet at fixed time intervals of length TL and at a rate given by

1L

L

RT

= (2.1)

Note for voice sources (i.e. On/Off sources), (2.1) is only applicable during the talk

intervals.

Packets generated by sources are delivered through the Service Access Points (SAP) to

the AAL2 SSCS without any delay variation. Let tL,k represent the time at which the Lth

transmitter presents the last bit of the kth packet to the SSCS layer and is expressed as

, ,0L k L Lt t kT= + (2.2)

where tL,0 denotes the time at which the Lth transmitter presents the last bit of the first

AAL2 Multiplexer (SSCS and CPS)

1 2 n

Voice Transmitters

t1,k t2,k tn,k

S(t)

1 2 n

Voice Receivers Delay

Constancy required

AAL2 De-Multiplexer

Play-out buffer

ξ1,k ξ2,k ξn,k

B1,ξ B2,ξ Bn,ξ

Spacer/

ATM Network Dp + τi’

Policer

γi

γ1,k γ2,k γn,kBζ

R(t)



41

When a packet arrives at the SSCS from connection L, it is passed onto the CPS layer

packet to the SSCS.

without any delay variation and is encapsulated into an AAL2 packet. Multiplexing of

individual AAL2 connections is performed in the CPS where AAL2 packets are packed

into CPS PDU cells and queued into the spacer/policer buffer before being sent to the

ATM network (see Section 1.3.6). Due to the multiplexing effects experienced in the

CPS, the characteristics of this aggregate cell stream will be of a Variable Bit Rate

(VBR) nature. Thus it can be considered that the rate of cells being generated by the

CPS corresponds to some time series denoted by S(t). These cells are placed into a

buffer of size Bζ for spacing and/or policing. Let ti represent the time at which the CPS

places the last bit of the ith cell in the spacer buffer or policer. Therefore S(t) is shown as

( ) ( )0

ii

S t t tδ∞

=

= −∑ (2.3)

where δ(…) denotes a Dirac delta function. The time series S(t) can be characterised by

a peak rate (PCRAAL2 Aggregate), a mean rate and some measure of its burstiness. It is

assumed that S(t) satisfies some burstiness constraint

( ) ( )1b

sa

S d b a Rτ τ σ< + + −∫ (2.4)

for all b ≥ a, where the constant Rs represent an upper bound for the long term average

rate of the traffic generated. The term Rs is known as the Sustainable Cell Rate (SCR)

that is associated with the aggregate AAL2 traffic stream. The term σ represents the

limit on the amount of traffic that can be generated in excess of 1 + (b - a)Rs during any

interval [a, b]. The term σ is generally in the form

2 IBT AAL Aggregate sRσ τ= (2.5)

where τIBT AAL2 Aggregate is the Intrinsic Burst Tolerance of the aggregate AAL2 traffic



42

The aggregate traffic stream departing the CPS can be transported using either the DBR

For the DBR ATC:

With the use of the DBR transfer capability, the traffic generated at the CPS is required

p

with respect to the Sustainable Cell Rate, Rs. 1 This also represents how early a cell can

arrive before it is theoretically due with respect to the SCR.

or the SBR transfer capabilities.

to be spaced to the Peak Cell Rate (PCR) of the DBR connection which is related to the

bandwidth allocated and is guaranteed for the aggregate traffic stream across the

network. Note the peak cell rate of the DBR connection denoted by Rp must be chosen

greater than or equal to the long term average rate Rs (i.e. Rp ≥ Rs). For the first cell that

arrives into the spacer, it is immediately served. After this, traffic spacing is performed

for subsequent cells and the time between each emission from the spacer buffer is at

least the specified minimum Tp where the term Tp is obtained from the inverse of the

PCR of the DBR connection (i.e. Tp = 1/Rp). Let ζi represent the time at which the

server removes the first bit of the ith cell from the spacer buffer. This is

( )0 0

1max ,i i i

t

t T

ζ

ζ ζ −

=

= +

(2.6)

The transmission delay incurred by the cell is equal to the cell size (i.e. 53 octets)

Let R(t) denote the output time series at which cells are presented to the network by the

divided by the link rate of the ATM connection. This delay is insignificant when the

link rate is large.

1 When individual VBR sources are multiplexed to form an aggregate VBR source, the parameters Rs and

τIBT AAL2 Aggregate of the aggregate traffic can be difficult to determine unless the actual traffic characteristics

are measured. Here we have assumed that it is possible to determine these parameters.



43

spacer buffer. R(t) can be written as

( ) ( )0

ii

R t tδ ζ∞

=

= −∑ (2.7)

Let si denote the time spent in the spacer buffer by the ith cell. Therefore ζi can be

i

obtained as the time the ith cell waits to be served upon arrival into the buffer and is

shown as

i it sζ = + (2.8)

Equation (2.8) can be extended to AAL2 packets within those cells by

( ) ( ),, ,i L k iL k t s L kζ = + (2.9)

where ζi(L,k) represents the time at which the first bit of the cell is removed from the

The maximum time spent in the spacer buffer is bounded by

spacer buffer that contains the kth packet of connection L and si(L,k) represents the time

spent in the spacer buffer of the cell which contains the kth packet of the Lth connection.

In the case where a AAL2 packet has been split over two cells, si(L,k) refers to the time

spent in the spacer buffer by the trailing cell. A description of the DBR cell dispatch

process is given in Appendix A.

0 i spacers D≤ ≤ (2.10)

where

spacerp

BD Rζ= (2.11)



44

For the SBR ATC:

When carrying the aggregate traffic stream in a SBR connection, traffic generated by

the CPS must be policed according to the appropriate peak cell rate, sustainable cell rate

and the intrinsic burst tolerance of the SBR traffic contract (see Section 1.2.3.2). Here, it

is assumed that the aggregate AAL2 traffic stream parameters (i.e PCRAAL2 Aggregate, Rs

and τIBT AAL2 Aggregate) match the SBR traffic contract parameters (i.e PCRSBR, SCRSBR, τIBT

SBR) and is shown by

2

2

SBR AAL Aggregate

SBR s

IBT SBR IBT AAL Aggregate

PCR PCR

SCR Rτ τ

=

=

=

(2.12)

Under these conditions, conforming cells are presented immediately to the ATM

network without the need for spacing (i.e. si = 0). Non-conforming cells are

unconditionally discarded. In the case where traffic generated by the CPS is presented to

the network without policing, the network can only guarantee serving the aggregate

connection at a rate equal to the agreed sustainable cell rate. Therefore during network

congestion, the SBR traffic may still be shaped within the network in any of the

switches along its path (else non-conforming cells may be dropped). A description of

the SBR cell dispatch process is given in Appendix A.

2.2.1.2 Network

Once a cell enters the ATM network, it hops from one switch to the next along its path

within the network before reaching the destination. The cell experiences a propagation

delay, Dp (see Section 1.2.3.1) and a queuing delay. Let τi’ denote the total queuing

delay experienced by the ith cell along its path.



45

V

For the DBR ATC:

In the case when using the DBR ATC, the queuing delay experienced by the cell in the

network is statistically bounded by the Cell Delay Variation Tolerance (τCDV) (see

Section 1.2.3.1) and is given by

'0 i CDτ τ≤ < (2.13)

The value of τCDV can be obtained by some (1 - α) quantile of the queuing delay. Note

that this value refers to the difference between the best and worst case expectation of the

cell transfer delay. The best case is equal to Dp and the worst case is equal to Dp + τCDV,

a value likely to be exceeded with a probability less than α.

Let γi denote the time at which the last bit of the ith cell is presented to the CPS

demultiplexer. Therefore, the ith cell reaches the destination at time

'i i pD iγ ζ τ= + + (2.14)

At the destination, the CPS demultiplexes the individual AAL2 packets with a fixed

interval from the time of arrival of the cell and places them into play-out buffers. Let γL,k

denote the time at which the last bit of the kth packet is inserted into play-out buffer L,

(2.14) is written as

( ) ( )', , ,L k i p iL k D L kγ ζ τ= + + (2.15)

Note in (2.15), the term ζi(L,k) is given in (2.9) and τi’(L,k) represents the total network

queuing delay of the cell that contains the kth packet of connection L. In the case where

a packet is split over two cells, the network delay τi’(L,k) refers to the total queuing

delay of the trailing cell.



46

R

For the SBR ATC:

Equations (2.14) and (2.15) equally apply when using the SBR ATC. However, in the

case when using the SBR ATC, the queuing delay τi’ is statistically bounded by not

only the Cell Delay Variation Tolerance (τCDV) but also a smoothing delay (i.e Intrinsic

Burst Tolerance (τIBT SBR)) (see Section 1.2.3.2) and is given by

' 0 i CDV IBT SBτ τ τ≤ < + (2.16)

2.2.1.3 Destination

As cells arrive at the destination, the timing structure for the sequence γi may not

necessarily be the same as when they left the source (i.e. sequence ti). This is due to the

variable delays experienced by these cells either at the spacer buffer or within the

network, or both. After demultiplexing by the destination CPS, source packets are

placed into play-out buffers of size BL,ξ for each connection L. The functionality of the

play-out buffers is to allow incoming packets to be queued for play-out when the

receiver is busy playing out the previous voice packet. This prevents packet loss. Also,

the timing structure of the sequence γL,k is not necessarily the same as the sequence tL,k.

For each connection L, the sink consumes one packet in every TL time units which is the

same rate at which the source generates packets. A common strategy for recovering the

initial timing structure is to delay the consumption of the first packet by a fixed amount

of time denoted by DL,ξ and set the size of the play-out buffer to be BL, ξ. For each

connection L, the continuity of data flow is maintained by consuming packets from the

play-out buffers using the same timing structure generated by the sequence tL,k. Let ξL,k

represent the time at which the first bit of the kth packet of connection L is read from the

play-out buffer. Therefore, ξL,k is expressed as

( ) ( )', , , 0 ,0 ,L k L k i p i Lt s L D L D ξξ τ= + + + + (2.17)

where si(L,0) and τi’(L,0) correspond respectively to the time spent in the spacer buffer



47

and the network delay of the first packet of connection L. For the SBR ATC, cells are

passed to the network without entering the spacer/policer buffer (i.e. si(L,0) = 0).

The fundamental requirement in any real time connection is that end-to-end packet

delay for all packets must be equal to the end-to-end delay of the first packet in

connection L i.e. ξL,k - tL,k = ξL,0 - tL,0. Equally said, the end-to-end delay (i.e. the

difference of ξL,k - tL,k) must remain constant. Note that for acceptable voice quality, the

end-to-end delay budget for a voice application is limited to approximately 100ms [48].

In the following sections, the requirements for setting up a real time connection and to

maintain the data flow continuity are described.

2.2.1.4 Continuity of Data Flow

As previously described, data flow continuity requires that every packet generated by

the source at fixed intervals is consumed at the sink after a fixed delay but with the

same fixed intervals as generated at the source, without subject to packet loss or play-

out interruption for the duration of the connection. In the case of packet loss, this occurs

when a packet upon arrival finds no storing capacity in the buffer and is consequently

discarded. For the case of play-out interruption, this occurs when the sink is unable to

consume a packet due to unavailability of the packet at the play-out buffer in time.

Therefore it is necessary to determine the minimum buffer sizes (i.e. for the

spacer/policer buffer and the play-out buffer) that will prevent buffer overflow and sink

starvation for maintaining the data flow continuity.

2.2.1.4.1 Spacer buffer

In the case where the aggregate traffic stream is carried by the SBR connection and

under the assumptions in (2.12), cells are immediately presented to the network without

entering a buffer. However for DBR connections, a spacer buffer is required when its

peak cell rate is less than the peak cell rate of the aggregate AAL2 stream departing

from the CPS.



48

For the DBR connection, the departure of cells from the spacer buffer can also be

visualised as contiguous time intervals of variable length on the time axis. The service

rate Rp must be at least equal to or greater than the sustainable cell rate of the AAL2

aggregate traffic Rs (i.e. Rp ≥ Rs). For an arbitrary time t, there exists an integer v such

that

1v vtζ ζ +≤ < (2.18)

Note that ζv has been defined in (2.6). Let Nζ(t) denotes the integral fill level of the

spacer buffer. This value increases when cells arrived into the buffer and decreases

when they depart. Hence the fill level of the spacer buffer at time t is the difference

between the number of cells generated and the number of cells departed up to time t and

can be expressed as

( )

( )

0

( )

v

t

t

t

N t S d v

S d

ζ τ τ

τ τ

+

+

= −

=

∫

∫

(2.19)

There exists a cell denoted by s whose arrival time begins a busy period that includes

the interval (tv, ζv]. The number of cells generated in this interval can be expressed as

( ) ( ) ( )v v v

s sv

t

tt

S d S d S dζ ζ

ζ

τ τ τ τ τ+

= −∫ ∫ ∫ τ

s

(2.20)

Using the bound in (2.4), the first term in (2.20) can be expressed as

( ) ( )1v

s

v sS d Rζ

ζ

τ τ σ ζ ζ< + + −∫ (2.21)

and the second term can be expressed as 1 + (v - s). Using these, (2.20) can be

simplified to



49

)

( ) ( ) ( )

( ) (( ) ( )

( ) 2

1 1

1 1

1 1

v

v

v

v

v s st

v s p

IBT AAL Aggregate st

S d R v s

R v s

v s v s

S d R

ζ

ζ

τ τ σ ζ ζ

σ ζ ζ

σ

σ

τ τ τ

+

+

< + + − − + −⎡ ⎤ ⎡ ⎤⎣ ⎦ ⎣ ⎦

⎡ ⎤≤ + + − − + −⎡ ⎤⎣ ⎦⎣ ⎦= + + − − + −⎡ ⎤ ⎡ ⎤⎣ ⎦ ⎣ ⎦=

<

∫

∫

(2.22)

The maximum possible fill level during the service of the vth cell must occur at the

instant before the (v+1)st cell begins service. This is shown by

( ) ( )

( )

v+1

+v

v+1

+v+1

S

1 S

t

t

N t d

d

ζ

ζ

ζ

τ τ

τ τ

<

< +

∫

∫

(2.23)

The second term in (2.23) is given by (2.22), therefore (2.23) can be re-written as

( )( )

2

2

1 IBT AAL Aggregate s

IBT AAL Aggregate s

N t R

N t Rζ

ζ

τ

τ

< +

⎡ ⎤∴ = ⎢ ⎥

(2.24)

Note in (2.24), we have used the well known result that for every real number x and

integer n, n < x if and only if n < ⎡x⎤. Referring to (2.24), when the spacer buffer size is

chosen to be equal to or larger than ⎡τIBT AAL2 AggregateRs⎤, buffer overflow will not occur.

However if the buffer size is chosen smaller than ⎡τIBT AAL2 AggregateRs⎤, the buffer will

overflow whenever the fill level Nζ (t) reaches ⎡τIBT AAL2 AggregateRs⎤. This is summarised

as



50

1

2

2

no spacer buffer overflow

spacer buffer overflow

IBT AAL Aggregate s

IBT AAL Aggregate s

B R

B R

ζ

ζ

τ

τ

⎡ ⎤≥ ⎢ ⎥⎡ ⎤< ⎢ ⎥

(2.25)

2.2.1.4.2 Play-out buffer overflow

Packet arrivals to the play-out buffer can also be visualised as contiguous time intervals

of variable length on the time axis. For an arbitrary time t, there exists an integer w such

that

, ,L w L wtγ γ +≤ < (2.26)

Note that γL,w has been defined in (2.15). The buffer fill level at time t is the difference

between the number of packets that have arrived at the play-out buffer (denoted by w)

and the number of packets consumed by the sink. Letting NL,ξ(t) denote the fill level of

the play-out buffer, this is expressed as

( ), ,( )L LN t w t Rξ ξ 0 L⎢ ⎥= − −⎣ ⎦ (2.27)

Where , 1x R n n x x n x∀ ∈ ∈ = ⇔ − < ≤⎢ ⎥⎣ ⎦¢

Referring to (2.27), the term ⎣(t - ξL,0)RL⎦ gives the number of packets consumed by the

sink since play-out of the first packet.

In the interval [γL,w, γL,w+1), NL,ξ(t) is a non-decreasing function of t. It obtains its peak

level when the wth packet arrives. Substituting γL,w for t in (2.27), the fill level NL,ξ(t) is

expressed as

( ), , ,( )L L w LN t w Rξ γ ξ 0 L⎢ ⎥≤ − −⎣ ⎦ (2.28)

Equation (2.28) can be expanded using (2.15), (2.9), and (2.2) into



51

0

0

L

( )( )( ) ( )( )

( ) ( )( )( ) ( )( )

', ,

', ,

',0 ,0

', ,0 ,0

( ) ( , ) ,

, ,

, ,

( ) , ,

L i p i L L

L w i p i L L

L L i p i L

L L L i p i L

N t w L w D L w R

w t s L w D L w R

w t wT s L w D L w R

N t t s L w D L w R

ξ

ξ

ζ τ ξ

τ ξ

τ ξ

ξ τ

⎢ ⎥≤ − + + −⎣ ⎦⎢ ⎥= − + + + −⎣ ⎦

⎡ ⎤= − + + + + −⎢ ⎥⎡ ⎤≤ − − − −⎢ ⎥

(2.29)

Note that the identity x - ⎣y⎦ = ⎡x - y⎤ has been used where x is an integer and y is a real

number. In the SBR ATC, the value of si(L,w) is null as cells are immediately passed to

the network without entering the spacer/policer buffer.

An upper bound for NL,ξ(t) can be obtained by considering the first packet which

experiences the maximum queuing delay through the source CPS and the network, and

subsequent packets that experience minimum delay. Letting NL,ξ max denote the

maximum fill level NL,ξ(t) of the play-out buffer can reach, this can be expressed as

( ), , max ,0 ,0( )L L L L pN t N t D Rξ ξ ξ L⎡ ⎤≤ = − −⎢ ⎥ (2.30)

By choosing the size of the play-out buffer BL,ξ, to be at least equal to or greater than

NL,ξ max, buffer overflow will not occur. However, if a smaller buffer size is chosen,

whenever the fill level NL,ξ(t) reaches NL,ξ max, the play-out buffer will overflow. This is

summarised as

( )( )

, ,0 ,0

, ,0 ,0

no play-out buffer overflow

play-out buffer overflow

L L L p L

L L L p L

B t D R

B t D R

ξ

ξ

ξ

ξ

⎡ ⎤≥ − −⎢ ⎥⎡ ⎤< − −⎢ ⎥

(2.31)

2.2.1.4.3 Sink starvation

Sink starvation occurs whenever a sink is unable to consume a packet due to

unavailability of a packet at the corresponding play-out buffer in time. To prevent this

and for maintaining data flow continuity for each connection L, the following



52

relationship is required.

, ,L w L wξ γ> (2.32)

From (2.17), the time at which the first bit of the wth packet of connection L is read from

the play-out buffer is

( ) ( )', , , 0 ,0 ,L w L w i p i Lt s L D L D ξξ τ= + + + + (2.33)

Using (2.9) and (2.15), (2.33) can be rewritten as

( ) ( )( ) ( )

( ) ( )( ) ( )

', , ,

',

', ,

',

, ,

,0 ,0

, ,

,0 ,0

mL w L w i L w p i

i p i

mL w i L w i

i i L

s L w d D L w

s L D L D

s L w d L w

s L L D

L ξ

ξ

ξ γ τ

τ

γ τ

τ

= − − −

+ + + +

= − −

+ + +

(2.34)

Using the condition in (2.32), (2.34) is obtained as

( ) ( )( ) ( )

( ) ( ) ( ) (

', , ,

'

' ',

, 0 ,0

, , 0

, , ,0 ,0

L w L w i i L

i i

L i i i i

s L L D

s L w L w

D s L w L w s L L

ξ

ξ

ξ γ τ

τ

τ τ

− = + +

− − >

)∴ > + − −

(2.35)

For the DBR ATC:

Referring to (2.35), the bound on τi’(L,w) is τCDV as given in (2.13) and the bound on

si(L,w) is given in (2.10). Under the conditions where the peak cell rate of the DBR

connection Rp is equal to the sustainable cell rate of the aggregate traffic Rs with no

spacer buffer overflows, we find that after substituting (2.25) into (2.11), the upper

bound for Dspacer becomes the intrinsic burst tolerance of the aggregate AAL2 traffic

stream (i.e. τIBT AAL2 Aggregate). That is, for the DBR case

( ) 2 0 ,i IBT AAL Aggregates L w τ≤ < (2.36)



53

Furthermore, it is found that a cell accepted by the spacer in the DBR ATC is also found

conforming by the policer in the SBR ATC. Similarly, a cell that is discarded by the

spacer is also found non-conforming by the policer. The analysis showing these

relationships are found in Appendix B. Thus, with the appropriate selected values for

the DBR PCR and the spacer buffer size, there will not be any cell loss due to spacer

buffer overflow.

For sink starvation to occur, it can be considered that the first packet arrives with

minimum queuing delay (i.e. si(L,0) = 0 and τi’(L,0) = 0) and subsequent packets arrived

at the maximum delay. Therefore using (2.35) and making use of the upper bounds for

τi’(L,w) and si(L,w) in the DBR case, we can avoid sink starvation by delaying the first

packet of each connection L an amount DL,ξ which is given by

, 2 L IBT AAL Aggregate CDVD ξ τ τ≥ + (2.37)

For the SBR ATC:

For the SBR ATC and under the assumptions given in (2.12), the value of si(L,w) is nil

and the bound on τi’(L,w) is given in (2.16). That is, for the SBR case

( )' 2 0 ,i CDV IBT AAL AggregateL wτ τ τ≤ < + (2.38)

Similar to the DBR case, for sink starvation to occur, it can be considered that the first

packet arrives with minimum queuing delay (i.e. si(L,0) = 0 and τi’(L,0) = 0) and

subsequent packets arrived at the maximum delay. Therefore using (2.35) and making

use of the upper bounds for τi’(L,w) and si(L,w) in the SBR case, we can also avoid sink

starvation by delaying the first packet of each connection L an amount DL,ξ given by

, 2 L IBT AAL Aggregate CDVD ξ τ τ≥ + (2.39)

Note by comparing (2.37) and (2.39), it is found that the value of DL,ξ required to avoid

sink starvation is the same whether the DBR or SBR ATC is used.



54

2.2.1.5 Establishing the real time connection

When establishing a real time connection, the buffer sizes such as the spacer buffer and

the play-out buffer as well as the initial wait delay (or equalisation delay) for the first

packet to be read from the play-out buffer must be considered. In Section 2.2.1.4.2, the

minimum play-out buffer size has been determined as a function of the end-to-end delay

of the first packet, the propagation delay and the play-out rate. Here, the play-out buffer

size is again determined but will be obtained in terms of the queuing delay bounds that a

packet will experience in the CPS and the network.

For the DBR ATC:

The minimum spacer buffer size can be obtained from (2.25) as

2 IBT AAL Aggregate sB Rζ τ⎡ ⎤= ⎢ ⎥ (2.40)

The minimum equalisation delay can be obtained from (2.37) as

, 2 L IBT AAL Aggregate CDVD ξ τ τ= + (2.41)

The end-to-end delay using the minimum equalisation delay is

( ) ( )', , 2 , 0 ,0L k L k i p i IBT AAL Aggregate CDVt s L D Lξ τ τ τ− = + + + + (2.42)

The Intrinsic Burst Tolerance (τIBT AAL2 Aggregate) is inversely proportional to the

sustainable cell rate Rs (see (1.3)). If the value of Rs is set to the peak cell rate of the

aggregate traffic generated by the source CPS, then τIBT AAL2 Aggregate is 0. In the case of

the DBR transfer capability, if the peak cell rate is chosen larger than Rs then the end-to-

end delay for the connection will be smaller.

The play-out buffer size can be obtained by substituting (2.42) into (2.31) (for the case

of no buffer overflow) as



55

( ) ( )( )', 2 , 0 ,0L i i IBT AAL Aggregate CDV LB s L L Rξ τ τ τ⎡ ⎤≥ + + +⎢ ⎥ (2.43)

From (2.43), the worst case queuing delay is obtained when the first packet experiences

maximum delay. This can be expressed as

( ) ( )( )( ) ( )( )

( )

' 2

' 2

2

, 0 ,0

,0 ,0

2

i i IBT AAL Aggregate CDV L

i i IBT AAL Aggregate CDV L

IBT AAL Aggregate CDV L

s L L R

s L L R

R

τ τ τ

τ τ τ

τ τ

⎡ ⎤+ + +⎢ ⎥⎡ ⎤≤ + + +⎢ ⎥

< +

(2.44)

The minimum play-out buffer size is obtained from the upper bound of (2.44) as

( ), 2 2L IBT AAL Aggregate CDV LB Rξ τ τ= + (2.45)

For a real time DBR connection to be established and data flow continuity to be

maintained, dimensioning of the buffer sizes as well as the initial delay for the first

packet to be played out are important. This can be achieved by implementing the

minimum spacer buffer size given in (2.40), the minimum play-out buffer size given in

(2.45) as well as delaying the first packet to be played out by a minimum equalisation

delay as given in (2.41).

For the SBR ATC:

In the SBR ATC, no spacer buffer is required for the policer as cells are immediately

passed to the network without entering the spacer buffer. Using (2.39), the minimum

equalisation delay for the SBR case is also given by (2.41).

The end-to-end delay using the minimum equalisation delay is

( )', , 2 , 0L k L k p i IBT AAL Aggregate CDVt D Lξ τ τ τ− = + + + (2.46)



56

Similarly, the play-out buffer size can be obtained by substituting (2.46) into (2.31) as

( )( )', 2 , 0L i IBT AAL Aggregate CDV LB Lξ τ τ τ⎡ ⎤≥ + +⎢ ⎥R

)

(2.47)

From (2.47), the worst case queuing delay is obtained when the first packet experiences

maximum delay. This can be expressed as

( )(( )( )

( )

' 2

' 2

2

, 0

,0

2

i IBT AAL Aggregate CDV L

i IBT AAL Aggregate CDV L

IBT AAL Aggregate CDV L

L R

L R

R

τ τ τ

τ τ τ

τ τ

⎡ ⎤+ +⎢ ⎥⎡ ⎤≤ + +⎢ ⎥

< +

(2.48)

The minimum play-out buffer size is obtained from the upper bound of (2.48) as

( ), 2 2L IBT AAL Aggregate CDV LB Rξ τ τ= + (2.49)

Thus for the SBR case, a real time connection can be established and data flow

continuity maintained by implementing the minimum play-out buffer size given in

(2.49) as well as delaying the first packet to be played out by a minimum equalisation

delay as given in (2.41).

A summary of the derived design parameters necessary for the establishment of a real

time connection using AAL2 over a DBR and SBR connection is given in Table 4.

Design Parameter DBR ATC SBR ATC

Spacer Buffer Size ⎡τIBT AAL2 AggregateRs⎤ 0

Equalisation Delay τIBT AAL2 Aggregate + τCDV τIBT AAL2 Aggregate + τCDV

Play-out Buffer Size 2(τIBT AAL2 Aggregate + τCDV)RL 2(τIBT AAL2 Aggregate + τCDV)RL

Table 4: Design parameters for establishing AAL2 real time connections using the

DBR and SBR ATCs.



57

It can be noted from Table 4 that the equalisation delay and the play-out buffers when

using the DBR and SBR real time connections are equivalent. Given that both ATCs

give equal delay performance, the ATC selected for the AAL2 work described in later

chapters of this thesis is based on the DBR ATC for its simplicity.

2.2.2 Quality of Service (QoS) Framework

Once a connection is established, it is important to determine the voice quality across

the connection. This is described by the Quality of Service (QoS) framework where

acceptable QoS has been agreed between the user and the network providers prior to

establishing the voice connection. The QoS framework is defined by three parameters:

fixed delays, delay variation (denoted by Dα) and packet loss (denoted by α). Of these

three parameters, the delay variation parameter is the most important since, as we have

shown in this chapter, it quantifies the level of equalisation required for establishing the

voice connection and must be kept within strict bounds. When delay Dα (or τIBT AAL2

Aggregate) is set as the maximum buffer size, packets that experience delay greater than

this will find the buffer full and are then discarded. In this case, the fraction of packets

that will experience delay greater than Dα (denoted as α) becomes the loss probability.

Therefore when using the QoS framework, the number of connections over a fixed link

rate can be determined for any given sets of QoS requirements (i.e. maximum delay Dα

and packet loss α).

The QoS framework is used in later chapters as a tool for measuring the performance of

the AAL2 multiplexer. For voice applications with stringent delay budgets (i.e. around

100ms for one-way), the performance of the AAL2 multiplexer is measured in terms of

the number of sources that can be accommodated at a fixed link rate while maintaining

minimum acceptable voice quality. Acceptable voice quality for each voice connection

results by limiting source traffic admitted into the AAL2 multiplexer (i.e. by limiting

the number of sources supported). This is explained more clearly with the use of Figure

21. Referring to Figure 21, as the number of sources admitted into the AAL2



58

multiplexer increases, a greater number of AAL2 packets are being sent to the

multiplexer service queue, resulting in increased packet delay for a given fixed ATM

service rate. With a fixed size queue, AAL2 packets on arrival finding the queue full

(i.e. no place to fit in another packet) are then discarded. Continuous discarding of

AAL2 packets has a detrimental effect on the voice quality. Therefore the number of

sources that can be admitted into the queue must be limited so that the minimum

acceptable delay performance can be achieved for all the connections. This gives a

measure of the performance of the AAL2 multiplexer. In quantifying the number of

sources that can be accommodated while maintaining acceptable delay performance, the

(1-α) quantile delay must be specified. It is defined as the packet delay, Dα such that an

α fraction of packets have a delay greater than or equal to Dα. The delay Dα is

associated with the maximum multiplexing delay within the spacer buffer, defined as

Dspacer in (2.11), and is therefore a fraction of the end-to-end delay budget. Once the

values of Dα (or Dspacer) and α are chosen, the spacer buffer size, Bζ can be determined

by multiplying Dα with the PCR, Rp (i.e Bζ = Dα × Rp).

CHAPTER 3 AAL2 MULTIPLEXER MODEL


59

Chapter 3

AAL2 Multiplexer Model

In Chapter 2, a general ATM network incorporating AAL2 was examined. It was

determined, based on simplicity and practicality, that DBR was the most suitable ATC

to transport AAL2 traffic across the ATM network. It was also found that the AAL2

multiplexing delay plays an important role in terms of the equalisation delay that is

necessary for establishing an end-to-end voice connection. Therefore it is important to

understand and quantify the AAL2 multiplexer and its performance.

In this chapter, a general AAL2 multiplexer system is described. This system comprises

input sources and the AAL2 multiplexer. The input sources are modelled as voice

codecs with silence suppression and are characterised in terms of bandwidth and a

subjective quality score. The AAL2 multiplexer performance is described via the

statistical multiplexing gain that can be achieved while maintaining QoS requirements.

The study of the AAL2 multiplexer and its performance is investigated using the

network simulation tool called OPNET.



60

3.1. AAL2 Multiplexer System Model

The AAL2 multiplexer system model is shown in Figure 22. This system consists of

input sources (i.e. voice sources) and the AAL2 multiplexer. In the following sub-

sections, models for the input sources and the AAL2 multiplexer are described.

Voice sources

ATM cell creation

AAL2 Packetisation

1 n

FCFS Queue

AAL2 Multiplexer

DBR Service Rate R (kb/s)

Figure 22: General AAL2 multiplexer system model.

3.1.1. Voice Sources

In a voice conversation, it is observed that speech patterns within the conversation

consist of alternating talk and silence periods. A talk period is defined as the user

speaking over the phone (for example) and a silence period is defined as the user

listening to the other end of the phone. A simple speech pattern depicting talk and

silence periods is shown in Figure 23.



61

Voice Signal

Silence Period

Time V

olta

ge

Talk Period

Figure 23: Simple speech pattern.

Referring to Figure 23, talk periods are represented by high amplitude parts in the

waveform and silence periods are represented by the low amplitude areas. Note that

even though speech is not present in silence periods, the amplitude levels for the silence

periods in the analogue waveform in Figure 23 are not zero. This is a result of

background noises.

It has been found that both talk and silence intervals can be modelled as random

variables, with negative exponential distribution but different means [34]

[50] [51] [52] [53] [54]. The values for these means are shown in Table 5. These are taken

from the listed references and were measured using a number of voice conversation

samples. From this, it is observed that in each case the mean talk duration is less than

the mean silence period. The voice activity factor is defined as the ratio of the mean talk

period over a cycle of mean talk and silence periods given by

or 100%αα β= ×+ (3.1) Voice Activity Fact



62

Reference Mean Talk Interval α (ms) Mean Silence Interval β (ms)

[34] 400 600

[50] [51] 352 650

[51] 420 580

[52] 812 1579

Table 5: Typical mean values for talk and silence intervals.

When a voice conversation is to be conveyed across distance, a coder-decoder (codec) is

used. Analogue speech is converted by the coder into a digital form before being

transmitted across the network to be received and decoded by the designated user at the

receiving end. In the following sub-section, a model description for the digitised voice

traffic as well as the bandwidth characteristics for various codecs will be examined.

3.1.1.1. Codec Characteristics

Historically the bandwidth usage of a voice call over the PSTN is fixed at 64kb/s.

However the introduction of mobile telephony over the limited available bandwidth

across the air surface meant that speech was required to be compressed, thus reducing

the bandwidth required to support a connection. Also, silence suppression techniques

have been employed in these codecs in a further effort to reduce the average bandwidth

required to support a voice connection. Codecs with silence suppression techniques are

able to detect silence periods as described in the previous section through a voice

activity detector (VAD) and not generate packets during these periods (i.e. silence

suppression), thus further reducing the average bandwidth of a connection. Therefore

conversations described in Section 3.1.1 when digitised using these codecs can be

modelled as On/Off sources in terms of the packets they generate. The study of these

type of sources is found in many papers [50] [52] [53] [54] [55] [56]. Figure 24 illustrates

the packet creation characteristics of a typical voice codec employing silence

suppression that will be used to model voice sources within the OPNET simulation tool



63

(refer to Section 3.2).

ttalk

Figure 24: On/Off voice model characteristics.

Referring to Figure 24, during a talk period (ttalk), packets of size L bits are created at

fixed time intervals of length T. During a silence period (tsilence), there is no relevant

information except its duration and background noise levels but sending of these is non-

essential. Hence during a silence interval, no data needs to be sent as long as the

receiver is able to produce an adequate proxy of the background noise over the

appropriate silence period duration. This is achieved through the use of silence

suppression algorithms implemented within the voice codec, e.g. [57]. Referring to

Figure 24, the last packet containing the remaining voice information in the talk period

is sent at the beginning of the silence period after T time from the previous sent packet.

Some voice codec examples and their features such as nominal bandwidth, silence

suppression support, and frame interarrival times T are summarised in Table 6. Note

that the packet length L (shown in Figure 24) can be obtained by multiplying the codec

bandwidth with the frame interarrival time T. Note the voice codec G.711 PCM is not

used for mobile telephony and is shown in the table for illustration purposes.

Silence period Negative Exponential Distribution with mean β

T tsilence

T T

Time (secs) Packet Generation Events (L bits)

Talk period Negative Exponential Distribution with mean α



64

Codec Bandwidth

(kb/s)

Frame Interarrival,

T (ms)

VAD supported

G.711 PCM 64 0.125 No

E-ADPCM 40, 32, 24, 16 20 Yes

G.723.1 MPC-MLQ/ACELP 6.3, 5.3 30 Yes

G.729 CS-ACELP 8 10 Yes

RPE-LTP (GSM) 13 20 Yes

Table 6: Examples of voice codecs and their characteristics.

The use of different voice codecs results in a range of obtainable voice quality that is

perceived at the receiver.

3.1.1.2. Voice Codec Performance Measure

Voice quality of a connection is dependent on several issues such as voice clarity, codec

quality, echoes within the receivers (if any) and the signal to noise ratio (SNR) when

transmitting over the physical media. Collectively the overall quality is subjectively

measured and given in terms of either the Mean Opinion Score (MOS) [58] [59] [60] or

the Intrinsic R [59] [60] [61] [62].

MOS is a score obtained from subjective experiments. In these types of experiments, a

number of voice conversation samples from different codecs are given to a group of test

subjects to listen. The test subjects then rate these samples out of a maximum score of 5;

with 1 being the poorest quality through to 5 being toll quality. An average of these

scores gives an MOS score.

The Intrinsic R is a score predicted by the ETSI-Model (E-Model) that predicts the

subjective quality of a telephone call based on its characterising transmission

parameters. It combines the impairments caused by these parameters into a rating R

defined as



65

0 s d eR R I I I A= − − − + (3.2)

Referring to (3.2), the first term R0 represents the basic voice signal-to-noise ratio

(SNR). The second term Is includes impairments that occur simultaneously with the

voice signal, such as those caused by quantisation, by too loud a connection and by too

loud a side tone. The third term Id encompasses delayed impairments, including

impairments caused by talker and listener echo or by a loss of interactivity. The fourth

term Ie covers impairments caused by the use of special equipment; for example, each

low bit rate codec has an associated impairment value. This impairment term can also

be used to take into account the influence of packet loss. The fifth term A is the

expectation factor, which expresses the decrease in the rating R that a user is willing to

tolerate. An example of the factor A for mobile telephony is 10 [59] [60].

Intrinsic R rating covers a range between 0 to a 100. In [59] [60], it states that ratings R

in the ranges [90, 100], [80, 90], [70, 80], [60, 70], [50, 60] correspond to best, high,

medium, low and poor quality respectively. The rating R is related to MOS in reference

[61] and is as follows:

( ) ( )( )( )6

For 0 < R < 100

1 0.035 60 100 7 10MOS R R R −= + × + − − ×

(3.3)

For R < 01MOS =

(3.4)

For R > 1004.5MOS =

(3.5)

When performing subjective measurements to assess only the quality of a particular

voice codec in a test environment, other factors that contribute to voice degradation are

kept at a minimum. Under these conditions, the resulting MOS and Intrinsic R obtained

are essentially measures of the codec voice quality. Therefore either MOS or Intrinsic R

can be used to describe the codec performance. Examples of codec ratings in terms of



66

MOS and Intrinsic R are summarised in Table 7. It has been noted in [63] that a MOS

rating of 3.6 and above, is considered acceptable and will give good voice quality.

Rating Codec Bandwidth (kb/s)

MOS Intrinsic R

G.711 PCM 64 4.3 94.3

40 4.3 92.3

32 4.2 87.3

24 3.6 69.3

E-ADPCM

16 2.3 44.3

6.3 3.7 75.3 G.723.1 MPC-MLQ/ACELP

5.3 4.0 79.3

G.729 CS-ACELP 8 3.95 84.3

RPE-LTP (GSM) 13 3.7 75.3

Table 7: Examples of codec’s MOS and Intrinsic R.

The next part of the AAL2 multiplexer system to be examined is the AAL2 multiplexer

itself. A description of the AAL2 multiplexer as well as its performance will be

examined in the next section.

3.1.2. AAL2 Multiplexer Model

An example of an AAL2 multiplexer model is shown in Figure 22. Referring to Figure

22, n voice traffic sources are multiplexed into the AAL2 multiplexer. In a connection,

packets are collected by the AAL2 SSCS sublayer. These are AAL2 encapsulated (i.e.

prepended a 3-octet header) and queued into a FCFS queue that services at a rate R. At

every ATM cell creation instant (related to the service rate R), AAL2 packets are

removed from the queue and multiplexed into an AAL2 CPS PDU before being sent to



67

the ATM layer. Note that an AAL2 CPS PDU cell can contain AAL2 packets from

different voice connections.

In the next sub-section, a method for evaluating the performance of the AAL2

multiplexer will be described.

3.1.2.1. AAL2 Multiplexer Performance Measure

As was described previously, the average bit rate produced by a codec employing

silence suppression techniques is less than the codec’s peak rate. This results in

inefficient link utilization when peak rate is allocated to individual voice connections. In

an effort to increase bandwidth utilisation, statistical multiplexing is employed. When

considering the service of a number of these type of sources, statistical multiplexing can

be described as the process by which available bandwidth resulting from silence periods

in one voice conversation is used to send traffic from another connection. Because of

the random nature of the On/Off voice activity time, this will result in less aggregate

bandwidth being required to service a given number of voice connection than if the sum

of the codecs’ peak rate were used.

However, it is possible that the sum of the bandwidths from active sources can be

greater than the service rate capacity. Therefore a buffer has been implemented in the

AAL2 multiplexer (as shown in Figure 22) that prevents packet loss during periods

when the aggregate source traffic is greater than the service capacity by allowing

packets to be queued. This is illustrated in Figure 25. Note that is defined

as the sum of the codec peak rate and is defined as the sum of the codec

average rate.

_1

n

peak ratei

BW=∑

_1

n

average ratei

BW=∑



68

Incoming On/Off Traffic

Buffer Fill depends on number of active sources

_ _1 1

Service Raten n

average rate peak ratei i

BW BW= =

≤ <∑ ∑

Figure 25: Buffer in AAL2 multiplexer.

Referring to Figure 25, the buffer fill is dependent on the nature of the sources and the

chosen service rate. The size of this buffer determines the maximum waiting delay an

AAL2 packet will experience and becomes an important part of the QoS framework

described in Section 2.2.2.

It is useful then to quantify the level of statistical multiplexing that can be achieved and

we do this by defining a parameter called the Statistical Multiplexing Gain (SMG). For a

given aggregate AAL2 multiplexer service rate R, and given characteristics of the voice

sources, the statistical multiplexing gain is defined as the ratio of the number of sources

that can be accommodated when statistical multiplexing is employed (denoted by nmux)

to the number of sources that can be accommodated when peak rate is allocated to all

sources (denoted by npeak_rate) and is given by

(3.6)

_

mux

peak rate

nSMGn

=

In the case where AAL2 is used to support the voice connections, the value of npeak_rate

for a particular codec is related to the maximum allowable link loading γ, the

multiplexer service rate R (kb/s), the codec packet size L (bits), the AAL2 packet header

size LAAL2 (i.e. 24 bits) and the codec packet interarrival time T (sec) and is given by



69

( )( )_2

4753peak rate

AAL

RnL L T

γ=+

(3.7)

Note that the 47/53 in (3.7) accounts for payload to total cell ratio of the ATM cell

header.

The value of nmux is dependent on the nature of the source traffic and the QoS

requirements of the individual voice connections. A method of obtaining the value of

nmux and performance of the AAL2 multiplexer is by simulating the general AAL2

multiplexer system in an ATM network using the OPNET simulation tool. Descriptions

for modelling the AAL2 multiplexers and obtaining values for nmux are covered in the

following section.

3.2 Simulation in OPNET

The general AAL2 multiplexer system of Figure 22 is implemented in an ATM network

and is shown in Figure 26 as nodes in the OPNET simulation tool.

Figure 26: OPNET network model.

Referring to Figure 26, the source node is connected to a destination node via a physical

link with properties such as link capacity that can be set in OPNET. Both source and

destination nodes are implemented with identical functions. They contain voice sources,

AAL2 multiplexers at the source (and demultiplexers at the destination), ATM layer,



70

transmitters and receivers as well as a sink for the collection of statistical data. The

OPNET model of a node of the present network is shown in Figure 27.

Figure 27: OPNET node model.

Referring to Figure 27, each icon in the node model is implemented by a state-driven

process. During simulation, n independent voice sources are generated in the voice

codec process with traffic characteristics as described by the On/Off voice model that is

shown in Figure 24. Packet stream interrupts are generated when packets are sent from

one icon to the next (e.g. packets sent from voice sources to the AAL2). In the AAL2

process, voice packets are first prepended with 3 octet headers and then queued into a

FCFS buffer. Note that segmentation is performed for packets larger than 44 octets. At

every 1/PCR of the service rate (using DBR ATM transfer capability), AAL2 packets



71

are taken out of the buffer and multiplexed into an AAL2 CPS PDU cell before being

sent to the ATM to be prepended with ATM headers. These are then sent via the

transmitters and collected by the receivers at the destination.

Received ATM cells are then stripped of their headers before being sent to the AAL2

process from which the appropriate AAL2 headers are stripped. For packets that are

carried over multiple ATM cells, packet reassembly is performed in the AAL2 process.

The reassembled packets are then sent to the sink where statistical information such as

the end-to-end packet delays are obtained.

From the individual end-to-end packet delays, AAL2 multiplexing delays can be

determined and the performance of the AAL2 multiplexer can be obtained. The AAL2

multiplexing delays are calculated by subtracting the arrival times of the voice packets

at the sink with their packet creation times, the fixed propagation delay and the network

delays. To simplify the calculations involved, the propagation delay and the network

delays are made negligible. These calculated delays are then sorted and placed into

appropriate delay bins from which the delay histograms are obtained. Hence each bin

contains the number of packets that would have experienced delays within a certain

range. At the end of the simulation, the number of packets within each delay bin is

converted into a fraction of the total number of packets received and plotted against

delay. This plot gives the delay performance of the AAL2 multiplexer. A simulation

example using the OPNET models described previously will be used to illustrate

statistical multiplexing and the statistical multiplexing gains that are achievable for

different QoS delay requirements.



72

0

Simulation Example:

In this simulation example, voice sources used are based on the 8kb/s CS-ACELP

codecs with silence suppression. Its characteristics are shown in Table 6. It is assumed

that voice conversations follow a mean talk interval of 400ms and a mean silence

interval of 600ms [34]. Voice packets are queued immediately into a First-Come-First-

Serve (FCFS) buffer after being AAL2 encapsulated and are sent at fixed time intervals

using a service rate R of 384 kb/s (i.e. 1/4 of T1 rate).

The resultant delay performance of the AAL2 multiplexer for different number of

sources admitted into the multiplexer is plotted as a function of the delay d against the

probability that the packet delay exceeds d and is shown in Figure 28.

0 10 20 3-4

-3

-2

-1

0

α = 10-3

Dα =

16m

s

Log[

Pr{d

elay

>d}]

d (ms)

55 CELP Sources 57 CELP Sources 58 CELP Sources 59 CELP Sources 60 CELP Sources

Figure 28: Example of delay performance curves.

The number of voice connections that can be accommodated for a service rate of

384kb/s can be determined by choosing a set of QoS parameters: Dα and α (see Section



73

2.2.2) such that the resultant perceived voice quality at the receiver is essentially the

characteristics of the codec. Using typical values that result in negligible degradation of

voice quality from [34], Dα is chosen as 16ms for the maximum AAL2 multiplexing

delay and α is chosen as 10-3. According to this set of QoS parameters, the total number

of voice sources that can be accommodated from Figure 28 is 57 (i.e. nmux). Note that

with different sets of QoS parameters (i.e. Dα and α), a different total number of sources

can be supported. By setting either QoS parameters larger, more connections can be

accommodated. However, increasing these parameters may result in a degradation of

voice quality.

For this simulation example, it is assumed that link loading is close to 100% (i.e. 95%).

Therefore, the total number of CS-ACELP sources possible with peak rate allocation for

a service rate of 384kb/s is calculated using (3.7) to be 32 (i.e. npeak_rate). Therefore the

SMG that can be obtained is thus determined using (3.6) as 1.78.

This simulation example illustrates the statistical multiplexing of voice traffic into a

single-queued general AAL2 multiplexer. The model can be extended into multiple

queues and implemented as a prioritised AAL2 multiplexer. In the next chapter, a two-

queued prioritised AAL2 multiplexer is proposed and described.

Confidence Interval:

A confidence interval is required to show the accuracy of this simulation example as

well as the simulation studies that will follow in later chapters of the thesis. This

interval can be calculated by using the Method of Batch Means [64].

From [64], it is gathered that a sample mean from n simulations (denoted by Mn), a

sample variance (denoted by V2n) and a value (denoted by Zα/2,n-1) obtained from the

Student’s t-distribution for various degrees of freedom are required to calculate the

confidence interval. This is given by (3.8) for the sample variance and (3.9) for the

confidence interval.



74

( )22 11

n

n ii

V Xn

= −− ∑ nM

(3.8)

, 1 , 12 2,n n

n a n an n

V VM z M zn n− −

⎛ ⎞− +⎜ ⎟⎝ ⎠

(3.9)

where X is a value obtained from each simulation and Vn is the standard deviation of the

sample.

Using these equations, a confidence interval for the delay Dα can be calculated for the

simulation example shown in Figure 28. Note that there are a number of confidence

intervals that can be calculated from Figure 28 however we have chosen the point of

interest (i.e. α of -3.8 and Dα of 16ms) from the delay plot of the 55 CELP sources.

Using the same parameters as the simulation example but with different seeds and with

simulation duration of 6000ms, the obtained Dα values at α of -3.8 are shown in Table

8.



75

Simulation Run (with different seed) Delay Dα (ms) 1 14 2 18 3 12 4 14 5 14 6 17 7 12 8 14 9 25 10 17 11 23 12 26 13 12 14 16 15 17 16 12 17 28 18 20 19 15 20 15 21 16

Table 8: Simulation runs with different seeds.

The sample mean for Table 8 is calculated to be 17ms and the sample variance

calculated from (3.8) to be 22.9. For a 95% interval, this corresponds to a Zα/2,n-1 value

of 2.086. Given this information, the confidence interval is calculated from (3.9) to be

in the range 14.8ms to 19.2ms which also covers the point of interest. The simulation

studies to be followed in later chapters of this thesis will be based on this multiplexer

model with simulation durations greater than 6000ms, and thus it can be concluded that

the accuracy of the obtained simulation results will also be within the 95% confidence

interval.

CHAPTER 4 PRIORITY QUEUING


76

Chapter 4

Priority Queuing

In Chapter 3, a general AAL2 multiplexer system consisting of voice sources and a

single-queued AAL2 multiplexer was described. Input traffic is multiplexed and served

in a FCFS scheduling manner. This method may be unsuitable in cases where various

voice applications have different delay requirements.

In this chapter, we propose to differentiate these various voice applications within the

AAL2 multiplexer and serve those which require a tighter bound on the AAL2

multiplexing delay with a higher priority than those with a more relaxed delay bound

[65] [66]. In the simulation example shown in this chapter, two different voice call types

are examined. The assigning of high priority to the example call types is obtained by

comparing their associated delay budgets. From this we proceed to examine the delay

performance of the prioritised AAL2 multiplexer and to compare its performance to the

general model in terms of the statistical multiplexing gains (SMG) that they achieve.

4.1 Delay Budget

AAL2 multiplexing delay requirements can be obtained using a delay budget consisting

of various delay components described in Section 2.2.1.3. Note that the spacer/policer

buffer in the model shown in Figure 21 have been implemented as a single FCFS AAL2

multiplexer queue with DBR service shown in Figure 25. The maximum delay budget

DB,L for the Lth voice connection that is allowable in order for conversations to occur

without any lapse is typically around 100ms [48]. This includes the codec packetisation



77

delay T, the AAL2 multiplexing delay statistically bounded by Dspacer, the propagation

delay Dp, the network delays statistically bounded by the cell delay variation tolerance

(τCDV) (see Section 2.2.1.2) and the equalisation delay, ξL. Therefore DB,L is summarised

as

,B L spacer p CDV LD T D D τ ξ= + + + + (4.1)

The first delay component in (4.1) is the codec’s packetisation delay, which depends on

the type of codec used. Some examples are shown in Table 6. A typical codec

packetisation delay is around 20ms.

The AAL2 multiplexing delay Dspacer can be re-written as a fraction of the delay budget

DB,L as

,spacer B L p CDV LD D T D τ ξ= − − − − (4.2)

This is the maximum delay a packet can experience while waiting in the AAL2

multiplexer for service and is also the maximum buffer size. When describing the queue

size in terms of the number of AAL2 packets, this can be obtained by dividing the

maximum delay Dspacer with the service time of an AAL2 packet (denoted by Tservice).

The service time for an AAL2 packet is a function of the source packet size (denoted by

L (bits)), the length of the AAL2 header (denoted by LAAL2 (i.e. 24 bits)) and the

effective service rate of the AAL2 multiplexer (denoted by R (bits/sec)).

2AALservice

L LTR

+=

(4.3)

The propagation delay Dp is a function of the signal propagation speed in the physical

medium (denoted by Vs (m/s)) and the total distance transversed by the cell from source

to destination (denoted by Ds (m)).



78

sp

s

DDV

= (4.4)

Note in (4.4), the propagation speed in a typical medium such as air is 3×108m/s.

The last delay component in (4.1) is the equalisation delay ξL. In Section 2.2.1.3, the

minimum value of this delay was derived to be the sum of the upper bounds on the

variable delays experienced by the cell. These are the AAL2 multiplexing delay, Dspacer

and the network delay, τCDV (see Section 1.2.3.1.2).

L spacer CDD Vξ τ= + (4.5)

From (4.4) and (4.5), (4.2) can be re-written in terms of the codec packetisation T, the

distance Ds travelled by the cell from source to destination, the velocity Vs of the cell

travelling in the medium and the network delays τCDV.

( )2 2

2

2

2 2 2

spacer B p CDV spacer CDV

sspacer B CDV

s

sB CDV

sspacer

sBCDV

s

D D T D D

DD D T VDD T VD

DD TV

τ τ

τ

τ

τ

= − − − − +

= − − −

− − −=

= − − −

(4.6)

For a given network topology and distance between source-destination pairs, the

maximum multiplexing delay Dspacer tolerable can be determined via (4.6).

In the next section, a simulation example is used to illustrate differences in the

multiplexing delays obtained from the delay budgets for two types of calls (i.e. long

distance and local calls). Also, a prioritised AAL2 multiplexer will be proposed and its

delay performance compared to the single-queue general AAL2 multiplexer.



79

4.2 Scenario Example

In this simulation example, two different types of voice traffic; local (or national) calls

and international calls are presented to the single-queue general AAL2 multiplexer and

the proposed prioritised AAL2 multiplexer. A local call is defined as any connection

made within Australia and an international call is defined as any calls made between

Australia and overseas. In this example, both call types follow the characteristics of the

On/Off model described in Section 3.1.1.1. Note that GSM (i.e. RPE-LTP codecs) voice

codecs are used. The characteristics for this codec are shown in Table 6 [57] [67].

In a local call connection, the maximum distance travelled by a cell across the breadth

of Australia from Western Australia (i.e. Perth) to the Eastern states such as Sydney is

typically around 4,000km. For an international call, the maximum distance travelled by

the cell must be at least half the earth’s circumference and is typically around

19,000km. It has been assumed that in a local call, the number of hops a cell

experiences within the network is less than that for international calls. By examining the

travelling distance of a cell, the propagation delay of the international call dominates its

delay budget. Therefore, this results in a tighter delay constraint for the AAL2

multiplexer. With the local call, its delay budget is evenly distributed amongst the delay

components, hence resulting in a looser delay constraint for the AAL2 multiplexer. A

summary of the delay budgets for both call types is shown in Table 9. Note the

propagation delay Dp is obtained via (4.4) and the AAL2 multiplexing delay Dspacer is

derived via (4.6).



80

Delay Component Local/National Calls (ms)

International Calls (ms)

Total delay budget (DB,L) 100 100

Packetisation delay (T) 20 20

Propagation delay (Dp) 13 60

Network delay (τCDV) 1 5

Derived AAL2 multiplexing delay, Dspacer

32.5 5

Table 9: Delay budget parameters.

Also in this simulation example, it is assumed that conversations have a mean talk and

silence intervals of 420ms and 580ms respectively [51]. Both call types are assumed to

use GSM codecs that have a nominal bandwidth of 13kb/s and packetisation intervals of

length 20ms, resulting in an AAL2 packet size of 288 bits (i.e. voice packet of 264 bits

+ 24 bits AAL2 header). The service rate for these AAL2 multiplexers is assumed to be

1.536Mb/s. A summary of these simulation parameters is shown in Table 10.

Simulation Parameters Value

Voice packet length 288 bits

Mean Talk spurt interval 420ms

Mean Silence interval 580ms

ATM VCC service rate 1.536 Mb/s

Table 10: Simulation parameters.

In the following sub-sections, the performance for the general and prioritised AAL2

multiplexer are described and compared under this simulation example.



81

4.2.1 General AAL2 Multiplexer Performance

In the general AAL2 multiplexer model described in Section 3.1.2, both call types’

packets are serviced in a FCFS discipline regardless of the different delay requirements.

Therefore its delay performance described via the QoS framework (see Section 2.2.2)

must be small enough to accommodate long distance calls.

For the purpose of this study, the value for the probability loss α in the QoS requirement

to be used is 10-3, which is considered acceptable for voice applications [34]. Given that

this is a single-queue AAL2 multiplexer where traffic from both call types are present,

the maximum multiplexing delay Dα in the QoS requirement that can be set is 5ms to

accommodate the call type with the tightest delay requirements. Using the simulation

parameters in Table 10, Figure 29 shows the resultant performance delay curves for a

test stream when different numbers of voice sources are presented to the multiplexer.



82

0 5 10 15 20 25 30 3510-4

10-3

10-2

10-1

100

α = 10-3

Dα = 5ms

Dα = 32.5ms

n = 181 sources n = 182 sources n = 197 sources n = 198 sources

Pr {

dela

y >

d}

d (ms)

Figure 29: Delay performance curves for conventional AAL2 voice multiplexing

scheme.

As can be observed from Figure 29, there are two components of delay present when

multiplexing variable bit rate (VBR) voice sources. The first component (shown by the

steeper part of the delay curve) is due to phase coincidences of packets arriving into the

multiplexer at the same time from different sources. This delay is expected to be quite

small. The second component of delay (shown by the shallower part of the delay curve)

is the smoothing delay and results from the sum of the peak rates of the individual

sources being greater than the ATM service rate. In other words, when packets from a

talk spurt arrives into the multiplexer from more than 94 sources (i.e. the total number

of source with peak rate allocation (denoted by npeak_rate) as calculated using (3.7)), then

some form of smoothing will take place within the multiplexer. As can be seen from

Figure 29, this component of delay is dominant especially when the number of admitted

sources is large.



83

Figure 29 indicates that under the conditions specified in Table 9, the maximum number

of sources that can be multiplexed (denoted by nmux) while maintaining the required

QoS in delay variation (i.e. α=10-3 and Dα=5ms in this case) is 181 sources. Therefore

the level of statistical multiplexing gain (SMG) this represents can be calculated via

(3.6) as

_

18194

1.92

mux

peak rate

nSMGn

=

=

=

(4.7)

It should also be noted from Figure 29 that if the calls entering the AAL2 multiplexer

were only of the local/national type, then the allowable delay budget for the AAL2

multiplexer is 32.5ms. In this case, 197 sources could be admitted into the multiplexer

while maintaining acceptable delay performance.

4.2.2 Performance of Prioritised AAL2 Multiplexer

The proposed prioritised AAL2 multiplexer method is shown in Figure 30. In this

scheme, the multiplexer is able to distinguish (during call setup) between international

(designated high priority) and local/national (designated low priority) calls. Voice

packets from the high and low priority calls are then placed in separate queues each in a

FCFS manner. Upon an ATM cell transmission boundary, the ATM cell creation

function obtains AAL2 CPS packets from the high priority queue and forms a CPS PDU

for insertion into the ATM cell payload. CPS packets are only taken from the low

priority queue when the high priority queue is empty. Bandwidth starvation for the low

priority sources is avoided by limiting the number of high priority sources admitted into

the multiplexer. In the case where remaining payload results from the previous CPS

PDU transmission, this is placed first into the next CPS PDU payload regardless of the

presence of high priority packets.



84

Low Priority voice sources

High Priority voice sources

ATM cell creation

AAL2 Packetization

Low priority Service queue

High priority Service queue

m +1

n 1 m

Codec Sampling

VCC

Figure 30: Simulation model for prioritised AAL2 multiplexing scheme.

Simulations were carried out to determine the delay performance of the high and low

priority traffic classes using the prioritised AAL2 multiplexing scheme in Figure 30.

The parameters used in the study were as indicated in Table 10 with the total number of

sources kept constant at 197. Although the total number of sources was constant, the

mix of high and low priority sources was varied. Figure 31 and Figure 33 show the

performance delay curves for the low and high priority sources, respectively.



85

0 5 10 15 20 25 30 3510-4

10-3

10-2

10-1

100Total number of active sources = 197

α = 10-3

Dα = 32.5ms

Pr {

dela

y >

d}

d (ms)

High priority sources = 0 High priority sources = 5 High priority sources = 10

Figure 31: Delay performance curves for low priority sources.

Referring to Figure 31, it is evident that the nature of the performance delay curve for

low priority sources is similar to that of Figure 29, except that the actual delay curves is

a function of the number of high priority sources admitted into the multiplexer. It is

clear that to achieve the required Dα=32.5ms for low priority sources, no more than 10

high priority sources are allowed. This would be achieved in the multiplexer’s call

admission process. Of course, if the total number of sources were limited to less than

197 sources, then the number of admissible high priority sources would be increased. In

Table 11, we show the maximum number of admissible high priority sources when the

total number of connections is limited to 197, 196 and 195 sources, while maintaining

the Dα=32.5ms for the low priority sources (the delay for the high priority sources is

below 5ms in all cases). Note that as the number of high priority sources increases for

decreasing total number of sources, the number of low priority sources decreases.



86

Total number of sources

Maximum number of admissible high priority sources

197 8

196 12

195 33

Table 11: Maximum high priority sources for decreasing total number of sources.

Referring to Table 11, it is observed that the maximum number of admissible high

priority sources increases dramatically when the total number of sources decreased from

196 to 195. This can be explained using Figure 32 which shows the delay performance

curves for the number of low priority sources shown in Table 11.

0 5 10 15 20 25 30 35

-3

-2

-1

0

α = 10-3

Dα = 32.5ms

Pr {d

elay

> d

}

d (ms)

197 Sources 196 Sources 195 Sources

Figure 32: Delay performance curves for different number of low priority sources.

Referring to Figure 32, when the number of low priority sources is decreased, the delay



87

performance curves improve and for a given packet delay Dα, the probability of packet

loss α decreases, as can be observed at the QoS requirement co-ordinates (i.e. α of 10-3

and Dα of 32.5ms). By admitting high priority sources into the AAL2 multiplexer, the

low priority delay curves degrade and the probability of packet loss α for a given delay

Dα increases. The number of high priority sources that can be admitted for a given

number of low priority sources is therefore dependent on the initial value of α at Dα of

32.5ms. It can be observed that the initial value of α is much larger for the delay curve

with 196 low priority sources than for the delay curve with 195 low priority sources.

Therefore, a dramatic increase in the number of high priority sources that is admissible

in this case is observed.

0.0 0.5 1.0

10-3

10-2

10-1

100

Pr{d

elay

>d}

d(ms)

High priority sources = 5 High priority sources = 10

Figure 33: Delay performance curves for high priority sources.

Referring to Figure 33, it is evident that the nature of the performance delay curves for

high priority sources is different to that of the low priority sources. In this case, there is



88

no smoothing delay since the sum of the peak rates of the high priority sources is

always less than the ATM service rate and the only delay component is due to

simultaneous packet arrivals (i.e. phase coincidences) within the high priority service

queue. It can also be seen that the delay for the high priority sources is well below the

requirements of 5ms as indicated in Table 9 and is relatively insensitive to the number

of high priority sources.

The statistical multiplexing gain that is achievable for the prioritised AAL2 multiplexer

is determined to be 2.10 (i.e. nmux=197, npeak_rate=94 and SMG=197/94), which

represents a 9% increase in bandwidth efficiency over when the single queue AAL2

model is used. In general, the priority queuing scheme provides significant statistical

multiplexing gains when the number of low priority sources is much greater than the

high priority sources. In practice this would actually be the case. In fact, the largest

GSM service provider in Australia (Telstra) quotes that the fraction of international calls

at a base station is less than 0.2% of the total number of calls.

From the presented simulation example, significant statistical multiplexing gains were

realised when we based our assumptions that all sources adhere strongly to the

characteristics of the On/Off model. To ensure that all sources exhibit these

characteristics, some form of usage parameter control (UPC) is required. In the next

chapter, we examine the effect of source sensitivity on the delay performance of the

AAL2 multiplexer and describe a common UPC that might be used to enforce a source

to behave with the desired On/Off characteristics.

CHAPTER 5 SOURCE SENSITIVITY


89

Chapter 5

Source Sensitivity In previous chapters, it had been observed that due to the variable bit rate nature of

normal voice conversations, a high level of statistical multiplexing gain can be achieved

when multiplexing a number of voice channels onto a single voice trunk using AAL2.

The exact level of statistical multiplexing gain achievable is a function of the nature of

the individual voice sources and also depends on the strict QoS delay requirements of

the voice packets within the multiplexer service queue.

In this chapter, we examine the effects of source sensitivity on the performance of the

AAL2 multiplexer by considering sources that exhibit different characteristics to the

On/Off model described in Section 3.1.1.1. The desired behaviour of a source can be

obtained by enforcing its traffic through some form of usage parameter control. A

common Usage Parameter Control (UPC) and the selection of its parameters will be

described in this chapter.

5.1 Simulation Example

The simulation example under consideration is modelled using the single-queued

general AAL2 multiplexer described in Section 3.1.2. Source packet generation

characteristics such as packetisation time T, packet length L, mean talk and silence

periods of a conversation and link service rate parameters used in the simulation study

are shown in Table 12. Packets from all voice connections enter the multiplexer and are

placed into the service queue in a FCFS scheduling manner.



90

Simulation parameters Values

Packetisation (T) 5ms (i.e. 32kb/s E-ADPCM [68]

Packet length (L) 184 (20 byte payload + 3 bytes AAL2 header)

Talk interval [51] Negative Exponential

(mean = 420ms)

Silence interval [51] Negative Exponential

(mean = 580ms)

Service rate (R) T1(1.536Mb/s)

Table 12: Simulation attributes for simulation example.

As previously mentioned, the number of voice sources that the multiplexer can support

is dependent on both the aggregate service rate and the QoS requirements of the

individual sources (i.e. Dα and α as described in Section 2.2.2). For the purposes of this

study, typical values of α=10-3 and Dα=20ms are chosen [34]. Using the simulation

parameters in Table 12 the delay curves for a test stream were measured as a function of

the total number of sources admitted into the multiplexer. These results are shown in

Figure 34. From this figure it can be seen that a maximum of 75 sources can be

accommodated in the multiplexer while maintaining the specified QoS requirements.



91

1 10 20 3010-4

10-3

10-2

10-1

100

Dα =

20m

s

α = 10−3

Pr{d

queu

e>d}

d (ms)

75 Sources 76 Sources 77 Sources 78 Sources

Figure 34: Performance delay curves for conventional multiplexer.

5.2 Performance Sensitivity of AAL2 Multiplexer

In obtaining the maximum number of voice sources the multiplexer can accommodate,

it was necessary to assume that every codec produced traffic according to the On/Off

traffic model. In this section, we examine the effects of source sensitivity on the

performance of the AAL2 multiplexer by considering sources that exhibit different

characteristics to the On/Off model.

It is useful to consider how a source may produce traffic that is different to the normal

voice traffic model. First, the peak rate of the voice sources is unlikely to be exceeded,

given that the radio link layer at the air interface inherently limits the intensity of

incoming voice traffic [69]. Therefore, the source behaviour that will most deleteriously

affect the performance of the AAL2 multiplexer is when a source produces packets at

the peak rate. That is, there are no silence periods, where no packets are generated



92

within a source session. This will increase both the loading on the service queue and the

delays experienced by packets within that queue. A source may behave this way or in a

similar way for a number of reasons, for example:

• Hardware fault (e.g. Microphone) which, causes the voice codec to produce packets

always at the peak rate.

• The voice codec could be behaving correctly but on average, the voice activity is

high. E.g. Different speech patterns due to cultural background differences.

• Malicious users.

We now examine the effect on delay performance when a number of sources generate

constant rate traffic. In Figure 35 we show the effect of the delay performance of a test

stream within the AAL2 multiplexer system examined in Section 5.1 when 1, 2, and 10

out of a total of 75 sources admit traffic into the multiplexer constantly at the peak rate.

This figure shows an immediate and significant degradation in delay performance when

particular sources exhibit behaviour different to the assumed On/Off traffic model. In

fact it can be seen from Figure 35 that when only 2 sources exhibit constant packet

generation behaviour the delay performance of the test stream no longer meets the

specified QoS requirements (i.e. Dα=20ms and α=10-3). It is undesirable for the

multiplexer to be so sensitive to source behaviour and therefore some form of source

isolation is required. This will allow the multiplexer to guarantee performance to a

normal behaving source independent of the behaviour of other sources.



93

1 5 10 15 2010-4

10-3

10-2

10-1

100

Pro

b{de

lay>

d}

d (ms)

Original (75 Sources) 1 Constant Rate Source 2 Constant Rate Sources 10 Constant Rate Sources

Figure 35: Performance of AAL2 multiplexer for 1, 2 and 10 misbehaving sources.

5.3 Usage Parameter Control (UPC)

In the previous section we saw that if the AAL2 multiplexer is to guarantee a delay

bound it cannot merely assume a particular source behaviour but must be able to enforce

that behaviour. In other words, some form of Usage Parameter Control (UPC)

monitoring of the traffic on each connection is required. The function of the UPC is to

unconditionally discard packets from a connection where the call parameters are

exceeded thereby enabling the multiplexer to make guarantees to a connection

independent of the behaviour of other connections [51] [70] [71]. This is shown in Figure

36.



94

Voice sources

AAL2 Multiplexer

2 1 n

Policer Policer Policer

FCFS Queue

Service Rate, R (bits/sec)

To ATM layer

Figure 36: Source policing.

Referring to Figure 36, packets from each source are individually policed according to a

set of policer parameters. Conforming packets enter the AAL2 multiplexer while non-

conforming packets are discarded, thus allowing the output traffic from each connection

to exhibit the desired On/Off characteristics. The most common UPC (and also being

used here) is the token bucket policer. This is described in the following sub-section.

5.3.1 Token Bucket Policer

The token bucket policer has three traffic descriptors associated with it. These are the

peak packet rate (PPR), sustainable packet rate (SPR) and maximum burst size (MBS).

The first parameter, peak packet rate (PPR) is associated with the peak rate at which

packets from a connection can enter the multiplexer. In terms of a codec’s packetisation

time, T, this is defined as

1PPRT

= (5.1)

The second parameter, sustainable packet rate (SPR) represents the long-term average

input rate the policer will allow source packets through as conforming traffic. In



95

principle, the value of this parameter can be chosen from between the sources’ long-

term average rate and the PPR. However if the sum of the SPRs for all connections

entering the AAL2 multiplexer is greater than the service rate, R, then the total load on

the service queue can be greater than unity. To avoid the possibility of service queue

overload, the SPR parameter is considered as the maximum rate that the server can

guarantee service to a connection independent of the behaviour of other connections.

From this definition, and assuming all connections are of the same type with identical

SPR specifications, the maximum value for the SPR of individual connections will be

given by

max

RSPRn

= (5.2)

where nmax is the maximum number of individual connections that can be admitted into

the AAL2 multiplexer. Equivalently, given the specified SPR of a connection, the total

number of connections that can be admitted into the multiplexer without the possibility

of service queue overload is given by (5.3) via rewriting of (5.2).

maxRn

SPR=

(5.3)

The final parameter to be specified is the maximum burst size (MBS). This parameter in

conjunction with the SPR must be chosen by the network service provider such that

there is low probability that normal On/Off voice traffic through the policer is declared

non-compliant. Some studies have investigated the choice of token bucket parameters

for On/Off voice sources [72] [73]. These studies have concluded that it is very difficult

to choose meaningful parameters. Their studies have shown that either the SPR must be

chosen very close to the PPR or the MBS must be chosen very high (i.e. greater than

5000 packets when SPR equals the long term average rate of sources).

The token bucket polices traffic based on the algorithm model shown in Figure 37.



96

Packet arrival time, ta

ta < TATPPR -

τ

Conforming packet TATPPR = max(ta, TATPPR) + TPPRTATSPR = max(ta, TATSPR) + TSPR

Next packet

Non-conforming packet

No

Yes

No ta < TATSPR - τIBT voice source

Yes

Figure 37: Token bucket algorithm.

Referring to Figure 37, when a packet arrives, it is first tested for peak packet rate

conformance. This is achieved by comparing its arrival time ta with its theoretical

arrival time denoted by TATPPR less the tolerance τ. The value of TATPPR is equal to the

interarrival time T and the tolerance τ is to account for the jitter effect due to timing

differences when sending packets. If the packet is found conforming to the peak packet

rate, it is then tested for sustainable packet rate conformance by comparing the value of

ta with the value obtained from subtracting the Intrinsic Burst Tolerance denoted by

τIBT voice source from the theoretical arrival time associated with the sustainable packet rate

denoted by TATSPR. A packet is declared non-conforming when it fails either tests and is



97

discarded. However when packets are conforming, the TATPPR and TATSPR values are

updated. Note the intrinsic burst tolerance τIBT voice source in Figure 37 is defined as

( ) 1 11IBT voice source MBS

SPR PPRτ ⎛ ⎞= − −⎜ ⎟

⎝ ⎠

(5.4)

5.3.2 Selection of Token Bucket Parameters

From the previous descriptions in Section 5.3.1, the selection of the appropriate token

bucket parameters for the simulation example are described in this section. The first

bucket parameter to be determined is the peak packet rate (PPR) and is calculated using

the packetisation value in Table 12 and (5.1) as

10.005200 packets/sec

PPR =

=

(5.5)

The SPR is calculated by first obtaining the number of sources that can be admitted into

the AAL2 multiplexer. In Figure 34, it is observed that 75 sources (i.e. nmax) can be

admitted into the AAL2 multiplexer while meeting the QoS requirements. From this

nmax value and the service rate in Table 12, the following SPR is calculated using (5.2)

as

61.536 1075

111 packets/sec

SPR ×=

=

(5.6)

Using the SPR value obtained in (5.6), the violation probability of packets generated

according to the On/Off traffic model through a policer as a function of the SPR and

MBS has been measured through simulation and is shown in Figure 38.



98

200 400 600 800 1000 1200 1400 1600 180010-5

10-4

10-3

10-2

10-1

100

MBS = 1440 packets/sec

Vio

latio

n P

roba

bilit

y

Maximum Burst Size (MBS)

SPR = 111 pps(nmax = 75 sources)

Figure 38: Influence of SPR and MBS on the violation probability of a token

bucket policer.

The selection of a MBS value is obtained by choosing a suitable violation probability

such that voice quality is still acceptable after being policed. It is assumed that a

violation probability of 10-4 will give acceptable voice quality. In Section 5.3.4 where

source policing is performed in an actual experiment, it is observed that choosing this

violation probability results in acceptable voice quality. Referring to Figure 38, the

violation probability of 10-4 corresponds to a MBS of 1440 packets. The token bucket

parameters for this simulation study are summarised in Table 13.



99

Token bucket parameters Values

PPR 200 packets/sec

SPR 111 packets/sec

MBS 1440 packets

Table 13: Token bucket parameters for a VBR voice source.

5.3.3 Simulation with Obtained Token Bucket Parameters

Upon obtaining the token bucket parameters, we now consider the delay performance of

the multiplexer with the presence of appropriate UPCs. Using Figure 36 and with the

simulation parameters shown in Table 12, the delay curves of a test stream within the

AAL2 multiplexer when 1, 2, and 10 out of a total of 75 sources admit traffic into the

multiplexer constantly at the peak rate are shown in Figure 39. We note that the delay

performance when the UPC is present is much less sensitive to misbehaving sources

compared to the case when no UPC is used (see Figure 34). In fact, it can be seen from

Figure 39 that the QoS requirements of Dα=20ms and α=10-3 are met even when there

are 10 sources presenting traffic to the multiplexer constantly at the peak rate.



100

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 201E-5

1E-4

1E-3

0.01

0.1

1

Pro

b{de

lay>

d}

d (ms)

Original (75 sources) Policer - 1 Peak Source Policer - 2 Peak Source Policer - 10 Peak Source

Figure 39: Performance delay curves for misbehaving sources with policer.

The simulation studies described above have shown that implementing a token bucket

policer enables sources to be isolated such that the behaviour of one source has only a

small effect on other sources, thus maintaining QoS guarantees for each connection. In

the next section, a software implementation of the token bucket policer is carried out in

an experiment involving some pre-recorded conversation samples. Descriptions for the

selection of appropriate token bucket parameters in policing these conversation samples

are presented.

5.3.4 Practical Policing Experiment

In the previous section, the MBS policing parameter used for voice sources was

obtained by measuring the violation probability of a theoretical On/Off voice source

through the policer. In this section, an attempt is made to verify the choice of the

policing parameters by applying them to real voice sources.



101

In this experiment, conversation samples are taken between two test subjects; one of

which is in the laboratory room and the other which is located in a remote location. A

microphone connected to the computer is set up in the laboratory room to record a one

way conversation (i.e. only the voice of the test subject in the laboratory room is

recorded) using 64kb/s Pulse Code Modulation (PCM). Factors (i.e. outside noises and

echoes in the room) that degrade the conversation are kept minimal. Some recorded

voice samples and their associated voice activity are shown in Table 14 (see Appendix

C for silence suppression parameters).

Voice sample Voice activity (%)

Conversation A 50.7

Conversation B 44.8

Conversation C 39.3

Reading a book 91.1

Table 14: Voice activity for some recorded voice samples.

Referring to Table 14, voice activities of these samples correspond to those in [34] [52].

It ranges from 30% to 50% for a normal conversation while for a one-way conversation

such as reading a book, voice activity is well over 50%. For this experiment, we have

chosen conversation A to be policed. Using the token parameters shown in Table 13, the

violation probability is experimentally measured and the resultant curve is obtained and

plotted in Figure 40. This figure also shows two additional violation probability curves:

the curve obtained when a theoretical modelled voice source is used with a voice

activity of 42% (see Figure 38) and when a theoretical modelled voice source is used

but with a voice activity of 50%.



102

500 1000 1500 2000 25001E-5

1E-4

1E-3

0.01

0.1

MBS = 1440

Violation Probability = 4x10-3

MBS = 2010

Violation Probability = 10-4

Viol

atio

n Pr

obab

ility

MBS (Packets)

Theoretical (50% activity) Experiment (50% activity) Theoretical (42% activity)

Figure 40: Comparisons of violation probability.

Referring to Figure 40, it is observed that the curve obtained from Figure 38 does not

match the curve for conversation A. This can be explained by the fact that the voice

activity for conversation A is much higher than the 42% voice activity shown in Figure

38. The violation probability curve for conversation A matches that of the theoretically

modelled case for 50% voice activity.

When applying the MBS obtained in Section 5.3.2 for the conversation A in Figure 40

together with a SPR of 111 packets/sec, a violation probability of 4×10-3 is

experimentally measured. When this policer is used, it was noted that voice blurring can

be heard at the end of each long talk interval due to packet discard. This can be

eliminated by choosing a larger MBS value which gives a smaller violation probability.

It was experimentally measured that the minimum required MBS to achieve good voice

quality is 2010 packets (which corresponded to an experimentally measured violation

probability of 10-4). Similar procedures are required to obtain suitable token bucket



103

parameters for other voice samples shown in Table 14. This is due to fluctuations in

voice activities between different conversations. The experiments carried out above

highlight the difficulties in choosing appropriate policing parameters (especially the

MBS values) for real life VBR voice sources.

5.4 Conclusion

Due to the variable bit rate nature of voice, sources that do not conform to the expected

On/Off behaviour will affect the performance of the AAL2 multiplexer. Source isolation

between sources is required to eliminate the effects of non-conforming sources in the

form of the usage parameter control (UPC). A common UPC is the token bucket

policer. Through the appropriate policer parameters, degradation of the AAL2

multiplexer performance due to non-conforming sources is less significant. Also, the

multiplexer is able to achieve high statistical multiplexing gain while maintaining QoS

guarantees to individual voice connections.

When policing sources, different sets of UPC parameters are required due to the VBR

nature of voice. In the next chapter, we examine the effectiveness of source policing by

considering the worst case traffic behaviour that can pass the UPC and re-evaluate the

delay performance of the AAL2 multiplexer.

CHAPTER 6 ALTERNATIVE MULTIPLEXING METHOD


104

Chapter 6

Alternative Multiplexing Method

In Chapter 5, we observed that the behaviour of a source has a significant impact on the

delay performance of the AAL2 multiplexer when it exhibits traffic generating

characteristics different to the conventional On/Off model (i.e. non-conforming

sources). This also impacts on the performance of the conforming sources. Therefore to

maintain QoS guarantees to all connections, it is necessary to enforce a source’s

behaviour through some form of Usage Parameter Control (UPC). A common UPC is

the token bucket policer. Through the appropriate token bucket parameters (i.e. PPR,

SPR and MBS), source isolation is achieved and QoS guarantees can be maintained.

In this chapter, we re-evaluate the performance of the AAL2 multiplexer by considering

the worst case traffic behaviour that can pass the UPC. This is achieved by obtaining the

number of sources that can be accommodated using the derived worst case delay within

the multiplexer. From these results we determine if statistical multiplexing is a suitable

method for increasing link utilisation.

6.1 Simulation Example

The simulation example is based on the model shown in Figure 41 using E-ADPCM

codecs. Voice conversations are assumed to exhibit On/Off characteristics with mean

talk interval of 400ms and mean silence interval of 600ms. Source traffic is individually

policed before being sent to the AAL2 multiplexer that services packets in a FCFS

scheduling manner and at a rate equal to the T1 link (i.e. 1.536Mb/s). These simulation



105

parameters are summarised in Table 15.

Voice sources

AAL2 Multiplexer

2 1 n

Policer Policer Policer

FCFS Queue

Service Rate, R (bits/sec)

To ATM layer

Figure 41: Simulation model for source policing.

Simulation Parameters Values

Packetisation Time 5ms (E-ADPCM)

AAL2 Packet Length 184 bits (20 byte payload and 3 byte AAL2 header)

Talk Interval Negative exponential with mean 400ms

Link Service Rate, R T1 (1.536Mb/s)

Table 15: Simulation attributes for conventional voice and data multiplexing.

In Figure 42 the resultant probability delay curves are shown for a test stream when

different numbers of voice sources are presented to the multiplexer. This figure

indicates that under the conditions outlined in Table 15, the maximum number of

sources that can be multiplexed while maintaining the required QoS in delay variation

of α=10-3 and Dα=15ms (see Section 2.2.2) is 77 sources. Note that these are typical

scenario values as illustrated in Section 3.2.



106

0 10 20 30-5

-4

-3

-2

-1

0

Dα = 15 ms

α = 10-3

log(

Pr {d

elay

> d

})

d (ms)

n = 79 sources n = 78 sources n = 77 sources

Figure 42: Delay curves for FCFS multiplexing scheme.

From the results obtained in Figure 42, the level of statistical multiplexing gain (SMG)

can be determined using (3.6). The number of connections allowed with peak rate

allocation denoted as npeak_rate is given by

( ) ( )

peak_rate

6

3

Service RateSource Peak Rate

1.536 10184 5 10

41 sources

n

−

⎢ ⎥= ⎢ ⎥⎣ ⎦⎢ ⎥×⎢ ⎥=

×⎢ ⎥⎣ ⎦=

(6.1)

Therefore the level of SMG that is achievable is 1.88. This value indicates that a

significant increase in the number of sources can be admitted into the multiplexer when

statistical multiplexing is employed. However, the SMG is only possible when it is

based on the assumption that all of these sources exhibit the desired On/Off traffic

generating characteristics (i.e. are conforming sources) or that only a small number of



107

sources are non-conforming at a time. In the following section, we re-evaluate the delay

performance of the AAL2 multiplexer by considering the worst case traffic behaviour

that is passed by the UPC.

6.2 Worst Case Behaviour of the Token Bucket Parameters

In the analysis to obtain the worst case multiplexer delay, we assume that there are nmax

active connections into the multiplexer and that each connection is policed with

identical sets of token bucket parameters; PPR, SPR and MBS (see Section 5.3.1). For

the FCFS scheduling queue shown in Figure 41, the worst case traffic behaviour occurs

when a burst of packets equal to the MBS arrives at the multiplexer from each

connection at the same time [74].

For this case, the duration of a burst of size MBS packets denoted by TMBS is

( )1MBS

MBST

PPR−

= (6.2)

The total number of packets denoted as Ntotal arriving from nmax active connections

during the time TMBS is then

maxtotalN n MBS= × (6.3)

From (5.3) in Section 5.3.1, the multiplexer service rate R is

maxR n SPR= × (6.4)

The number of packets denoted by NB that can be served in time TMBS is

( ) 1B MBSN T R= × + (6.5)

The unity term in (6.5) appears since we assume that the first packet is served

immediately upon its arrival into the multiplexer. The number of packets left in the



108

service queue after the last packet of any connection’s burst arrives denoted as NL is

L total BN N N= − (6.6)

The last packet of the last connection’s burst will then be served at a time

maxLN

Rτ =

(6.7)

after its arrival into the multiplexer service queue. Using (6.2) to (6.6), (6.7) can be re-

written as

max phase IBT voice sourceτ τ τ= + (6.8)

where

( )max1 1phase nR

τ = − (6.9)

and

( ) 1 11IBT voice source MBS

SPR PPRτ ⎛ ⎞= − −⎜ ⎟

⎝ ⎠

(6.10)

Equation (6.8) represents the worst-case delay within the multiplexer using the FCFS

service discipline. It consists of two different delay components. The first delay

component, τphase arises due to phase coincidences of packet arrivals into the service

queue. The second component, τIBT voice source arises when a connection’s minimum

guaranteed service rate is less than its peak rate and thus is associated with smoothing

delay within the multiplexer service queue. This component of delay is expected to be

the dominant of the two.

Although (6.8) described the maximum delay for a FCFS service discipline, other types

of service disciplines within the multiplexer can be considered. A list of delay bounds



109

for some rate based scheduling disciplines that have appeared in the literature is given in

Table 16. In all the studies listed in Table 16, it was assumed that the input traffic

conformed to the token bucket parameters (PPR, SPR, MBS) as it is presented to the

network. Given this assumption, the second term in each of the bounds in Table 16 turns

out to be the τIBT voice source of the signal, as given by (6.10). Also note that the bound on

this smoothing delay is independent of the behaviour of other real time signals. The first

term in each of the bounds given in Table 16 is that associated with phase coincidences

of packet arrivals and its value depends on the actual scheduler being used.

Scheduling disciplines Variable delay bounds

Parekh-Gallager [7] PGPS PGPSphase IBTD τ τ< +

Golestani [8] SCFQ SCFQphase IBTD τ τ< +

Stiliadis-Varma [9] LR LRphase IBTD τ τ< +

Goyal-Vin-Cheng [10] SFQ SFQphase IBTD τ τ< +

Table 16: Variable delay bounds for various scheduling disciplines.

For a given peak packet rate PPR, service rate R, and acceptable loss probability α, the

following procedure can now be followed to determine the maximum delay in the

multiplexer as a function of the number of sources admitted into the multiplexer.

1. Choose a value for maximum number of allowed sources, nmax.

2. Using (5.2), determine the maximum value of SPR that can be allocated to

individual voice connections.

3. Using the value of SPR obtained from the above step, choose the value of MBS

that will achieve the acceptable violation probability α.

4. Use (6.8) to calculate the worst case delay.



110

Applying the above procedure to the simulation example in this chapter, the maximum

delay through the AAL2 multiplexer under worst case traffic conditions is shown in

Figure 43.

40 50 60 70 80100

101

102

103

104

n = 42

n = 43

Dα = 15msMax

imum

Dela

y (m

s)

Number of Sources

Figure 43: Maximum delay through AAL2 multiplexer under worst case input

traffic behaviour.

Figure 43 indicates the maximum number of sources that can be multiplexed while

maintaining the required QoS in delay variation (i.e. α=10-3 and Dα=15ms) is 42

sources which represents only one additional source that is achievable over peak rate

allocation (see (6.1)). The level of statistical multiplexing gain that this represents is

calculated to be 1.02. This indicates that a significantly lower value of statistical

multiplexing gain is obtained when the worst case traffic behaviour of the multiplexer is

considered than when a particular source behaviour is assumed and not policed [74].

Also these results indicate that policing this type of VBR voice traffic is only

meaningful at the peak packet rate, and therefore peak rate allocation is the only

alternative. To increase link efficiency, alternative methods other than statistical



111

multiplexing of voice sources must be examined. This is shown in the next section.

6.3 Future Work - Integrated Multiplexing Scheme

In this section, we propose an alternative multiplexing scheme called integrated

multiplexing where the QoS can still be guaranteed to each and every voice connection.

With this scheme, bandwidth utilisation is increased by integrating low volume data

applications with voice within the ATM channel or within the AAL2 channel.

Figure 44 shows data and voice integration within the ATM channel. Note that we have

assumed the total traffic from both voice and data do not exceed the service rate of the

queue. Referring to this figure, data packets and voice packets are queued into different

prioritised FCFS queues within the AAL2 multiplexer where voice is assigned higher

priority than data. Upon an ATM cell transmission boundary, the ATM cell creation

function obtains AAL2 packets from the higher priority queue and forms a CPS PDU

for insertion into the ATM cell payload. AAL2 packets are only taken from the low

priority queue (i.e. data application) when the high priority queue (i.e. voice) is empty.

Bandwidth starvation for the low priority sources is avoided by limiting the number of

high priority sources admitted into the AAL2 multiplexer (see Chapter 4).



112

Data Sources Voice Sources

1 n 1 m

Low Priority

High Priority FCFS Queue FCFS Queue

ATM

Figure 44: Integration of data and voice within ATM channel.

The advantage of this method is that during silence periods, the entire link bandwidth

can be used to send data packets resulting in relatively low packet delays. However,

when using this integration method and fixed queue size, data packet losses are

expected.

Another method under the integrated multiplexing scheme is to integrate data and voice

within the AAL2 channel. This is shown in Figure 45. We have also assumed the total

traffic from both voice and data do not exceed the service rate of the queue. Referring to

this figure, each voice source has an in-built voice activity detector (VAD) that detects

silence periods and sends segmented data packets. Note that we have assumed constant

data packet generation that is independent of voice packet generation. In each source, a

small buffer is used for storing data packets and is required to prevent packet discard

during speech activity. Both data and voice packets are then sent to a FCFS queue and

served at a constant rate.



113

ATM

Voice Packet

Data Packets

VAD

Data Packets

Voice Packet

VAD

Sources

FCFS Queue Data packets queued. Segmented packets send during silence periods

AAL2 Multiplexer

Figure 45: Integration of data and voice within AAL2 channel.

This method has not been realised in any technologies. However, the 3G mobile

networks show certain capabilities of incorporating small volume data applications with

voice within the same AAL2 channel. Hence it is proposed to use 3G as the platform to

carry out this multiplexing method. The implementation of this scheme requires the use

of reserved values within the 3G headers. At the source, both voice packets and data

packets are each prepended with a 3G header that contains a specific reserved frame

number within the range 10 to 13 in the Frame Type field [75]. An example of this is to

select reserved frame number 10 for voice traffic and reserved frame number 11 for data

traffic. This field indicates to the codec the type of information contained in the packet.

These packets are then AAL2 encapsulated and multiplexed into AAL2 CPS PDU cells

before being transported across the network. At the destination, associated AAL2

headers are stripped and the packets are then passed to the appropriate 3G receivers.

These receivers are able to distinguish between voice packets and data packets based on

the frame numbers in the Frame Type field and to play these packets accordingly.



114

Some application examples of this multiplexing scheme include multiparty multimedia

conferencing, picture phones, multimedia bulletin boards and multimedia mail, file

transfer during speech, automated teller machines with voice support, credit card

verification and other small volume data applications [53].

The advantage with this method is that packet loss is non-existent as the data buffer in

the 3G source can never overflow. The packet delay is much larger than those

experienced in the AAL2 multiplexer when data and voice are integrated within the

ATM channel, as data packets can only be sent during silence periods of the

conversations. This is acceptable as the performance of data applications is dependent

only on packet loss. Therefore this method not only increases the link efficiency but

also achieves a better performance for low volume data applications.

6.4 Conclusion

In this chapter, we have re-evaluated the delay performance of the AAL2 multiplexer by

considering the worst case traffic behaviour. It was shown that under worst case

conditions, the statistical multiplexing gain that is achievable is significantly lower and

results in only one additional source over the 41 sources that can be accommodated over

peak rate allocation. An alternative multiplexing method has been proposed (i.e.

integrated multiplexing). Under this multiplexing scheme, link utilisation is increased

by integrating low volume data applications with voice within the ATM channel or

within the AAL2 channel. It is observed that the integration of voice and data within the

AAL2 channel achieves better performance for the data applications in terms of packet

loss.

CHAPTER 7 CONCLUSION


115

Chapter 7

Conclusion

AAL2 has been developed to efficiently carry low and variable bit rate traffic such as

voice and small data applications. It is able to achieve high packing efficiency without

incurring additional packet delay. Due to its multiplexing capability, AAL2 has found

its place in many recent mobile technologies such as CDMA and 3G. In this thesis, the

delay performance of AAL2 multiplexers and in particular the traffic management

issues associated with the multiplexer was examined. The specific and original

outcomes of this thesis are:

• A QoS framework for the study of AAL2 voice multiplexers was developed.

The QoS framework is defined by three parameters: fixed delays, delay variation

(denoted by Dα) and packet loss (denoted by α). When delay Dα is set as the

maximum buffer size, packets that experience delay greater than this will find

the buffer full and are then discarded. In this case, the fraction of packets that

will experience delay greater than Dα (denoted as α) becomes the loss

probability. Therefore when using the QoS framework, the number of

connections over a fixed link rate can be determined for any given set of QoS

requirements (i.e. maximum delay Dα and packet loss α).

• The performance limitation of the conventional AAL2 multiplexer system for

the transportation of VBR voice was examined. This general system consists of

input sources and the AAL2 multiplexer itself. Input sources are modelled as

voice sources that exhibit On/Off characteristics. The performance of these



116

voice sources is measured in terms of the MOS and the intrinsic R. When

measuring the performance of the single queue AAL2 multiplexer, statistical

multiplexing is used. It was observed that through statistical multiplexing, more

sources can be accommodated by the multiplexer than the number of sources

that could be allocated peak rate, thus increasing bandwidth utilisation.

• An extension of the conventional AAL2 multiplexer in the form of prioritised

multiplexer to achieve higher bandwidth utilisation was proposed and examined.

This prioritised multiplexer isolates the multiplexing requirements between

different sources and offers a modest (9%) increase in the statistical

multiplexing gain.

• The performance sensitivity of the AAL2 multiplexer with respect to input voice

traffic was examined. Significant degradation to the delay performance of the

AAL2 was observed when sources exhibit traffic generation characteristics

different to the On/Off voice model.

• Source policing of VBR voice sources are examined. It was observed that when

statistical multiplexing is employed, the behaviour of a source must be enforced

through some form of Usage Parameter Control (UPC). This is to prevent

degradation on the delay performance of the AAL2 due to non-conforming

sources (i.e. sources that exhibit non-On/Off traffic characteristics). A common

UPC is the token bucket policer. Through the appropriate token bucket

parameters, isolation between sources is achieved and QoS guarantees can be

maintained for all connections even when a small fraction of the number of

sources is non-conforming.

• Statistical multiplexing and the extent to which it is possible for real time VBR

voice were examined. Here it was observed that statistical multiplexing is not

able to guarantee a significant gain in the number of sources that can be

accommodated by the multiplexer. In the simulation example, under the worst-



117

case traffic scenario, only one additional source is gained over that admitted for

peak rate allocation. Therefore given that statistical multiplexing is not able to

guarantee any significant gains, it is only meaningful then to police the peak rate

of the source.

• An alternative multiplexing method to statistical multiplexing in terms of

utilising available bandwidth was proposed and described. The alternative

multiplexing method is peak rate allocation. Under this method, bandwidth

utilisation is increased by integrating small volume data applications with voice

in an ATM channel or an AAL2 channel. For the integration of data and voice in

an ATM channel, this is implemented using prioritised queues (Chapter 4). The

advantage of this method is that packet delay is much lower as the whole link is

used to send data traffic during silence periods of the voice traffic. With the

integration of data and voice in an AAL2 channel, it has been proposed that this

method be implemented based on the 3G mobile technology. Voice and small

volume data applications are distinguished by the 3G codec through the use of

reserved frame values in the 3G packet headers.

Many new low and variable bit rate applications such as picture phone and multiparty

multimedia conferencing are being introduced into mobile telephony. These

applications require guaranteed quality of service. Here AAL2 has been described as a

suitable transport mechanism to carry these traffic types across the network. It not only

achieves high packing efficiency but also low transmission delays. AAL2 has already

been deployed in the 3G networks and can be considered also for the next generation

mobile networks.

REFERENCES


118

References [1]. ITU Recommendation I.150, “B-ISDN asynchronous transfer mode (ATM)

functional characteristics”, Nov. 1993.

[2]. ITU Recommendation I.361, “B-ISDN ATM layer specification”, Nov. 1993.

[3]. ITU Recommendation F.811, “Broadband connection-oriented bearer service”,

Aug. 1992.

[4]. ITU Recommendation F.812, “Broadband connectionless data bearer service”,

Aug. 1992.

[5]. ITU Recommendation I.356, “B-ISDN ATM layer cell transfer performance”,

Feb. 1995.

[6]. F. Halsall, 1996, Data Communications, Computer Networks, and Open

Systems, 4th Edition, Addison Wesley, New York.

[7]. A.K. Parekh, R.G. Gallager, “A Generalized Processor Sharing Approach to

Flow Control in Integrated Services Networks: The Multiple Node Case”,

IEEE/ACM Transactions on Networking, vol. 2, no. 2, pp. 137-150, April 1994.

[8]. S.J. Golestani, “Network Delay Analysis of a Class of Fair Queuing

Algorithms,” IEEE Journal on Selected Areas in Communication (JSAC), vol.

13, no. 6, pp 1057-1070, Aug. 1995.

[9]. D. Stiliadis, A. Varma, “Latency-Rate servers: A General Model for Analysis of

Traffic Scheduling Algorithms”, in Proceedings of IEEE INFOCOM’96, vol. 1,

pp. 111-119, March 1996.

[10]. P. Goyal, H.M. Vin, H. Cheng, “Start-time Fair Queuing: A Scheduling

REFERENCES


119

Algorithm for Integrated Services Packet Switching Networks,” Proceedings of

SIGCOMM’96, 1996.

[11]. ITU Recommendation I.363, “B-ISDN ATM Adaptation Layer (AAL)

Specification”, Mar. 1993.

[12]. P.M. Gopal, J.W. Wong, and J.C. Majithia, “Analysis of playout strategies for

voice transmission using packet switching techniques”, Performance Evaluation

4, North-Holland, pp 11-18, 1984.

[13]. ITU Recommendation I.363.1, “B-ISDN ATM Adaptation Layer Specification:

Type 1 AAL1”, Aug. 1996.

[14]. ITU Recommendation I.363.2, “B-ISDN ATM Adaptation Layer Type 2

specification”, Nov. 2000.

[15]. ITU Recommendation I.366.1, “Segmentation and Reassembly Service Specific

Convergence Sublayer for AAL Type 2,” Jun. 1998.

[16]. ITU-T Recommendation I.366.2, “AAL Type 2 Service Specific Convergence

Sublayer for Narrow-band Services,” Nov. 2000.

[17]. The ATM Forum Technical Committee “ATM Trunking using AAL2 for

Narrowband Services AF-VTOA-0113”, Feb. 1999.

[18]. M. McLoughlin, and J. O’Neil, “A management Briefing on Adapting Voice for

ATM Networks An AAL2 Tutorial”, General DataComm, 1997.

[19]. J. H. Baldwin, B. H. Bharucha, B. T. Doshi, S. Dravida, and S. Nanda, “AAL-2

– A New ATM Adaptation Layer for Small Packet Encapsulation and

Multiplexing”, Bell Labs Technical Journal, vol. 2, no. 2, pp. 111-131, 1997.

[20]. D.W.Petr, R.R.Vatte, and Y.Q.Lu, “Efficiency of AAL2 for Voice Transport:

Simulation Comparison with AAL1 and AAL5”, in Proceedings of IEEE

REFERENCES


120

International Conference on Communications (ICC’99), Vancouver, California,

vol. 2, pp. 896-901, 1999.

[21]. N. Gerlich, and M. Ritter, “Carrying CDMA Traffic over ATM Using AAL2 A

Performance Study”, Report no. 188, University of Wũrzburg, Apr. 1998.

[22]. N. Gerlich, and M. Menth, “The Performance of AAL-2 Carrying CDMA Voice

Traffic”, in 11th ITC Specialist Seminar Yokohama, pp. 17-24, 1998.

[23]. X. Liu, S. Hu, and W. Wu, “QoS Simulation of AAL2 in CDMA/ATM Based

Mobile Systems”, in Proceedings of the International Conference on

Communication Technology (WCT-ICCT’00), vol. 1, pp. 884-887, 2000.

[24]. G. Eneroth, G. Fodor, G. Leijonhufvud, A. Racz, and I. Szabo, “Applying

ATM/AAL2 as a switching technology in third-generation mobile access

network”, IEEE Communication Magazine, vol. 37, no. 6, pp. 112-122, 1999.

[25]. J. H. Chung, Y. H. Kwon, K. H. Cho, and D. K. Sung, “Performance Evaluation

of an AAL2 Link Transmission Scheme for Voice and Data Packets in BS-BSC

Links”, in 52nd IEEE Vehicular Technology Conference (VTC’00), vol. 4, pp.

1610-1614, 2000.

[26]. S. Nananukul, Y. Guo, M. Holma, and S. Kekki, “Some Issues in Performance

and Design of the ATM/AAL2 Transport in the UTRAN”, in IEEE Wireless

Communications and Networking Conference (WCNC’00), vol. 2, pp. 736-741,

2000.

[27]. L. V. Gonzalez, S. Tsakiridou, L. O. Barbosa, and L. Lamont, “Performance

Analysis of Wireless ATM/AAL2 Over A Burst Error Channel”, in Canadian

Conference on Electrical and Computer Engineering (CCECN’01), vol. 1, pp.

703-708, 2001.

[28]. G. Fodor, G. Leijonhufvud, S. Malomsoky, and A. Racz, “Comparison of Call

REFERENCES


121

Admission Control Algorithms in ATM/AAL2 Based 3rd Generation Mobile

Access Networks”, IEEE Wireless Communications and Networking Conference

(WCNC ’99), vol. 3, pp. 1508-1512, 1999.

[29]. W. Brunnbauer and G. Cichon, “Bringing two worlds together: AAL2 over IP

for Radio Access Networks”, IEEE Global Telecommunications Conference

(GlobeCom ’01), vol. 4, pp. 2606-2610, 2001.

[30]. B. Subbiah, and Y. Raivio, “Transport Architecture for UMTS/IMT-2000

Cellular Networks”, in Proceedings of the IEEE International Conference on

Performance, Computing and Communications (IPCCC’00), pp. 208-214, 2000.

[31]. I. Szabo, S. Szekely, and I. Moldovan, “Performance Optimisation of AAL2

Signalling for Supporting Soft Handoffs in UMTS Terrestrial Radio Access

Networks”, in Proceedings of the Fifth IEEE Symposium on Computers and

Communications (ISCC’00), pp. 46-51, 2000.

[32]. O. Isnard, J. M. Calmel, A. L. Beylot, and G. Pujolle, “Handling Traffic Classes

at AAL2/ATM Layer over the Logical Interfaces of the UMTS Terrestrial Radio

Access Network”, in The 11th IEEE International Symposium on Personal,

Indoor and Mobile Radio Communications (PIMRC’00), vol. 2, pp. 1464-1468,

2000.

[33]. S. Jiang, Q. Ding, and M. Jin, “Flexible IP Encapsulation for IP over ATM with

ATM Shortcuts”, in Proceedings of IEEE International Conference on Networks

(ICON’00), pp. 238-242, 2000.

[34]. K. Sriram, and Y. T. Wang, “Voice over ATM Using AAL2 and Bit Dropping:

Performance and Call Admission Control”, IEEE Journal on Selected Areas in

Communication (JSAC), vol. 17, no. 1, pp. 18-28, Jan. 1999.

[35]. J. M. Ahn, and S. W. Seo, “Voice Over AAL2 System Using the Shared-Buffer

Scheme”, in Joint 4th IEEE International Conference on ATM and High Speed

REFERENCES


122

Intelligent Internet Symposium 2001 (ICATM’01), pp. 260-264, 2001.

[36]. C. Voo, “Performance of Statistical Multiplexed Voice over ATM using AAL2

and Deterministic Bit Dropping”, Inter-University Postgraduate Electrical

Engineering Symposium (IUPEES’00), pp. 71-74, July 2000.

[37]. C. Voo and J. F. Siliquini, “Performance Comparison of Multiplexing Methods

for Voice over ATM using AAL2”, in Proceedings of the 9th IEEE

International Conference on Telecommunications (ICT’02), vol. 1, pp. 593-597,

June, 2002.

[38]. G. Mercankosk, J. F. Siliquini, and Z. L. Budrikis, “Provision of Real-time

Services over ATM using AAL type 2”, in ACM First International Workshop

on Wireless Mobile Multimedia (WOWMOM’98), Dallas, Texas, Oct. 1998.

[39]. C. Liu, S. Munir, R. Jain, and S. Dixit, “Packing Density of Voice Trunking

Using AAL2”, IEEE Global Telecommunications Conference (GlobeCom ’99),

vol. 1b, pp. 611-615, 1999.

[40]. K. Zhang, “Packet Delay Variation in Voice Trunking Using AAL2”, ATM

Forum, Doc. 98-0630, Oct. 4-9, 1998.

[41]. T. Okutani, H. Watanabe, and T. Nisase, “Performance Evaluation of

Multiplexing AAL2 Voice Traffic and TCP/IP Data at the ATM Cell Level”,

IEEE Proceedings on ATM Workshop, pp. 391-396, 1999.

[42]. S. H. Jeong, D. K. Hsing, T. H. Wu, and J. A. Copeland, “Trunking Performance

Analysis of AAL2 for VBR Voice Transport”, in IEEE International Conference

on Communications (ICC’99), vol. 2, pp. 886-890, 1999.

[43]. H. Saito, “Bandwidth Management for AAL2 Traffic”, IEEE Transactions on

Vehicular Technology, vol. 49, no. 4, pp. 1364-1367, Jul. 2000.

REFERENCES


123

[44]. K. Sriram, T. G. Lyons, and Y. T. Wang, “Anomalies Due to Delay and Loss in

AAL2 Packet Voice Systems: Performance Models and Methods of Mitigation”,

in IEEE Journal on Selected Areas in Communication (JSAC), vol. 17, no. 1, pp.

109-123, Jan. 1999.

[45]. J. C. Wu, C. H. Huang, and R. T. Sheu, “Performance Study of AAL2 Protocol

for Low Bit-Rate Multimedia Services”, in Proceedings of the 15th

International Conference on Information Networking (IN’01), pp. 793-798,

2001.

[46]. Y. Kitamura, K. Nagato, S. Yasuda, and T. Toriyama, “Implementation of

AAL2 for Low Bit-Rate Voice over ATM”, in Proceedings of XVI World

Telecom Congress, Belgium, pp. 271-276, Fall 1997.

[47]. B. Subbiah, and S. Dixit, “ATM adaptation layer 2 (AAL2) for low bit rate

speech and data: issues and challenges”, in Proceedings of IEEE ATM

Workshop, pp. 225-233, 1998.

[48]. J. Holler, “Voice and Telephony Networking over ATM”, in Ericsson Review,

no. 1, pp. 40-45, 1998.

[49]. E. Lau, Real-Time Signal Transfer over IP Networks, Undergraduate thesis,

University of Western Australia, 2001.

[50]. K.Sriram and D.M.Lucantoni, “Traffic Smoothing Effects of Bit Dropping in a

Packet Voice Multiplexer”, IEEE Journal on Selected Areas in Communication

(JSAC), vol. 37, no. 7, pp 703-712, Jul. 1989.

[51]. M.Butto, E.Cavallero and A.Tonietti, “Effectiveness of the “Leaky Bucket”

Policing Mechanism in ATM Networks”, IEEE Journal on Selected Areas in

Communication (JSAC), vol. 9, no. 3, pp 335-342, April 1991.

[52]. F.Beritelli, A.Lomberdo, S.Palazzo and G.Schembra, “Performance Analysis of

REFERENCES


124

an ATM Multiplexer Loaded with VBR Traffic Generated by Multimode

Speech Coders”, IEEE Journal on Selected Areas in Communication (JSAC),

vol. 17, no. 1, Jan. 1999.

[53]. M.E.Perkins, et al., “Characterizing the subjective performance of the ITU-T 8

kb/s speech coding algorithm – ITU-T G.729”, IEEE Communications

Magazine, vol. 35, pp. 74-81, Sept. 1997.

[54]. H.Lee and C.Un, “A study of an on-off characterization of conversational

approach”, IEEE Transactions on Communications, vol. 34, no. 6, 1986.

[55]. Paul T. Brady, “A model for generating ON-OFF speech patterns in two-way

conversations”, Bell System Technical Journal, vol. 48, pp 2445-2472, Sept.

1969.

[56]. K. Sohraby, “On the Theory of General ON-OFF Sources with Applications in

High-Speed Networks”, INFOCOM Proceedings, Twelfth Annual Joint

Conference of the IEEE Computer and Communications Societies, Networking:

Foundation of the Future, IEEE, vol. 2, pp. 401-410, 1993.

[57]. ITU-T Recommendation G.723.1, “Annex A: Silence compression scheme”,

Nov. 1996.

[58]. ITU-T Recommendation G.109, “Definitions of Categories of Speech

Transmission Quality,” Sept. 1999.

[59]. D. D. Vleeschauwer, J. Janssen, and G. H. Petit, “Delay bounds for low bit rate

voice transport over IP networks”, in Proceedings of the SPIE conference on

Performance and Control of Network Systems III, vol. 3841, pp. 40-48, Boston,

Sept. 1999.

[60]. D. D. Vleeshauwer, J. Janssen, G. H. Petit and F. Poppe, “Quality bounds for

packetized voice transport”, Alcatel Telecommunications Review, 1st Quarter

REFERENCES


125

2000.

[61]. N. O. Johannesson, “The ETSI Computation Model: A Tool for Transmission

Planning of Telephone Networks”, IEEE Communications Magazine, Jan. 1997.

[62]. Emodel, [Homepage of ETSI], [Online]. 2001, Oct. 29-last update.

Available:http://www.etsi.org/stq/presentations/emodel.htm

[63]. V.R. Karanam, K. Sriram, and D.O. Bowker, “Performance Evaluation of

Variable-Bit-Rate voice in Packet-Switched Networks”, AT&T Technical

Journal, vol. 67, no. 5, pp. 41-56, Sept.-Oct. 1988.

[64]. A.L. Garcia, “Probability and Random Processes for Electrical Engineering”, 2nd

ed. Reading, Addison-Wesley, May 1994.

[65]. C. Voo, J. F. Siliquini, and G. Mercankosk, “Service differentiation of variable

bit rate voice in AAL2 multiplexers”, in Proceedings of IEEE Region 10

International Conference on Electrical and Electronic Technology

(TENCON’01), vol. 2, pp. 631-635, 2001.

[66]. H. Heffes and D. M. Lucantoni, “A Markov Modulated Characterisation of

Packetized Voice and Data Traffic and Related Statistical Multiplexer

Performance”, IEEE Journal on Selected Areas in Communication, vol. 4, no. 6,

pp 856-867, Sept. 1986.

[67]. ITU-T Recommendation G.723.1, “Dual rate speech coder for multimedia

communications transmitting at 5.3 and 6.3 kbit/s”, Mar. 1996.

[68]. ITU-T Recommendation G.729, “Conjugate structure-algebraic code-excited

linear prediction,” 1996.

[69]. C. Voo, J. F. Siliquini and G. Mercankosk, “Performance of AAL Type 2 Voice

Multiplexers”, Proceedings of the 9th IEEE International Conference on

REFERENCES


126


[70]. E. P. Rathgeb, “Modeling and Performance Comparison of Policing

Mechanisms for ATM Networks”, IEEE Journal on Selected Areas in

Communication (JSAC), vol. 9, no. 3, pp. 325-334, April 1991.

[71]. D. W. Petr, G. K. Vaddi, and Y. Q. Lu, “UPC Parameter Estimation Using

Virtual Buffer Measurement with Application to AAL2 Traffic”, in Proceedings

of the IEEE Global Telecommunications Conference (Globecom’99), vol. 2, pp.

1373-1379, 1999.

[72]. W. Jiang, and H. Schulzrinne, “Analysis of On-Off Patterns in VoIP and Their

Effect on Voice Traffic Aggregation”, in Proceedings of the Ninth International

Conference on Computers Communications and Networks (ICCCN’00), pp. 82-

87, 2000.

[73]. D. A. Hughes, G. Anido, H. S. Bradlow, and S. Tan, “On Average Rate

Prediction and Enforcement in B-ISDN”, in A.T.R, vol. 26, no. 2, pp. 11-19,

1992.

[74]. J. F. Siliquini, G. Mercankosk, S. Ivandich, C. Voo, Z. L. Budrikis, and A.

Cantoni, “On Statistical Multiplexing Gain for Variable Bit Rate Voice

Sources”, in Proceedings of the 8th IEEE International Conference on


[75]. 3GPP TS 26.201, “AMR speech codec, Wideband; Frame structure,” Feb. 2001.

[76]. C. Voo, “A Review of the New Adaptation Layer Type 2”, Inter-University

Postgraduate Electrical Engineering Symposium (IUPEES’99), pp. 17-18, July

1999.

APPENDIX A IMPLEMENATION OF THE DBR AND SBR CELL DISPATCH PROCESSES


127

Appendix A

Implementation of the DBR and SBR Cell Dispatch Processes In this appendix, the DBR and the SBR cell dispatch processes as well as the AAL2

buffer management processes are described. The models for the DBR and SBR buffer

management and cell dispatch processes are shown in Figure A.1.

AAL2 Buffer DBR cell dispatch

AAL2 Buffer

SBR cell dispatch

Figure A.1: Model for DBR and SBR ATC incorporating AAL2

The AAL2 buffer in Figure A.1 for both the DBR and the SBR ATC is implemented

using a FCFS ring buffer of size N. The buffer management process is shown in Figure

A.2. In the ring buffer, a start pointer denoted by S and a finish pointer denoted by F

have been assigned to manage the buffer fill. The finish pointer is incremented only

when AAL2 packets are inserted into the buffer and the start pointer is incremented only

when AAL2 packets are taken from the buffer.



128

start

Set S, F = 0

Wait for packet

Packet payload arrives

Space left = N – ModN(F - S)

Packet size = Payload size + AAL2 header

Packet size > Space left?

Prepend AAL2 header Insert AAL2 packet to buffer

Discard packet payload

Point A

Yes

No

Packet policing task

Point B

F = ModN(F + Packet size)

Updating task

Figure A.2: AAL2 Buffer Management process.

Referring to Figure A.2, initialisation of the buffer pointers S and F are performed at the



129

start of the operation. This process then waits for an incoming packet to arrive from the

source. When a packet arrives, it determines via the packet policing task if the packet

conforms to its call connection parameters. The functions of the packet policing task are

shown in Figure A.3.

Get current time tb Get TAT[CID]

Get T[CID] Get τ[CID]

No tb < TAT[CID] – τ[CID]?

Point B Discard Packet

Point A

Yes

Figure A.3: Packet policing task.

Referring to Figure A.3, the packet policing task obtains the current time tb, the

theoretical arrival time of a packet TAT[CID], the packet interarrival time T[CID] and

the tolerance τ[CID] of a connection specified by CID. Packets that arrived are policed.

Non-conforming packets are discarded and the packet policing task returns via Point A

to wait for the next packet to arrive. Note that in practice, the time value has a finite

number storage and will overlap. Therefore further tests are required before the packet

can really be discarded. An example of such a test is to compare previous TAT[CID]

with the current time. If previous TAT[CID] is smaller than the current time, then the

current time has overlap.



130

However for conforming packets, the AAL2 buffer management process then proceeds

to determine the amount of available space in the ring buffer by subtracting the buffer

fill, given by the modulo term i.e. ModN(F – S) from the size of the buffer (i.e. N) and

compares this to the packet size that will be inserted into the buffer. If there is sufficient

buffer space, the packet is first prepended with an AAL2 header before being inserted

and the finish pointer is incremented by an AAL2 packet size. The next theoretical

arrival time to be used in the packet policing task is then updated in the updating task.

The functions of the updating task are shown in Figure A.4.

Get current time tb Get TAT[CID] Get T[CID]

tb > TAT[CID]?

Point A

TAT[CID] = tb + T[CID]

Yes No

TAT[CID] = TAT[CID] + T[CID]

Figure A.4: Updating task.

Referring to Figure A.4, upon completion of the updating task, it returns via Point A to

wait for the next incoming packet. In the case where there is insufficient AAL2 buffer

space, the packet is discarded and the theoretical arrival time is not updated.



131

DBR Cell Dispatch Process

The DBR cell dispatch process is shown in Figure A.5.

start

Wait for send signal

Send signal arrives

Buffer fill = ModN(F - S)

Buffer fill ≥ 47?

Take buffer fill octets as AAL2 CPS payload Prepend CPS header Send AAL2 CPS cell

Set S = F

Take 47 octets as AAL2 CPS payload

Prepend CPS header Send AAL2 CPS cell

Buffer fill > 0? No

Yes

Yes

No

Set S = ModN(S + 47)

Figure A.5: DBR cell dispatch process.



132

Referring to Figure A.5, at the start of this process, it waits for the send signal which is

generated at the Peak Cell Rate (1/PCR) of the DBR connection. When the send signal

arrives, the buffer fill is determined. If the buffer is empty, no cells are sent, and the

process then waits for the next send signal to arrive. However, if the buffer is non-

empty, it then determines if a full AAL2 CPS cell payload (i.e. 47 octets) can be

created.

In the case where a full cell payload can be created, 47 octets are taken from the buffer

and prepended with an AAL2 CPS header before being sent. The start pointer is then

incremented by 47 octets. In the case of a partially filled payload, the remaining

available spaces are padded with null values. In this case, the start pointer is

incremented to the finish pointer.

SBR Cell Dispatch Process

For the SBR cell dispatch process, this is shown in Figure A.6. In the SBR cell dispatch

process, an AAL2 CPS PDU cell is created on two conditions; when the fill timer

denoted by FillTimer has expired or when a full CPS PDU cell payload can be created.



133

start

Wait for send signal

Send signal arrives

Buffer fill = ModN(F - S)

Buffer fill ≥ 47?

Timer taskConformance test

Set SetTime, TATPCR, TATSCR to 0

Set TimeSet to false

Buffer fill > 0?

Yes

Yes

No

No

Get current time ta

Update task

Point A

Figure A.6: SBR cell dispatch process.



134

Referring to Figure A.6, at the start of the SBR cell dispatch process, the theoretical

arrival times for the peak cell rate denoted by (TATPCR) and the sustainable cell rate

denoted by (TATSCR) as well as the wait time denoted by SetTime is initialised to a value

of 0. The logic TimeSet which is used to indicate if the fill timer has been set is

initialised to false. This process then waits for the send signal which arrives on the cell

service time instants of the line rate. When the send signal arrives, the buffer fill is

determined. If the buffer is empty, no cells are sent, and the process then waits for the

next send signal to arrive. However, if the buffer is non-empty, it gets the current time

denoted by ta and then determines if a full AAL2 CPS cell payload (i.e. 47 octets) can

be created.

For the case where a full AAL2 CPS payload can be created, the conformance test

and/or the update task will be carried out. In the case where there is insufficient data to

fill the AAL2 CPS payload, the timer task is carried out. These tasks are examined next,

beginning with the timer task.

The functions in the timer task are shown in Figure A.7.



135

YesTimeSet = true ?

No SetTime =

ta +FillTimer SetTime ≤

ta?

Take buffer fill octets as AAL2 CPS payload Prepend CPS header Send AAL2 CPS cell

No

TATPCR = ta + 1/PCR TATSCR = ta + 1/SCR

YesTimeSet = true

Set S = F

TimeSet = false

Point A

Figure A.7: Functionality of SBR Timer Task.

Referring to Figure A.7, the logic TimerSet is examined to check if the fill timer has

already been set. If this has not been set, TimerSet is then set to true and SetTime is set

to expire after a period of time given by ta + FillTimer. However for the case where the

fill timer exists, the timer task then checks to examine if it has expired (i.e. when

SetTime delay is equal to or smaller than the current time ta). In the case where it has

expired, all the contents in the buffer is taken out and used to fill the AAL2 CPS



136

payload, and the remaining available spaces padded with null values. The theoretical

arrival times of the peak cell rate and the sustainable cell rate (i.e. TATPCR, TATSCR) are

then updated. Also the S pointer is incremented to the F pointer and the logic TimeSet is

set as false. The AAL2 CPS cell is sent and the timer task returns via Point A to wait for

the next send signal to arrive. For the case where the fill delay has not expired, no

operations are performed and the timer task also returns via Point A to await the next

incoming send signal.

The functions for the conformance test are shown in Figure A.8.

No ta < TATPCR

-τCDV?

Yesta < TATSCR

- τIBTSBR?

Take 47 octets as AAL2 CPS payload Prepend CPS header Send AAL2 CPS cell

Set S = ModN(S + 47)

Yes

No Point A

Update task

Figure A.8: Functionality of the SBR Conformance Task.

Referring to Figure A.8, the AAL2 CPS PDU cell is tested for both Peak Cell Rate



137

(PCR) and Sustainable Cell Rate (SCR) conformance. Note that the peak cell rate in the

conformance test is much smaller than the arrival rate of the send signal. If the cell fails

either condition, it is not sent and the conformance test returns via Point A to wait for

the next send signal to arrive. In the case where the cell passes both conformance tests,

47 octets are taken out of the buffer as AAL2 CPS payload and prepended with an

AAL2 CPS header before being sent. The S pointer is then incremented by 47 octets and

the theoretical arrival times of the peak cell rate and the sustainable cell rate for the next

conformance test are updated in Update Task.

The functions of the Update Task are shown in Figure A.9.



138

ta > TATPCR

?Yes

ta > TATSCR ?

No

No

Yes

Point A

TATPCR = ta + 1/PCR TATPCR = TATPCR + 1/PCR

TATSCR = ta + 1/SCR TATSCR = TATSCR + 1/SCR

TimeSet = false

Figure A.9: Functionality of the SBR Update Task.

Referring to Figure A.9, future TATs to be used in the conformance task are updated.

The logic TimeSet is set as false and the Update Task returns via Point A to wait for the

next send signal to arrive.

APPENDIX BAN ANALYSIS ESTABLISHING THE EQUIVALENCE BETWEEN DBR AND SBR ATC


139

Appendix B

An Analysis Establishing the Equivalence between DBR and SBR ATCs In this appendix, an analysis establishing the equivalence between DBR and SBR ATCs

is presented. This is based on the model shown in Figure B.1. Referring to this figure,

cells are presented to both the spacer and the policer at exactly the same time with an

arbitrary distribution {tk} where tk is defined as the time in which the last bit of the kth

cell is presented to the spacer and the policer.

DBR ATC Peak cell rate, PCRDBR

Spacer

{ξk}

Policer

Incoming Cells {tk}

LT SBR ATC Traffic conforming to PCRSBR, SCRDBR, and τIBT SBR

τIBT SBR

Figure B.1: DBR and SBR ATC model.

In the DBR ATC, when a cell enters the spacer, the maximum time a cell spends

waiting in the spacer is bounded by the buffer size, LT (cells). Note that LT is related to



140

the value of Dspacer (given by (B.1) described in Section 1.2.3.1.2) shown as

Tspacer

DBR

LDPCR

= (B.1)

where PCRDBR is defined as the Peak Cell Rate of the DBR connection.

The kth cell leaves the spacer at time ξk where ξk is defined as the instant at which the

last bit of the kth cell leaves the spacer. For the arrival of the (k+1)th cell to the spacer, if

the queue is empty, it is immediately served on the next cell slot. However, if the queue

is not empty, then the (k+1)th cell joins the end of the queue and waits until the kth cell is

served before departing from the spacer. Therefore the departure time for the (k+1)th cell

is

1 11max ,k k k

DBR

tPCR

ξ ξ+ +

⎛ ⎞= +⎜ ⎟

⎝ ⎠

(B.2)

When a cell on arrival to the spacer finds the queue full, it is discarded by the spacer. A

busy period is defined as the maximal interval of time during which the spacer is never

idle.

For the SBR ATC, cells that enter the policer are policed according to the Generic Cell

Rate Algorithm (GCRA (PCRSBR, SCRSBR, τIBT SBR)). The GCRA (PCRSBR, SCRSBR, τIBT

SBR) is an algorithm that uses the Theoretical Arrival Time (TAT) and τIBT SBR to

determine if a cell is conforming. Let TATk be the nominal arrival of the kth cell

assuming cells are sent equally spaced at the sustainable cell rate, SCRSBR. For the kth

cell to be conforming, its arrival time must be in the range TATk-τIBT SBR ≤ tk. A non-

conforming kth cell has an arrival time that is smaller than (TATk-τIBT SBR). When the kth

cell is found conforming, TAT is reset to tk and then updated to (tk+1/SCRSBR).

Conforming cells leave the policer immediately upon arrival while non-conforming

cells are discarded by the policer.



141

The analysis presented here for ATM cells is adapted from Lau’s work in [49] where he

showed the equivalence in performance between DBR and SBR for Internet packets.

Using the model of Figure B.1, let us begin the analysis.

The following variables are defined as follows:

Let tk be the arrival time of the kth cell into either the spacer (DBR) or the policer (SBR).

It also means that k number of cells have arrived into the spacer.

Let ξk be the departure time of the kth cell from the spacer (DBR).

Let J denote the start of a busy period. For the spacer, the start of the Jth busy period

occurs on the arrival of the very first cell of that period, and the end of the Jth busy

period is when the last cell in the spacer departs. For the policer, the start of the Jth busy

period occurs on the arrival of the very first cell of that period, or when a cell arrives

late relative to its TAT and the end of the Jth busy period is when a cell does not arrive

at or before its TAT. It is assumed that the Jth busy period in the spacer and the policer

coincides with the arrival of cell b at time tb.

Let PCRDBR be defined as the peak cell rate at which the spacer outputs cells.

In the analysis, the instant at which a cell is accepted or discarded by the spacer due to

input queue overflow, and the conditions where a cell is found conforming or non-

conforming by the policer are established. It will be shown that under all conditions

when a cell is discarded by the spacer, it is also found non-conforming by the policer.

Similarly, when the spacer accepts a cell, the same cell is also found conforming by the

policer.

For the DBR ATC:

A cell is discarded when the buffer fill LT is reached or exceeded. Let the kth cell be the

first cell to be discarded in the Jth busy period (i.e. in the period (tb, tk)) when upon

arrival it finds the spacer full. This will occur when the number of arrivals in the period



142

(tb, tk) denoted by A(tb, tk) subtract the number of departures in the same period denoted

by D(tb, tk) exceeds the spacer buffer size by one.

( ) ( ), , 1b k b k TA t t D t t L− = + (B.3)

Using the previous definitions for the time at which cells are presented to the spacer, at

time tk, k number of cells have been presented to the spacer. Similarly at time tb, b

number of cells have arrived at the spacer. Therefore the number of arrivals in the

period (tb,tk) (i.e. A(tb,tk)) is equivalent to

( )b kt , t 1A k b= − + (B.4)

The number of departures is equivalent to the number of cells served by the spacer.

Therefore in the period (tb,tk), the number of departures D(tb,tk) is

( ) ( ),b k k b DBRD t t t t PCR 1= − × +⎢ ⎥⎣ ⎦ (B.5)

Substituting (B.4) and (B.5) into (B.3), the condition for the kth cell being the first cell

discarded in the Jth busy period is

( ) ( ){ } ( ){ }

( )

, , 1

1 1 1

1

b k b k T

k b DBR T

k b DBR T

A t t D t t L

k b t t PCR L

k t t PCR L b

− = +

− + − − × + = +⎢ ⎥⎣ ⎦

= − × + + +⎢ ⎥⎣ ⎦

(B.6)

Using the identity that ⎣yx⎦ + n = ⎣x(y + n/x)⎦, (B.6) can be re-written as

( ) 1TDBR k b

DBR

Lk PCR t t bPCR

⎢ ⎛ ⎞= × + − +⎢ ⎥⎜ ⎟

⎝ ⎠⎣ ⎦

⎥+

(B.7)

Equation (B.7) shows the case where the kth cell is the first cell to be discarded by the

spacer during the Jth busy period in (tb,tk). Similarly, for the case where the kth cell is

accepted into the spacer, the number of arrivals in the period (tb,tk) subtract the number



143

of departures in the same period must be equal to or smaller than the spacer buffer size.

( ) ( ), ,b k b k TA t t D t t L− ≤ (B.8)

By substituting (B.4) and (B.5) into (B.8), the condition of kth cell being accepted in the

Jth busy period is

( ) ( ){ } ( ){ }

( )

, ,

1 1b k b k T

k b DBR T

k b DBR T

A t t D t t L

k b t t PCR L

k t t PCR L

− ≤

− + − − × + ≤⎢ ⎥⎣ ⎦

b≤ − × + +⎢ ⎥⎣ ⎦

(B.9)

Using the identity that ⎣yx⎦ + n = ⎣x(y + n/x)⎦, (B.9) can be re-written as

( )TDBR k b

DBR

Lk PCR t tPCR

⎢ ⎛ ⎞≤ × + −⎢ ⎥⎜ ⎟

⎝ ⎠⎣ ⎦b

⎥+

(B.10)

Equation (B.10) shows the case where the kth cell is accepted by the spacer during the

Jth busy period. The upper bound of (B.10) gives the condition that the kth cell is the last

packet to be accepted by the spacer is shown as

( )TDBR k b

DBR

Lk PCR t tPCR

⎢ ⎛ ⎞= × + −⎢ ⎥⎜ ⎟

⎝ ⎠⎣ ⎦b

⎥+

(B.11)

For the SBR ATC:

The TATk for the arrival of the kth cell in the Jth busy period that commences at time tb is

( )k b

SBR

k bTAT t

SCR−

= + (B.12)

For the kth cell to be found non-conforming by the policer, its arrival time (i.e. tk) must

be earlier than (TATk-τIBT SBR) shown by the condition



144

SBR < k k IBTt TAT τ− (B.13)

Substituting (B.12) into (B.13), this is

k b IBT SSBR

k bt tSCR

τ⎛ ⎞−

< + −⎜ ⎟⎝ ⎠

BR (B.14)

The condition where the kth cell is declared non-conforming by the policer is

( ){ }( ){ }

k b IBT SBRSBR

k b IBT SBRSBR

k b IBT SBR SBR

k b IBT SBR SBR

k bt tSCR

k bt tSCR

t t SCR k b

k t t SCR

τ

τ

τ

τ

⎛ ⎞−< + −⎜ ⎟

⎝ ⎠⎛ ⎞−

− + < ⎜ ⎟⎝ ⎠

− + × < −

> − + × + b

(B.15)

Recognising that k is an integer, the condition of (B.15) is re-written as

( ){ } 1SBR IBT SBR k bk SCR t t bτ⎢= × + − +⎣ ⎥ +⎦

SBR

(B.16)

Equation (B.16) shows the case where the kth cell is the first cell in the Jth busy period to

be declared non-conforming. Similarly for the kth cell to be conforming, the arrival time

of the kth cell (i.e. tk) must be greater than or equal to (TATk-τIBT SBR) shown by

k k IBTt TAT τ≥ − (B.17)

Substituting the value of TATk for the kth cell in the Jth busy period given in (B.12) into

(B.17), this is

k b IBT SSBR

k bt tSCR

τ⎛ ⎞−

≥ + −⎜ ⎟⎝ ⎠

BR (B.18)

The condition where kth cell is declared conforming is



145

( ){ }( ){ }

k b IBT SBRSBR

k b IBT SBRSBR

k b IBT SBR SBR

k b IBT SBR SBR

k bt tSCR

k bt tSCR

t t SCR k b

k t t SCR

τ

τ

τ

τ

⎛ ⎞−≥ + −⎜ ⎟

⎝ ⎠⎛ ⎞−

− + ≥ ⎜ ⎟⎝ ⎠

− + × ≥ −

≤ − + × + b

(B.19)

The upper bound of (B.19) gives the condition that the kth cell is the last packet found

conforming by the policer and is shown as

( ){ } SBR IBT SBR k bk SCR t tτ⎢= × + −⎣ b⎥ +⎦ (B.20)

Equivalence condition between DBR ATC and SBR ATC:

We now compare the conditions between the spacer and the policer in the Jth busy

period for cases where the kth cell is either discarded or accepted. The condition for

which the kth cell is accepted by the spacer in the DBR ATC and also found conforming

by the policer in the SBR ATC (and vice versa) is shown in (B.21) where the SCRSBR is

set equal to the service rate of the PCRDBR.

SBR DBRSCR PCR= (B.21)

For the DBR ATC, when the kth cell is discarded from the spacer due to buffer fill, LT

being reached then all cells that arrived during the busy period before this kth cell will

have been already accepted by the spacer. Note that the spacer performs no other action

when discarding the cell. Therefore on arrival of the next cell, the spacer treats this

recently arrived cell independent of the discarded cell. The conditions for the kth cell

being the first cell discarded or the last cell accepted by the spacer is summarised using

(B.7) and (B.11) respectively as



146

( )

( )

1 Cell Discarded

Cell Accepted

TDBR k b

DBR

TDBR k b

DBR

Lk PCR t t bPCR

Lk PCR t t bPCR

⎢ ⎥⎛ ⎞= × + − + +⎢ ⎥⎜ ⎟

⎝ ⎠⎣ ⎦⎢ ⎥⎧ ⎫

= × + − +⎨ ⎬⎢ ⎥⎩ ⎭⎣ ⎦

(B.22)

Similarly for the SBR ATC, when the kth cell is found non-conforming by the policer,

then all cells that arrived during the busy period before this kth cell will have already

been found conforming. Note that when a cell is found non-conforming, no actions are

performed by the policer (i.e. the TAT is not updated). Therefore on arrival of the next

cell, the policer treats this recently arrived cell independent of the discarded cell. The

conditions for the kth cell being the first cell found non-conforming or the last cell found

conforming by the policer is summarised using (B.16) and (B.20) respectively as

( ){ }( ){ }

1 Non-Conforming cell

Conforming cell

SBR IBT SBR k b

SBR IBT SBR k b

k SCR t t b

k SCR t t b

τ

τ

⎢ ⎥= × + − + +⎣ ⎦⎢ ⎥= × + − +⎣ ⎦

(B.23)

From (B.22), (B.23) and under the condition outlined in (B.21) where PCRDBR equals

SCRSBR, τIBT SBR is related to LT by

T

IBT SBRDBR

LPCR

τ = (B.24)

Alternatively, LT can be obtained by re-arranging (B.24) as

T IBT SBR DBRL PCRτ= × (B.25)

From (B.25), (B.1) can be written in terms of τIBT SBR as



147

Tspacer

DBR

IBT SBR DBR

DBR

IBT SBR

LDPCR

PCRPCR

τ

τ

=

×=

=

(B.26)

From (B.26), it is observed that the value of Dspacer in the DBR ATC equals the intrinsic

burst tolerance (τIBT SBR) of the SBR ATC under the condition outlined in (B.21). This

concludes the analysis for the equivalence between the DBR and the SBR ATCs.

APPENDIX C UPC – SOFTWARE IMPLEMENTATION


148

Appendix C

UPC – Software Implementation In this appendix, a software implementation of the token bucket policer is described. A

test environment has been set up to collect different voice conversation samples. These

are used as sources to test the implemented system.

Test Environment

Conversation samples are taken between two test subjects; one of which is in the

laboratory room. A microphone connected to the computer is set up in the laboratory

room to record a one way conversation (i.e. only the voice of the test subject in the

laboratory room is recorded). Factors (i.e. outside noises and echoes in the room) that

degrade the conversation are kept to a minimum. The encoding and the processing of

these samples are described in the following section

UPC Software Implementation

The software implementation model is shown in Figure C.1. Pre-recorded voice

conversation samples are encoded, silence suppressed and then policed according to a

set of token bucket parameters (Refer to Section 5.3.1).

Silence

suppression Policed traffic Policing Voice Encoding

Figure C.1: Software implementation of policing.



149

Voice Encoding

A voice signal is characterised by both positive and negative voltage amplitudes.

Encoding of a voice signal involves three steps. These are:

• Voice sampled at fixed rate.

• Quantisation of each voice sample into different quantised levels (µ-Law).

• Encoding of quantised levels into distinct binary words.

The conversation samples are encoded in Pulse Code Modulation (PCM). In PCM

encoding, a voice signal example shown in Figure C.2 is to be sampled at 8 kHz (i.e. at

fixed intervals of 0.125ms for a sample). Each sample is converted to a quantised value

in the range of 0 to 255 levels (i.e. 28-1). Once these voice samples have been quantised,

they are encoded into binary numbers representing decimal values in the range –127 to

128. Each encoded sample is an octet long. These encoded samples are then stored in a

Wave (WAV) file format defined by Microsoft. The header of the WAV is shown in

Figure C.3.

Voice Signal

Time

Vol

tage

Lower Threshold Level

Figure C.2: Voice signal sampling.



150

Figure C.3: WAV format header.

Referring to Figure C.3, the WAV header size is around 44 octets. However using the

Microsoft sound tools, an additional 10 octets are included in the header. When

processing the samples, the header fields remain unchanged.

Silence Suppression

Silence suppression requires that silence intervals in the conversation samples be first

detected. From Figure C.2, it is observed that there are multiple zero crossings on the

voltage axis with respect to the time axis. A period is considered silent only when a

minimum number of these samples are within a certain noise threshold (i.e. between

upper and lower noise threshold). Figure C.4 illustrates the process of detecting silence

in a voice conversation. This silence detection process is similar to that in G.723.1 in

[57].

R I F F

RIFF Chunk Length

W A V E

f m t

Format Chunk Length

Format Tag Channels

Sample Rate

Average No. of bytes P/second

Block Align Bits / Sample

d a t a

Data Length

Raw Data



151

Voice Signal

Silence Period

Time

Vol

tage


Figure C.4: Silence detection.

Upon identifying the silence periods in the conversations, these are suppressed using the

algorithm shown in Figure C.5.



152

start

Remove header and set pointer at beginning of file

Is pointer at end of file?

Is sample within noise threshold?

Write sample back to file

Reset counter

Is counter > silence threshold?

Write sample back to file

Yes

Yes

Yes

No

No

No

Read pointer

File is silence suppressed

Get sample Increment pointer

Increment counter

Write sample as nulls back to file

Figure C.5: Silence suppression algorithm.



153

Referring to Figure C.5, the header is first removed from the sound file and a pointer is

set to the beginning of the file. The pointer increments as each PCM sample are read

from the file. The sound file is “silence suppressed” when all samples have been read.

The content in each sample is compared to a noise threshold (as shown in Figure C.4).

When a number of these samples fall within the noise threshold consecutively, a silence

period is detected. A counter is implemented to keep track of these samples. The

detection of silence periods is different to the silence obtained from small pauses in the

speech. It is a result of the person listening on the phone. This is known as hangover

time. This is a technique to avoid sudden clipping of speeches and to bridge short

speech gaps such as those due to stop consonants. The length of the silence threshold is

set approximately as 200ms [57]. When a silence period is detected, samples that

arrived after are written to an output file as null (i.e. zero) values. A silence period ends

upon the detection of a sample with a value outside the noise threshold. Values of the

samples in the talk periods are preserved when they are written into the output file. The

header for the sound file is maintained. The resultant waveform with silence

suppression is shown in Figure C.6.

Voice Signal

Silence Suppressed

Time

Vol

tage

Upper Threshold

Level


Figure C.6: Silence suppressed voice signal.



154

Note that the values of the samples are in the range of 0 to 255 as the data in the file is

read as unsigned character. Given that a voice signal has both positive and negative

voltage amplitudes, positive voltage amplitudes are represented by a value in the range

0 to 128, with 128 representing zero. Any values from 129 to 255 (255 being the

smallest integer) are used to represent the negative voltage amplitudes.

The parameters for the silence detection and suppression algorithm in Figure C.5 are

summarised in Table 17.

Silence suppression parameters Value

Max silence interval 200ms (corresponds to 1600 packets)

Noise threshold 128±5

Table 17: Parameters for silence suppression algorithm.

The next part of the software implementation to be examined is the token bucket

policer. A simple algorithm (similar to the generic leaky bucket algorithm) has been

implemented to police the source traffic originating from pre-recorded samples.

Token Bucket Policing

Source traffic is policed according to the token bucket algorithm shown in Figure C.7.

This is a simplified version of the generic leaky bucket algorithm described in Section

5.3.1. In Figure C.7, only the SPR is policed. The PPR parameter does not need to be

monitored as the packet interarrival time has been fixed at 5ms. This corresponds to 40

PCM samples in a packet where a PCM sample has a length of 0.125ms.



155

Silence suppressed voice

ta < TATSPR - τIBT voice source ? Non-conforming

Packet

No

Yes

Conforming Packet TATSPR = max(ta, TATSPR) + TSPR

Figure C.7: Implementation of token bucket algorithm.

Referring to Figure C.7, the silence-suppressed voice is fed into the policer and the SPR

condition is monitored. The value of SPR is taken from Section 5.3.2 to be 111

packets/sec, hence the value of TSPR used is 1/111 secs. The intrinsic burst tolerance

τIBT voice source is obtained using the range of MBS values shown in Figure 40. For non-

conforming packets, nulls are written into the output sound file for these packets.

ackets.

Management of Low and Variable Bit Rate ATM … › files › 3214883 › Voo...CONTENTS MANAGEMENT...

Documents

Transcript of Management of Low and Variable Bit Rate ATM … › files › 3214883 › Voo...CONTENTS MANAGEMENT...