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VoIP Jitter in 3GPP Long Term Evolution Networks -Edición Única

Title

VoIP Jitter in 3GPP Long Term Evolution Networks -Edición Única

Issue Date

2009-12-01

Publisher

Instituto Tecnológico y de Estudios Superiores de Monterrey

Item Type

Tesis de maestría

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INSTITUTO TECNOLÓGICO Y DE ESTUDIOS SUPERIORES DE MONTERREY

CAMPUS MONTERREY

PROGRAMA DE GRADUADOS EN MECATRÓNICA Y TECNOLOGÍAS DE INFORMACIÓN

INSTITUTO TECNOLÓGICO Y DE ESTUDIOS SUPERIORES DE MONTERREY CAMPUS MONTERREY PROGRAMA DE GRADUADOS EN MECATRÓNICA Y

VoIP Jitter in 3GPP Long Term Evolution Networks

by

Christian Alberto Rodríguez García

Thesis

Presented as a partial fulfillment of the requirements for the degree of

Master of Science in Electronic Engineering Major in Telecommunications

Monterrey, N.L. December 2009

Instituto Tecnológico y de Estudios Superiores de Monterrey

Campus Monterrey

División de Mecatrónica y Tecnologías de Información Programa de Graduados

The members of the thesis committee hereby approve the thesis of Christian Alberto Rodríguez García, B.S. as a partial fulfillment of the requirements for the degree of Master of Science in:

Electronic Engineering Major in Telecommunications

Thesis Committee:

___________________________ David Muñoz Rodríguez, Ph.D.

Thesis Advisor

___________________________ César Vargas Rosales, Ph.D.

Synodal

___________________________ Gabriel Campuzano Treviño, Ph.D.

Synodal

___________________________ Joaquín Acevedo Mascarúa, Ph.D.

Director of the Graduate Program

December 2009

To my family,

Cristina García González, Juan José Rodríguez Uc and Karla Brisol Rodríguez García

ACKNOWLEDGMENTS

This work is devoted with affection to my parents, María Cristina and Juan José, for their unconditional support along my life. Without your love, guidance, and comprehension, I had never made it - thanks will never suffice. To my dear sister, Karla Brisol, for being more than my best friend. To all and every one of my family members, especially to my grandfathers, Trinidad González Anaya, Lucino García Ochoa, María Elena Catzín and Eladio Rodríguez for being an inspiration in my life. To my family in heaven that always encouraged me.

I would like to express my gratitude to my thesis advisor David Muñoz Rodríguez, Ph.D. for his professional advice. Because, without his guidance this thesis would not have been possible. I also want to thank to my professor, and

friend, Alejandro Aragón Zavala, Ph.D. for instilling telecommunications.

in

me

the

passion

for

Finally but not the last, I want to thank God for giving me strength, patience, and the wonderful opportunity to be alive.

Christian Alberto Rodríguez García December 2009

V

VoIP Jitter in 3GPP Long Term Evolution Networks

Christian Alberto Rodríguez García, B.S.

INSTITUTO TECNOLÓGICO Y DE ESTUDIOS SUPERIORES DE MONTERREY, 2009

Thesis advisor: David Muñoz Rodríguez, Ph.D.

3GPP LTE is the next step towards 4G mobile communications with performance comparable to wire-line networks. Careful planning and design must be carried out to assure a successful deployment for both, users and network operators. LTE must be able to adapt to a variety of traffic such as data, voice, and video. Services currently provided through circuit-switched systems are expected to have a similar equivalent in LTE, an all-IP based network.

Voice is the most widespread service and represents the main revenue for network operators. Users expect at least the same Quality of Service provided by CS networks while operators look for an increase in capacity and reduction of costs. Such objectives can be reached through Voice-over-IP (VoIP). This service has the following characteristics: Bursty low bitrate traffic, strict packet delay-based QoS, and a high number of simultaneous users. Furthermore, VoIP is highly sensible to jitter.

Jitter is a common issue in packet-switched networks, where packets arrive at random times at the receiver. For voice services, it implies disruptions in speech intelligibility and poor QoS. This master thesis studies the jitter phenomenon for the LTE downlink, where bottlenecks arise naturally due to user queues, when it is operated under VoIP traffic. Specifically the impact on jitter due to network congestion, retransmissions and different modulation and coding schemes, for diverse radio channel conditions, are analyzed. Abstract

CONTENTS

Acknowledgments

V

List of Figures

X

List

Of Tables

XII

Chapter 1 Introduction

1

1.1

Problem Description

2

1.2

Objective

3

1.3

Justification

3

1.4

Contribution

3

1.5

Thesis Organization

4

Chapter 2 3GPP Long Term Evolution

5

2.1

The standardization process

5

2.2

Design Targets

6

2.3

Architecture

7

2.4

LTE Physical Layer

9

2.4.1

Bandwidths, frequency bands and duplexing

............................. 9

2.4.2

OFDM

11

2.5

Physical

Resources

13

2.6

Physical

signals

15

2.6.1

Cell-Specific Downlink Reference Signals

15

2.6.2

Synchronization Signals

16

2.6.3

Downlink L1/L2 control signaling

16

2.7

Link Adaptation

17

2.7.1

Modulation and Coding

18

2.8

Scheduler

19

VII

  • 2.8.1 Channel-status reports

20

  • 2.9 Hybrid-ARQ

21

Chapter 3 Voice-over-IP

23

  • 3.1 VoIP Codecs

..................................................................................

23

  • 3.2 Quality Criteria ................................................................................

25

  • 3.3 VoIP traffic model

...........................................................................

26

  • 3.4 Generating VoIP traffic

28

  • 3.5 VoIP Traffic Simulator

29

Chapter 4 Introduction to Jitter

31

  • 4.1 The Jitter Concept

31

  • 4.2 LTE Jitter Sources

32

  • 4.2.1 Scheduler Buffer .......................................................................

32

  • 4.2.2 HARQ retransmissions

33

  • 4.2.3 Radio Link Control Functions

34

  • 4.2.4 Mobility

.....................................................................................

35

  • 4.2.5 Other Jitter Sources

35

  • 4.3 Jitter management ..........................................................................

35

  • 4.3.1 Jitter buffer

35

  • 4.3.2 Scheduler strategies .................................................................

36

Chapter 5 VoIP Jitter in LTE

37

  • 5.1 Simulation Scenario

.......................................................................

37

  • 5.2 Simulation

Description ....................................................................

39

  • 5.2.1 Physical layer ....................................................................

LTE

39

  • 5.2.2 MAC Protocol ....................................................................

LTE

41

  • 5.3 Simulation results

47

  • 5.3.1 ..............................................................................

SNR = 5 dB

48

  • 5.3.2 ..............................................................................

SNR = 8 dB

50

  • 5.3.3 ............................................................................

SNR = 12 dB

51

Chapter 6 Conclusions and Future Work

59

  • 6.1 General Conclusions

59

  • 6.2 Future Work ....................................................................................

60

VIII

 

Appendix A

Multi-carrier transmission

61

A.1

The Muti-carrier Concept

61

A.1.1

61

A.1.2

Wider

63

A.2

Multi-carrier transmission

64

A.3

OFDM as a multi-carrier transmission

64

A.3.1

66

A.3.2

A.3.3

Cyclic-Prefix

67

OFDM Subcarrier Spacing

68

A.3.4

Number of subcarriers ..............................................................

69

Vita

76

IX

LIST OF FIGURES

Figure 2.1: LTE Architecture

8

Figure 2.2: DL LTE protocols

9

Figure 2.3: FDD and TDD ...........................................................................

10

Figure 2.4: OFDM concepts

11

Figure 2.5 OFDM and OFDMA ...................................................................

12

Figure 2.6: Time-domain structure

13

Figure 2.7: Time slot

 

13

Figure 2.8: Resource Block for

14

Figure 2.9: Reference Symbols in a subframe

15

Figure 2.10: Synchronization

 

16

Figure 2.11: L1/L2 control region

17

Figure 2.12: LTE rate control

18

Figure 2.13: Constellation diagrams

18

Figure 2.14: Scheduling units

 

19

Figure 2.15: Channel dependent scheduling

20

Figure 2.16: Soft combining

 

22

Figure 3.1: VoIP Codecs: AMR-NB and AMR-WB

24

Figure 3.2: VoIP packets and SID packets

25

Figure 3.3: Voice Quality (Source: ITU)

26

Figure 3.4: VoIP Traffic Model

26

Figure

3.5: Inverse

Discrete Transform

29

Figure

4.1: Jittered

packets

32

Figure 4.2: Resource scheduler

33

34

Figure

4.4: Jitter buffer

36

Figure

38

40

43

44

Figure 5.5: BLER curves obtained from SISO AWGN simulations for all 15

CQI values. From CQI 1 (leftmost) to CQI 15 (rightmost)

46

X

Figure

5.6: SNR-CQI mapping

46

Figure 5.7: Jitter behavior (SNR = 5 dB)

49

Figure 5.8: Jitter behavior (SNR = 8 dB)

51

52

53

55

58

Figure A.1: Operation regions

62

65

65

65

Figure A.5: Digital implementation of OFDM

67

Figure A.6: There is no intra-cell interference for

67

Figure A.7: Corruption due to time dispersion

68

Figure A.8: Cyclic-prefix insertion

68

XI

LIST OF TABLES

Table

2.1: LTE

Frequency bands

10

Table

2.2: LTE

14

Table 3.1: VoIP traffic model parameters

24

30

Table 5.1: Simulation parameters

40

45

49

Table 5.4: Cell jitter; SNR = 8 dB

50

52

XII

Chapter 1

INTRODUCTION

With more than 2 billion users around the world, there is no doubt that 2G and 3G UMTS cellular technologies are a complete success adopted by most countries and mobile network operators [1]. The first release, published in 1999, considered a circuit-switched (CS) data network, establishing a dedicated channel between transmitter and receiver. Later on, the standard considered a packet- switched (PS) cellular network known as HSPA, but still supporting CS services. The latest release of the UMTS wireless technology is the so-called 3GPP Long Term Evolution (LTE), an all-IP network.

Initiated in 2004 by the 3rd Generation Partnership Project (3GPP), the Long Term Evolution (LTE) project focused on enhancing the Universal Terrestrial Radio Access Network (UTRAN) and optimizing 3GPP’s radio access architecture. Targets were to have peak data rates of 100 Mbps in the downlink and 50 Mbps in the uplink. Orthogonal Frequency Division Multiple Access (OFDMA) and Single- Carrier Frequency Division Multiple Access (SC-FDMA) were selected as the multiple access technologies for the DL and UL respectively. The defined data modulation schemes are QPSK, 16QAM, and 64QAM 1 for both DL and UL. Furthermore, Multiple-Input Multiple-Output (MIMO) antenna technology is also supported, increasing capacity.

LTE is extremely flexible, using a number of defined channel bandwidths between 1.4 and 20 MHz (contrasted with UTRA’s fixed 5 MHz channels). To suit as many frequency band allocation arrangements as possible, both paired (FDD) and unpaired (TDD) band operation is supported. LTE can co-exist with earlier 3GPP radio technologies, even in adjacent channels, and calls can be handed over

to and from all 3GPP’s previous radio access technologies.

1 Optional for the uplink

1

Chapter 1. Introduction

2

The LTE architecture has been greatly simplified compared to past 3GPP's technologies, turning the hierarchical structure into a flat structure. All the user functionality is centralized in a single entity, the so-called evolved-NodeB. This design has several advantages: reduces the Round-Trip delay Time (RTT), scheduling decision are made faster (1 ms), and coordination among entities is improved. LTE is an all-IP network; in other words, only packet-switched services are supported.

While data traffic and its corresponding revenue are increasing, the voice service still makes the majority of operators’ income. Therefore, LTE is designed to support not only data services efficiently, but also good quality voice service with high efficiency. As LTE radio only supports packet services, the voice service will

also be Voice over IP (VoIP), not CS voice [2]. The use of VoIP instead of CS voice represents savings for operators, since the CS related part of the network will not be needed anymore. It is expected that VoIP can bring better capacity than CS voice due to more efficient utilization of resources.

  • 1.1 PROBLEM DESCRIPTION

Voice-over-IP represents savings for users and network operators. However, supporting VoIP in packet-switched mobile networks faces certain challenges due to its strict delay requirements and jitter sensibility.

Jitter is the variation of delay, where packets arrive at random times at the receiver. In other words, the kth packet is expected to arrive at a time but it is received at , where is jitter. When jitter is constant, it can be filtered out or compensated in a deterministic way. However, it often exhibits a random behavior [3]. Jitter results in speech intelligibility disruptions [4]; hence the end-to- end jitter has to be small enough not to be noticeable.

3GPP Long Term Evolution, an all-IP based network, is not exempt from jitter. Hence, research about this phenomenon is necessary to assure the feasibility of VoIP services over LTE.

Chapter 1. Introduction

3

  • 1.2 OBJECTIVE

In order to determine the feasibility of VoIP services over the LTE mobile networks, the purpose of this thesis is to analyze the jitter phenomenon. Particularly the impact on jitter caused by network congestion, retransmissions, and the modulation and coding scheme, for different radio channel conditions is studied. VoIP traffic, physical layer and MAC layer simulations are developed.

  • 1.3 JUSTIFICATION

In the packet-switched LTE network, services must be provided in a fast, efficient and reliable way, including services substituting their CS counterparts. Voice services will be offered in the form of Voice-over-IP. Since voice is the most widespread service, special care must be taken to assure a successful deployment of future LTE networks.

In the literature exists a variety of studies about VoIP over LTE [5] [6] [7]. Nevertheless, they mainly focus on capacity, coverage, or scheduling issues. However, there are not researches identifying the jitter phenomenon and its behavior. This thesis pretends to research jitter under diverse channel conditions, and its impact on the VoIP QoS.

  • 1.4 CONTRIBUTION

3GPP LTE is the next step towards 4G mobile communications with performance comparable to wire-line networks. Careful planning and design must be carried out to assure a successful deployment for both, users and network operators. LTE must be able to adapt to a variety of traffic such as data, voice, and video. Services currently provided through circuit-switched systems are expected to have a similar equivalent in LTE, an all-IP based network.

Voice is the most widespread service and represents the main revenue for network operators. Users expect at least the same Quality of Service provided by CS networks, while operators look for an increase in capacity and reduction of costs. Such objectives can be reached through Voice-over-IP (VoIP). This service has the following characteristics: Bursty low bitrate traffic, strict packet delay-based

Chapter 1. Introduction

4

QoS, and a high number of simultaneous users. Furthermore, VoIP is highly sensible to jitter.

Jitter is a common issue in packet-switched networks, where packets arrive at random times at the receiver. For voice services, it implies disruptions in speech intelligibility and poor QoS. This master thesis studies the jitter phenomenon for the LTE downlink, where bottlenecks arise naturally due to user queues, when it is operated under VoIP traffic. Specifically the impact on jitter due to network congestion, retransmissions, and different modulation and coding schemes, for diverse radio channel conditions, are analyzed.

  • 1.5 THESIS ORGANIZATION

The thesis structure is described now. An overview of LTE is presented in Chapter 2. The discussion begins exposing the architecture and design targets established by 3GPP. Then, the main technologies necessary to support the outstanding key features of LTE are introduced. Focus is made on OFDM, link adaptation, scheduling, and the HARQ retransmission scheme; key elements in the performance of VoIP.

The VoIP concept is analyzed in Chapter 3. First, the AMR voice codec used in LTE is described. Then, the quality criterion for VoIP services is presented. Further discussion focus on the VoIP traffic model and its implementation. Chapter 4 provides a description of the jitter phenomenon under LTE. In Chapter 5, the performance of LTE under VoIP traffic is tested through simulations for different channel conditions. It will be shown that LTE, as an all-IP network, will be able to offer VoIP services successfully as long as the number of users in the cell can be estimated correctly.

Final conclusions are presented in Chapter 6. Further research under the same line of study is also proposed. Finally, Appendix A offers an explanation of the multi-carrier and OFDM concepts.

Chapter 2

3GPP LONG TERM EVOLUTION

3GPP

Long

Term

Evolution

is

the

next

step

towards

4G

mobile

communications. Higher user data rates, increased capacity, and

reduced

delay/jitter, are some of the driving forces behind the evolution of the

Universal

Terrestrial Radio Access Network (UTRAN). This Chapter provides the background

necessary to comprehend presented.

LTE.

Both,

the

architecture

and

air

interface

are

  • 2.1 TH E STANDARDIZATION PROCESS

LTE consists of a series of standards and specifications defined by the 3rd

Generation Partnership

Project

(3GPP).

A

clear

understanding

of

the

standardization

process

shows,

for

example,

why

certain

air

interfaces

were

chosen as part of the standard instead of any other alternative. The process

is

described now:

 

Standardization

starts

with

the

requirement

phase,

where

the

standardization body decides what should be achieved with the standard. In the architecturephase, the main architecture is decided, i.e., how to meet the requirements. The interfaces and technologies are proposed.

For the detailed specification phase, the parameters for the architecture are

detailed. Finally, in the testing and verification phase, the interfaces are proved to work as expected.

This is an iterative process since any phase can directly affect the others.

5

Chapter 2. 3GPP Long Term Evolution

  • 2.2 DESIGN TARGETS

6

Initiated in 2004, the Long Term Evolution project focused on enhancing the Universal Terrestrial Radio Access Network (UTRAN) and optimizing 3GPP's radio access architecture. The design targets were [5]:

  • 1. Support scalable bandwidths

    • a) 1.25, 2.5, 5.0,

10.0

and 20

MHz 1 .

  • 2. Peak data rate that scales with system bandwidth.

    • a) DL (2 Ch. MIMO) peak rate of 100 Mbps in 20 MHz channel.

    • b) UL (1 Ch. TX) peak rate of 50 Mbps in 20 MHz channel.

  • 3. Supported antenna configurations.

    • a) DL: 4x2, 2x2, 1x2, 1x1.

    • b) UL: 1x2, 1x1.

  • 4. Spectrum efficiency.

    • a) DL: 5 bit/s/Hz (3 to 4 x HSPA

  • Rel. 6)

    • b) UL: 2.5 bit/s/Hz (2 to 3 x HSPA Rel. 6)

    • 5. User throughput.

      • a) 1.6 - 2.1 bit/s/Hz (3 to 4 x HSPA Rel. 6 )

    DL average:

    • b) UL average: 0.66 -1.0 bit/s/Hz (2 to 3 x HSPA Rel. 6)

    • c) edge 2 : 0.04 - 0.06 bit/s/Hz (2 to

    DL cell

    • d) edge : 0.02-0.03

    UL cell

    3 x HSPA bit/s/Hz (2 to 3 x HSPA

    Rel. 6) Rel. 6 )

    • 6. Latency

      • a) Control-plane < 50-100 ms: Delay generated for transiting from a non-active state to an active-state, where the terminal is able to send/receive data. There are two measures. The first one corresponds to the transition from a camped state, where the user terminal is unknown to the RAN (100 ms). The other measure is the transition from a dormant state, where the user terminal is known by the RAN but radio resources have been not assigned (50 ms).

      • b) User-plane < 5 ms: The user-plane latency requirement is expressed as the time it takes to transmit a small IP packet from the terminal to the RAN edge node or vice versa in an unloaded network.

  • 7. Mobility

    • 1 Final specifications consider 1.4, 3, 5, 10, 15 and 20 MHz.

    • 2 5 t h percentile - 95% of the users have better performance

  • Chapter 2. 3GPP Long Term Evolution

    7

    • a) Optimized for low speeds (<15 Km/h)

    • b) High performance at speeds up to 120 Km/h

    • c) Maintain link at speeds up to 350 Km/h (500 Km/h for certain frequencies)

    • 8. Coverage

      • a) Full performance up to 5 Km

      • b) Slight degradation at 5 Km - 30 Km

      • c) Operation up to 100 Km is not precluded by the standard

    Additionally, the related VoIP service requirements are:

    • 1. The E-UTRA should efficiently support various types of service. These must include currently available services like web-browsing, FTP, video-streaming or VoIP, and more advanced services (e.g. real-time video or push-to-talk) in the packet-switched domain.

    • 2. VoIP should be supported with at least as good radio backhaul efficiency and latency as voice over UMTS circuit-switched (CS) networks.

    • 3. Voice and other real-time services supported in the CS domain in Release 6 shall be supported by E-UTRAN via the packet switched domain with at least equal quality as supported by UTRAN (e.g. in terms of guaranteed bit rate) - over the whole speed range.

    2.3

    ARCHITECTURE

    The LTE architecture has been greatly simplified compared to past 3GPP's technologies. An all-IP flat architecture has been adopted to support the outstanding design targets. The main entities and interfaces are shown in Figure 2.1. A lot of functionalities, which in past 3GPP's architectures were placed in different entities, have been centralized in the eNodeB (base station). A new interface called X2 connects the eNodeBs, enabling direct communication between them. The E-UTRAN 3 is connected to the Evolved Packet Core (EPC) through the S1 interface which connects the eNodeBs to the Mobility Management Entities (MME) and the Serving Gateway (S-GW or SAE Gateway) through a many to many relationship.

    • 3 E-UTRAN is the official standard's name for LTE, the entire radio network.

    Chapter 2. 3GPP Long Term Evolution

    8

    MME/SAE Gateway MME/SAE Gateway eNB eNB eNB UE
    MME/SAE
    Gateway
    MME/SAE
    Gateway
    eNB
    eNB
    eNB
    UE

    E-UTRAN

    eNB: Enhanced Node B, or base station

    UE: User Equipment EPC: Evolved Packet Core

    o

    MME: Mobility

    Management

     

    Entity (Control Plane)

     

    o SAE: System Architecture Evolved (User Plane)

    E-UTRAN: Evolved Universal Terrestrial Radio Access Network

    Figure 2.1: LTE Architecture

    The radio protocol architecture of E-UTRAN is specified for the control-plane and user-plane. The control plane performs the radio resource control (RRC). The user-plane is divided in protocols with the following functions (see Figure 2.2):

    Packet

    Data

    Convergence

    Protocol

    (PDCP)

    performs

    IP

    header

    compression to reduce the number of bits over the air interface. It also

    handles the ciphering/deciphering and integrity functions.

     

    Radio

    Link

    Control

    (RLC)

    is responsible for segmentation/concatenation,

    RLC retransmission handling, and in-sequence delivery to higher layers.

    Medium Access Control (MAC) handles the HARQ retransmissions, scheduling for DL and UL, link adaptation, etc.

    Physical

    Layer

    (PHY)

    is

    responsible

    for coding/decoding,

    modulation/demodulation, multi-antenna mapping, and other typical physical

    layer functions.

     

    Chapter 2. 3GPP Long Term Evolution

    9

    IP packet IP packet User#i User #j SAE bearers PDCP PDCP Header compression Header compression Ciphering
    IP packet
    IP packet
    User#i
    User #j
    SAE
    bearers
    PDCP
    PDCP
    Header compression
    Header compression
    Ciphering
    Deciphering
    Radio
    bearers
    MAC
    RLC
    RLC
    Payload
    selection
    Segmentation, ARQ
    Reassembly, ARQ
    Priority
    Logical
    handling,
    channels
    payload
    selection
    MAC
    MAC multiplexing
    MAC demultiplexing
    Retransmission
    control
    Hybrid ARQ
    Hybrid-ARQ
    MAC scheduler
    Transport
    channel
    PHY
    PHY
    Coding
    Decoding
    Modulation
    scheme
    Modulation
    Demodulation
    Antenna and
    resource
    assignment
    Antenna and
    Antenna and
    resource mapping
    resource demapping
    eNodeB
    Mobile terminal (UE)
    Chapter 2. 3GPP Long Term Evolution IP packet IP packet User#i User #j SAE bearers PDCP

    Redundancy version

    Figure 2.2: DL LTE protocols

    • 2.4 LT E PHYSICAL LAYER

      • 2.4.1 BANDWIDTHS, FREQUENCY BANDS AND DUPLEXING

    LTE can be operated in different bandwidth sizes. The main reason for this is that the amount of spectrum available depends on the frequency band and the particular operator's situation. Originally it was stated in [6] as a list of LTE spectrum allocations from 1.25 to 20 MHz, although final specifications consider only 1.4, 3, 5, 10, 15 and 20 MHz.

    Pair

    and unpair

    spectrum,

    i.e.

    FDD and TDD

    modes,

    are

    supported.

    Frequency Division Duplex (FDD) entails that downlink and uplink take place in

    Chapter 2. 3GPP Long Term Evolution

    10

    different, sufficiently separated, frequency bands. Time Division Duplex (TDD) implies that downlink and uplink transmission take place in different non- overlapping slots. Figure 2.3 shows this concept.

    Figure 2.3: FDD and TDD

    Table 2.1: LTE Frequency bands

    Band

    UL Range (MHz)

    DL Range (MHz)

    Mode

    Main Region (s)

    • 1 1980

    1920

    -

    2110

    -

    2170

    FDD

    Europe, Asia

    1850

    • 2 1910

    -

    1930

    -

    1990

    FDD

    Americas (Asia)

    • 3 1785

    1710

    -

    1805

    -

    1880

    FDD

    Europe, Asia (Americas)

    1710

    • 4 1755

    -

    2110

    -

    2155

    FDD

    Americas

    • 5 849

    824

    -

    869

    -

    894

    FDD

    Americas

    • 6 840

    830

    -

    875

    -

    885

    FDD

    Japan

    • 7 2570

    2500

    -

    2620

    -

    2690

    FDD

    Europe, Asia

    880

    • 8 915

    -

    925

    -

    960

    FDD

    Europe, Asia

    • 9 1784.9

    1749.9

    -

    1844.9

    -

    1879.9

    FDD

    Japan

    • 10 1770

    1710

    -

    2110

    -

    2170

    FDD

    Americas

    • 11 1452.9

    1427.9

    -

    1475.9

    -

    1500.9

    FDD

    Japan

    698

    • 12 716

    -

    728

    -

    746

    FDD

    Americas

    • 13 787

    777

    -

    746

    -

    756

    FDD

    Americas

    • 14 798

    788

    -

    758

    -

    768

    FDD

    Americas

    704

    • 17 716

    -

    734

    -

    746

    FDD

    -

    • 33 1920

    1900

    -

    1900

    -

    1920

    TDD

    Europe, Asia (not Japan)

    • 34 2025

    2010

    -

    2010

    -

    2025

    TDD

    Europe Asia

    1850

    • 35 1910

    -

    1850

    -

    1910

    TDD

    • 36 1990

    1930

    -

    1930

    -

    1990

    TDD

    - -

    • 37 1930

    1910

    -

    1910

    -

    1930

    TDD

    -

    • 38 2620

    2570

    -

    2570

    -

    2620

    TDD

    Europe

    1880

    • 39 1920

    -

    1880

    -

    1920

    TDD

    China

    • 40 2400

    2300

    -

    2300

    -

    2400

    TDD

    Europe, Asia

    Chapter 2. 3GPP Long Term Evolution

    11

    LTE can be deployed in current cellular frequency bands (IMT-2000) and new frequency allocations as they become available. For instance, the 700 MHz frequency band previously used for analog television in the United States, it is now considered a potential band for LTE operation. The identified frequency bands by 3GPP are shown in Table 2.1 [6].

    2.4.2 OFDM

    Orthogonal Frequency Division Multiplexing has been chosen as the downlink transmission scheme f

    The subcarrier spacing is Δf = 15 KHz 4 . Likewise the OFDM symbol duration is Tu=1/Δf = 66.7 µs. Both co

    of an OFDM

    signal

    is given

    by the relationship BW = Nc. Δf, where Nc is the number of subcarr

    Δf = 1/T U

    Pulse shape

    T u =

    1/Δf

    a) Subcarrier spacing

    b) OFDM symbol duration

    Figure 2.4: OFDM concepts

    OFDM can be implemented digitally through the IDFT at the transmitter, and

    the

    DFT at the receiver.

    The FFT size

    N FFT

    should

    be preferably

    selected

    as N FFT = 2 n

    for some integer , so OFDM can be performed by means of the efficient radix-2 IFFT/FFT. The sampling rate is given as f s = Δf N FFT = 15,000 N FFT , thus the FFT size must be chosen such that the sampling theorem is satisfied. For

    example, if LTE is operated in a 20 MHz bandwidth, then NFFT = 2048 and the resulting sampling rate would be 30.72 MHz. Commonly, specifications express

    time units relative to the smallest time unit in LTE,T s ,

    corresponding to the sample

    period of a 20 MHz OFDM signal with a FFT size of 2048. That

    isTs

    =

    1/f

    =

    1/

    (15

    OFDM can be made completely resistant to multi-path delay spread. This is possible because the long symbols used for OFDM can be separated by a guard

    interval known as the cyclic

    prefix (CP), where

    the CP is a copy

    of the end of the

    4 7.5 KHz is also considered for multi-cell broadcast messages

    Chapter 2. 3GPP Long Term Evolution

    12

    OFDM symbol inserted at the beginning. The CP has been chosen to be slightly longer than the longest expected delay spread in the radio channel. LTE defines two cyclic prefix lengths, the normal CP and the extended CP. Normal CP is the expected operation mode for LTE, and its size has been set at ~4.7 us, enabling the system to cope with path delay variations up to about 1.4 Km. Since the length of an OFDM symbol is ~66.7 us, about a reduction of 6.6% in the effective data rates is experienced. Extended CP is designed to provide robustness against multi- path effect in larger cells, and for use with multi-cell broadcast messages. It provides protection for up to 10 Km delay spread with a capacity loss of 20%.

    Physical resources are organized in both, time and frequency domains. However, traditional OFDM systems split the system bandwidth into diverse frequency bands and assign them to the users indefinitely (similar to FDM). Hence radio resources may not be fully exploited. OFDM can be used as a robust Multiple Access scheme, the so-called Orthogonal Frequency-Division Multiple Access (OFDMA) which incorporates elements of TDMA. OFDMA allows subsets of the subcarriers to be allocated dynamically among the different users on the channel, for every unit of time. The result is a more robust system with increased capacity. This is due to the trunking efficiency of multiplexing low rate users and the ability to schedule users by time and frequency, which provides resistance to multi-path fading. A comparison between OFDM and OFDMA is shown in Figure 2.5.

    Subcarriers Subcarriers Used 1 Symbols (Tine) Symbols (Tine) User 2 User 3 OFDM OFDMA
    Subcarriers
    Subcarriers
    Used 1
    Symbols (Tine)
    Symbols (Tine)
    User 2
    User 3
    OFDM
    OFDMA

    Figure 2.5 OFDM and OFDMA

    A drawback of OFDM/OFDMA is that parallel transmission of multiple subcarriers leads to larger variations in the instantaneous transmission power. Thus, multi-carrier transmission will have a negative impact on the transmitter power-amplifier efficiency, implying increased transmitter power consumption and increased power-amplifier cost. This is specially an issue at the mobile terminal. Single-Carrier Frequency Division Multiple Access (SC-FDMA) was selected as the multiple access scheme for the uplink because it shares multi-carrier characteristics, while decreasing the Peak-to-Average Power Ratio (PAPR). A detailed description of SC-FDMA can be found in [1] [8].

    Chapter 2. 3GPP Long Term Evolution

    13

    2.5

    PHYSICAL

    RESOURCES

    LTE radio resources are organized as a bi-dimensional time-frequencygrid 5 . The largest unit of time in LTE is the 10 ms radio frame, which is further subdivided into ten 1 ms subframes, each of which is split into two 0.5 ms slots (Figure 2.6). Each slot comprises 7 OFDM symbols for normal cyclic prefix operation, and 6 for the extended cyclic prefix case (see Figure 2.7)

    One radio frame, T f =

    3 072 00xT s ;= 10 ms

    One slot, T slot

    = 15360xT s ; = 0.5 ms

    #0

    #1

    One subframe

    #2

    #3

    #18

    #19

    Figure 2.6: Time-domain structure

    The

    basic physical

    resource

    unit is composed by a subcarrier during an

    OFDM symbol, the so-called resource element(RE). Theoretically the scheduler could assign resources in a per-RE basis, increasing flexibility. Albeit, an overwhelming amount of overhead would be required to handle every single resource element, causing a reduction in power efficiency and user data rates.

    Normal CP

    Δf=15kHz

    5.2 µs

    160 samples

    LTE slot: 0.5 ms 15360 samples (Assumed Sampling Frequency

    f s = 30.72 MHz)

    4.7 µs

    144 samples

    Special OFDM symbol:

    71.9 µS

    2208 samples

    66.7 µs

    • 2048 samples

    16.7 µs

    512 samples

    OFDM symbol:

    71.3 µs

    2192 samples

    Extended CP Δf = 15kHz

    66.7 µs

    • 2048 samples

    OFDM symbol:

    83.3 µs

    2560 samples

    Figure 2.7: Time slot

    • 5 Spatial-domain is also considered for MIM O communication s (1 grid per antenna)

    Chapter 2. 3GPP Long Term Evolution

    14

    LTE defines a resource

    block(RB)

    as 12 consecutive subcarriers (180 KHz)

    during one time slot (0.5 ms). One slot consists of 7 or 6 OFDM symbols for
    during
    one time
    slot (0.5 ms). One slot consists
    of
    7
    or
    6
    OFDM
    symbols for
    normal-CP and extended-CP respectively 6 . Figure 2.8 illustrates a RB for normal-
    CP.
    One resource element
    QPSK. 2bits
    16QAM, 4bits
    64QAM, 6bits
    \f
    =
    15 kHz
    One resource block
    (12x7 = 84 resource elements)
    12 sub-carriers, 180 kHz

    Figure 2.8: Resource Block for normal-CP

    The number of RBs depends on the total system bandwidth, as shown in Table 2.2. Note that a frequency guard band is considered at the end of the LTE spectrum to avoid out-of-band emissions. For example, for a 5 MHz system bandwidth there are 25 RBs occupying a transmission bandwidth of 4.5 MHz.

    Table 2.2: LTE Resource configuration

    System BW

    1.4 MHz

    3 MHz

    5 MHz

    10 MHz

    15 MHz

    20 MHz

    Subframe-duration

    0.5 ms

     

    Subcarrier-spacing

    15 KHz

    Sampling frequency

    1.92 MHz

    3.84 MHz

    7.68 MHz

     

    15.36 MHz

    23.04 MHz

    30.72 MHz

    FFT size

    128

    256

    512

    1024

    1536

    2048

    Number of occupied

    72

    180

    300

    600

    900

    1200

    subcarriers Number of RBs

    6

    15

    25

    50

    75

    100

    Transmission BW

    1.08 MHz

    2.7 MHz

    4.5 MHz

     

    9 MHz

    13.5 MHz

    18 MHz

    (Efficiency)

    (77%)

    (90%)

    (90%)

    (90%)

    (90%)

    (90%)

    Number of OFDM symbols per subframe (Normal/Extended)

     

    7/6

    CP length

    Normal

    (4.7/9) x 6

    (4.7/18) x 6

    (4.7/36) x 6

     

    (4.7/72) x 6

    (4.7/108)

    x 6

    (4.7/144) x 6

    (us/samples)

    (5.2/10) x 1

    (5.2/20)

    x 1

    (5.2/40)

    x 1

    (5.2/80)

    x 1

    (5.2/120)

    x 1

    (5.2/160) x1

     

    Extended

    (16.7/32)

    (16.7/64)

    (16.7/128)

     

    (16.7/256)

    (16.7/384)

    16.7/512

    Chapter 2. 3GPP Long Term Evolution LTE defines a resource block (RB) as 12 consecutive subcarriers
    • 6 Hence, a RB is formed by 84 or 72 resource elements.

    Chapter 2. 3GPP Long Term Evolution

    15

     

    A resource

    block pairis

    formed by two consecutive time-domain RBs. That

    is,

    12 consecutive subcarriers

    (180 KHz) along

    1 subframe

    (1

    ms).

    A

    resource

    block pair is the minimum resource unit for scheduling purposes. The reason to

    define

    a

    RB

    in

    the

    first

    place

    is because

    certain control signals are

    mapped in

    particular RBs.

     
    • 2.6 PHYSICAL SIGNALS

      • 2.6.1 CELL-SPECIFIC DOWNLINK REFERENCE SIGNALS

    To carry out coherent demodulation of different downlink physical channels, a mobile terminal needs estimates of the downlink channel. More specifically, in case of OFDM transmission, the terminal needs an estimate of the complex channel of each subcarrier. One way to enable channel estimation in case of OFDM transmission is to insert known reference symbols into the OFDM time- frequency grid, the so-called Cell-specific downlink reference signals 7 . These reference signals are transmitted in every downlink subframe, and span the entire downlink cell bandwidth [8].

    LTE defines four reference symbols per resource block, separated in time

    and frequency as shown

    in Figure

    2.9. To estimate the channel

    over the

    entire

    time-frequency

    grid

    as well

    as

    reducing the noise

    in the channel

    estimates, the

    mobile terminal should carry out interpolation/ averaging over multiple

    reference

    symbols.

    Chapter 2. 3GPP Long Term Evolution A resource block pair is formed by two consecutive time-domain

    Figure 2.9: Reference Symbols in a subframe

    Chapter 2. 3GPP Long Term Evolution A resource block pair is formed by two consecutive time-domain

    7 Additionally, there also exist the UE-specific reference signals (to be used for an explicit UE) and MBSFN reference signals (for multi-cell broadcast).

    Chapter 2. 3GPP Long Term Evolution

    16

    • 2.6.2 SYNCHRONIZATION SIGNALS

    To

    assist the cell search, two special signals

    are transmitted on the

    LTE

    downlink, the Primary Synchronization Signal (PSS) and the Secondary Synchronization Signal (SSS). In case of FDD, the PSS is transmitted within the last symbol of the first slot of subframes 0 and 5, while the SSS is transmitted within the second last symbol of the same slot (i.e., just prior to the PSS). In the frequency domain they are transmitted on 62 subcarriers within 72 reserved subcarriers around DC subcarrier.

    10 ms radio frame subframe #0 #1 #2 #3 #4 #5 #6 #7 #8 #9 72
    10 ms radio frame
    subframe
    #0
    #1
    #2
    #3
    #4
    #5
    #6
    #7
    #8
    #9
    72 subcarrier or 6 resource blocks 1.08 Mhz
    012
    3
    4
    56
    0
    1
    2
    3
    4
    5
    6
    0
    1
    2
    3
    4
    012345
    6
    Systembandwidth
    OFDM symbol
    Secondary Synchronization Signal
    Primary Synchronization Signa
    Other resource allocation
    of variable bandwidth
    Figure 2.10: Synchronization Signals
    • 2.6.3 DOWNLINK L1/L2

    CONTROL SIGNALING

    To support the transmission of downlink and uplink transmissions, there is a need for downlink control signaling. This control signaling is often referred to as downlink L1/L2 control signaling, indicating that the corresponding information

    partly originates from the physical

    layer (Layer

    1) and partly from

    the MAC

    layer

    (Layer 2). The downlink control signaling corresponds to three physical channels:

    Physical

    Control

    Format

    Indicator

    Channel

    (PCFICH): Informs the terminal

    about the size of the control region (1, 2, or 3 OFDM symbols). There is one

    PCFICH in each cell.

     

    Physical

    Downlink

    Control

    Channel

    (PDCCH): It is used to signal downlink

    scheduling assignments and uplink scheduling grants. Each PDCCH carries signaling for a single terminal (or a group of terminals).

    Chapter 2. 3GPP Long Term Evolution

    17

    Physical

    Hybrid-ARQ

    Indicator

    Channel (PHICH): It is used to signal hybrid-

    ARQ ACKs in response to uplink transmissions. There are multiple PHICHs in each cell.

    The downlink L1/L2 control signaling is transmitted within the first part of each subframe. Thus each subframe is divided into a control region followed by a data region. The control region occupies 1, 2, or 3 OFDM symbols (up to 4 in case of a 1.4 MHz bandwidth). The size of the control region can be dynamically varied on a per-subframe basis to adjust to the instantaneous traffic situation. In case of a small number of users being scheduled in a subframe, the required amount of control signaling is small and a larger part of the subframe can be used for data transmission.

    One subframe

    Control region (1-3 OFDM symbols)

    Control

    Reference

    signaling

    symbols

    Figure 2.11: L1/L2 control region

    • 2.7 LINK ADAPTATION

    Link adaptation deals with how to set the transmission parameters of a radio link to handle variations of the radio-link quality. Unlike the early versions of UMTS, which used closed-loop power control to support CS services with a roughly constant data rate, link adaptation in LTE adjusts the transmitted information data rate dynamically (Figure 2.12). The radio-link data rate is controlled by adjusting the modulation scheme and/or the channel coding rate. In case of advantageous radio-link conditions a higher-order modulation, for example 16QAM or 64QAM, together with a high code rate is appropriate. Similarly, in case of poor radio-link conditions, QPSK and low-rate coding is used. For this reason, link adaptation by means of rate control is sometimes also referred to as Adaptive Modulation and Coding (AMC) [8].

    Chapter 2. 3GPP Long Term Evolution

    18

    Chapter 2. 3GPP Long Term Evolution Figure 2.12: LTE rate control A key issue in the

    Figure 2.12: LTE rate control

    A key issue in the development of LTE was if the RBs allocated to a user in a subframe should use the same Modulation and Coding Scheme (MCS), or whether the MCS should be frequency-dependent within each subframe. It was shown that the throughput gains for a frequency-dependent MCS does not justify the overhead required handling the RBs. Consequently, all the RBs assigned to a user within a subframe uses the same MCS, but it can change between subframes.

    • 2.7.1 MODULATION AND CODING SCHEME

    Modulation

     

    Digital

    modulation

    allows

    for

    higher

    data

    rates

    in

    a

    fixed

    bandwidth.

    According to the modulation scheme, one or more bits can be carried per

    modulation

    symbol.

    LTE defines

    the QPSK,

    16QAM

    and 64QAM

    modulation

    schemes which

    can carry

    2,

    4

    and

    6

    bits

    respectively,

    for both downlink and

    downlink. The constellation diagrams for these modulation schemes are shown in

    Figure 2.13.

    QPSK 16QAM 64QWI 2 bits/symbol 4 bits/symbol 6 bits/symbol Figure 2.13: Constellation diagrams
    QPSK
    16QAM
    64QWI
    2 bits/symbol
    4 bits/symbol
    6 bits/symbol
    Figure 2.13: Constellation diagrams

    Chapter 2. 3GPP Long

    Term Evolution

    19

    Coding Rate

    The channel coding scheme chosen for user data was turbo coding. Turbo codes have the benefits of their near-Shannon limit performance outweighing the associated costs of memory and processing requirements. A nominal rate-1/3 Turbo Code is used in LTE. Additional coding rates are obtained by puncturing/repetitions.

    2.8

    SCHEDULER

    The scheduler determines at a large extend the overall system performance, especially in highly loaded networks. The scheduler controls, for each instant of time, to which users the shared resources should be assigned. It also determines the data rate to be used for each link. LTE has access to both, time and frequency domains. Scheduling decisions are made every 1 ms (TTI) with a granularity of 180 KHz in the frequency domain, as illustared in Figure 2.14. This is commonly refered as a RB-pair, the scheduling unit in LTE. Gains in system capacity can be achieved if the channel conditions are taken into account in the scheduling decisions. This is known as channel-dependent scheduling (Figure 2.15).

    Chapter 2. 3GPP Long Term Evolution Coding Rate The channel coding scheme chosen for user data

    Figure 2.14: Scheduling units

    Channel-dependent

    scheduling

    allows

    for full

    flexibility

    in terms

    of the

    resources used and can handle large variations in the amount of data to transmit at the cost of the scheduling decision being sent on the control channel of each sub- frame for both, time and frequency domains. However, some services, most notably VoIP, are characterized by regularly occurring transmission of relatively small payloads. In order to avoid control channel limitations for VoIP traffic in LTE, the concept of semi-persistent scheduling was adopted.

    Chapter 2. 3GPP Long Term Evolution

    20

    Figure 2.15: Channel dependent scheduling

    The semi-persistent resource allocation method adopted in LTE is talk spurt based persistent allocation, and in DL direction the method works as follows. At the beginning of a talk spurt, a persistent resource allocation is done for the user and this dedicated time and frequency resource is used to transmit initial transmissions of VoIP packets. At the end of the talk spurt, persistent resource allocation is released. Thus, the released resource can be allocated to some other VoIP user [10]. Usually, only time-domain decisions are allowed.

    2.8.1

    CHANNEL-STATUS REPORTS

    An important part of the support for downlink scheduling is channel-status reports provided by terminals to the network, reports on which the base station can base its scheduling decisions. Although referred to as channel-status reports, what a terminal delivers to the network are not explicit reports of the downlink channel status. Rather, what the terminal delivers are recommendations on what transmission configuration and related parameters the network should use if/when transmitting to the terminal on the downlink shared channel. The terminal has typically based these recommendations on estimates of the instantaneous downlink channel conditions, thus the term channel-status report. The most important channel-status report is the so-called Channel Quality Indicator 8 [6]:

    The

    CQI

    provides

    the

    eNodeB

    information

    about

    the

    link

    adaptation

    parameters the UE can support at the time (taking into account the transmission

    mode, the UE receiver type, number of antennas and interference

    situation

    • 8 Rank Indicator (RI) and Pre-coding Matrix Indicator reports are used for MIMO schemes.

    Chapter 2. 3GPP Long Term Evolution

    21

    experienced at the given time) [2]. This report is represented by a CQI index which indicates the modulation scheme and coding rate that should, preferably, be used for the downlink transmission such that the BLER does not exceed 10%. Details about the CQI will be given in Chapter 5.

    2.9

    HYBRID-ARQ

    Transmissions over wireless channels are subject to errors, for example due to variations in the received signal quality. LTE implements error detection and correction through HARQ, which makes use of the following techniques:

    Forward Error Correction(FEC):

    The basic principle is to introduce redundancy

    in the transmitted signal. This is done by adding parity bits computed from the

    information bits. Thus, the number of bits transmitted over the channel is larger than the number of original information bits.

    Automatic Repeat Request(ARQ):

    The receiver uses an error-detecting code,

    typically a Cyclic Redundancy Code (CRC), to determine if the packet is in error or not. If the packet is error-free, a positive acknowledgment (ACK) is sent to the transmitter. On the other hand, if an error is detected a retransmission is requested through a negative acknowledgment (NACK).

    The LTE HARQ protocol uses multiple parallel stop-and-wait process (see [9] for details about this protocol). The number of hybrid ARQ processes directly affects the delay budget in the UE and the eNodeB. The smaller the number of hybrid ARQ processes the better from a round-trip time perspective but also the tighter the implementation requirements. Taking transmission, reception, and processing delays into account, it can be calculated that the retransmission of the packet is possible 8 ms after the previous transmission. Thus, the number of parallel HARQ processes is fixed to 8.

    LTE implements HARQ slightly different for the DL and UL.

    For Downlink, an adaptive asynchronous HARQ retransmission scheme is considered. Adaptive means that retransmission can take place using different MCS and distinct RBs. Asynchronous refers to the fact that retransmission can happen any time after the 8 ms retransmission delay.

    Chapter 2. 3GPP Long Term Evolution

    22

    For Uplink, non-adaptive synchronous HARQ is considered. Non-adaptive indicates that the retransmission must be completed using the same resource allocation and MCS as the original transmission. It requires less overhead than the HARQ for DL. Since it is synchronous, a retransmission occurs exactly 8 ms after the previous transmission.

    LTE supports soft combining. In traditional ARQ schemes, the erroneous data packets are discarded and a retransmission is requested. However, these packets still contains portions of useful data that could be used for future retransmissions. There are two types of soft combing:

    Chase Combining:

    Retransmissions use exactly the same coding scheme as

    the original transmission, as shown in Figure 2.16 a). The receiver

    uses

    maximum-ratio

    combining

    to

    increase

    the

    received

    for

    each

    retransmission.

    Incremental Redundancy:

    Every time a retransmission occurs, the coding

    rate is adapted to increase redundancy. Thus, additional to the received E b /N 0 gain, there is also a coding gain (Figure 2.16).

    a) Chase Combining

    b) IR Combining

    Figure 2.16: Soft combining

    Chapter 3

    VOICE-OVER-IP

    VoIP (Voice-over-IP) is simply the transmission of voice traffic over

    IP-based

    networks. The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption to voice networking.

    3.1

    VOIP CODECS

    G S M networks started with the Full

    rate (FR) speech codec and evolved to

    Enhanced Full Rate

    (EFR). The Adaptive Multi-Rate

    (AMR ) codec was added to

    3GPP Release 98 for GSM to enable codec rate adaptation to the radio conditions. A M R data rates range from 4.75 Kbps to 12.2 Kbps. The highest AM R rate is equal to the EFR . AM R uses a sampling rate of 8 KHz, which provides 300-3400 Hz audio bandwidth.

    The

    AMR-Wideband

    (AMR-WB)

    codec

    was added

    to

    3GPP

    Release

    5.

    AMR-WB uses a sampling rate of 16 kHz, which provides 50-7000 Hz audio bandwidth and substantially better voice quality and mean opinion score (MOS). As the sampling rate of AMR-WB is double the sampling rate of AMR, AM R is often referred to as AMR-NB (narrowband). AMR-WB data rates range from 6.6 Kbps to 23.85 Kbps. The typical rate is 12.65 Kbps, which is similar to the normal AM R of 12.2 Kbps. AMR-WB offers clearly better voice quality than AMR-NB with the same data rate and can be called wideband audio with narrowband radio transmission.

    The

    bandwidths

    for the AMR-NB

    and AMR-WB

    codecs

    are illustrated

    in

    Figure 3.1.a, while a comparison of these codec with audio bandwidth is exemplified in Figure 3.1.b. The smallest bit rates, 1.8 and 1.75 Kbps are used for the transmission of Silence Insertion Descriptor Frames (SID) [2].

    23

    Chapter 3.

    Voice-over-IP

    24

     

    1 [unian

    ear

    20-20000

    Hz

     

    Wideband

    AM R

    50-7000

    Hz

    Narrowband

    A M R

    300-3400

    Hz

    a) Codec bandwidth

    b) Audio bandwidth

    Figure 3.1: VoIP Codecs: AMR-NB and

    AMR-WB

    LTE specifications suggest that for VoIP performance tests, the AMR 12.2 codec should be used with the parameters described in Table 3.1. It provides a similar reference point for comparisons with actual cellular systems [8]. The resulting capacity of the AMR-NB12.2 Kbps would also be approximately valid for AMR-WB 12.65 Kbps.

    Table 3.1: VoIP traffic model Value

    parameters

    Parameter Voice Codec Encoder frame

    RTP AM R 12.2, Source Rate 12.2 Kbps 20 ms

    Voice activity factor (VAF)

    50%

    ( a = b =0.01 )

    SID payload

    SID Packet every 15 bytes (5 Bytes

    160 ms during silence Header)

    +

    Protocol overhead with compressed header Total voice payload on air interface

    10bits+padding (RTP-pre-header), 4 bytes (RTP/UDP/IP), 2 Bytes (RLC/Security), 16 bits (CRC) 40 Bytes

    There are two types

    of VoIP frames for the AM R

    12.2 codec: Voice

    frames

    Chapter 3.

    Voice-over-IP

    25

    Voice

    Frames.

    At a voice source

    rate of

    12.2 Kbps, a voice frame

    generated

    every 20 ms consists of 244 bits. The total protocol overhead per voice

    frame includes

    10-bits of RTP pre-header and 2-bits padding resulting in a

    total of 236 bits (32 bytes). Furthermore, a compressed RTP/UDP/IP

    header

    consisting of 4 bytes is attached to the packet

    making

    the total

    size

    of

    36

    bytes. With 2 bytes of Layer 2 overhead consisting of RLC and security header and 2 bytes CRC, the total VoIP payload size transmitted over the air interface becomes 320 bits (40 bytes) every 20 ms [7].

    SID Frames.

    These

    frames

    contain

    comfort

    noise,

    and

    connection

    information. SID frames are delivered during silent states every

    160 ms

    (8

    voice frames) consisting of 5 bytes of information and

    10 bytes

    of

    header.

    For the AM R 12.2 codec, 120 bits must be carried every 160 ms.

    VoIP packets on the active period

    SID packets on the silent period

    20 ms

    Headers

    160 ms

    Payload

    Figure 3.2: VoIP packets and SID packets

    • 3.2 QUALITY CRITERIA Considering the nature of radio communication

    it

    is

    not

    practical to aim

    for

    100% reception of all the VoIP packets in time. Instead, certain degree of missing packets can be tolerated without notably affecting the QoS perceived by the users. For voice services, usually 1 % is tolerated [7].

    The voice quality perception degrades as the end-to-end delay increases as depicted in Figure 3.3 [10]. LTE assumes a delay below 200 ms for mobile-to- mobile communication. Under this assumption, the delay budget available for radio interface is considered as 50 ms (from eNodeB to UE).

    Chapter 3.

    Voice-over-IP

    2 6

    100 Users Very Satisfied 9 0 Users Satisfied E-Model Rating R 80 Some Users Dissatisfied 70
    100
    Users
    Very Satisfied
    9 0
    Users
    Satisfied
    E-Model Rating R
    80
    Some Users
    Dissatisfied
    70
    Many Users
    Dissatisfied
    60
    Nearly All
    Users
    Dissatisfied
    50
    0
    10 0
    20 0
    30 0
    40 0
    50 0
    Mouth-to-Ear-Delay/ms

    Figure 3.3: Voice Quality (Source: ITU)

    The system capacity for VoIP is defined as the number of users supported in the cell when more than 95% of the users are satisfied. A VoIP user is satisfied if 98% of its packets experience a delay of less than 50 m s [8].

    3.3

    VOIP TRAFFIC MODEL

    Consider

    the two-state

    voice

    activity

    model

    shown

    in

    Figure

    3.4.

    The

    probability of transitioning from state 0 (silence or in active state) to state 1 (talking

    or active state) is a while the probability of staying in state 0 is

    (1- a ) .

    O n

    the

    other hand, the probability of transitioning from state 1 to state 0 is denoted b while the probability of staying in state 1 is (1- b ) . The updates are made at the speech encoder frame rate R= 1/T, where T is the encoder frame duration (20ms).

    a Silence Talking (1 - a) (State 0) (State 1) (1-b) Figure 3.4: VoIP Traffic Model
    a
    Silence
    Talking
    (1
    -
    a)
    (State 0)
    (State 1)
    (1-b)
    Figure 3.4: VoIP Traffic Model

    Chapter 3.

    Voice-over-IP

    27

    The

    probabilities

    of

    being

    in

    state

    0

    and

    state

    1

    denoted

    as

    and

    respectively:

     

    b

     

    P O

    =

     

    a

    +

    b

     

    a

     

    P 1

    =

    a

    + b

    The Voice Activity state 1:

    Factor is the probability

    of being

    in taking

    state, that is,

     

    a

     

    VAF

    =

    P 1

    =

     

    a

    + b

    The mean silence duration and mean talking duration in terms of number of voice frames 1 can be written as:

     

    1

    E

    [TS]

    =

     

    a

    1

    E [TS] =

     
     

    b

    The probabilities that silence duration or a talking duration is n voice frames long are given by:

     

    P T s

    =

    a( 1- a ) n-1 ,

    n

    =

    1,2 ,

    .

    (3.1)

     

    ( P Ts = b( 1-b) n-1 , n = 1,2

    )

    (3.2)

    Since

    the

    states

    transitions

    from

    state

    1

    to

    state

    0

    and

    vice

    versa

    are

    independent, the mean E[ T A E ]

    between active state entries is given simply

    by

    the

    sum of the

    mean time in each state, That is:

    1

    A voice frame duration is 20 ms.

    Chapter 3.

    Voice-over-IP

    28

    E[T A E ]

    = E[TS] +E[ T T ]

    • 1 1

    =

    a +

    b

    Accordingly, the mean rate of arrival R AE of transitions into the active state is given by

    R A

    E

    =

    1 / E[T A

    E ]

    • 3.4 GENERATING VOIP TRAFFIC

    The

    VoIP

    model

    traffic

    ca n serve

    as

    a

    guide

    on the number

    of resource

    allocation requests. Likewise, it can also be used to generate VoIP traffic as will be

    described in this section.

    The number of packets in silence an d talking states must be determined to

    simulate VoIP traffic. Th e discrete inverse transform (DIT) would be used for such

    purpose [12]. Consider the probability mass functions

    in Equation 3.1 an d Equation

    3.2

    describing

    the probability

    of

    a

    user

    staying

    in talking

    and silence

    duration

    respectively. For any probability mass function the following condition must hold:

    P(N

    =

    n i ) = p i ,

    i

    =

    1,

    .

    .

    ., Σ p i

    i

    =

    1

    To

    generate

    N,

    the discrete

    random

    variable

    representing

    the

    number

    packets in any state, generate a uniform random number u an d set:

    That is,

    N

    =

    n i

    if

    p

    +

    ...+

    p

    i

    -

    1

    +p i

    N

    =

    n 1 , n 2 , n 3 ,

    n t

    p 1

    +

    u

    p 1 p 1 <up 1 +p 2

    Pi + P2
    Pi + P2

    <

    u

    p i + p 2 + p3

    + -p i _ 1

    + Pi- 1 <u<p 1

    +

    p i

    Chapter 3.

    Voice-over-IP

    29

    Because

    is uniform distributed on

    (0,

    1), it follows that for

    (0< a <b<1) :

    P(a<

    u

    b)=b-a

    Consequently,

    P

    i-1

    Σ pj < u

    j=1

    Σ

    pj

    i

    j=1

    =

    Pi

    which proves that N has the desired probability mass function.

     

    Graphically,

    it represents

    a mapping

    between

    the CDF

    and the number

    of

    packets. For example, consider Equation 3.1 for the talking state case. The CDF is

    easily obtained and plotted

    in

    Figure 3.5.

    Since

    u

    is

    a

    uniform

    number

    between

    0

    and

    1, it represents

    the

    probability that the duration

    in talking state

    is n packets.

    Find the value n that produces u. Finally set N=n.

    Figure 3.5: Inverse Discrete Transform

    • 3.5 VOIP TRAFFIC SIMULATOR

    The VoIP traffic simulator was developed applying the VoIP traffic model concepts; its respective parameters were initialized according to the 3GPP recommendations in Table 3.1 [8]. Figure 3.4 depicts the theoretical and simulated

    Chapter 3.

    Voice-over-IP

    30

    probabilities that a talking subframe is n frames long. Since the voice activity factor

    is 50%, the silence probability mass function will be exactly the same. The obtained

    results

    of

    the

    simulation

    are

    displayed

    in

    Table

    2.1. Simulation

    and

    theoretical

    results do agree, proving the validity of the results

    Chapter 3. Voice-over-IP 30 probabilities that a talking subframe is n frames long. Since the voice

    Figure 3.3: VoIP Traffic Model

    Parameter

    Table 3.2: VoIP Traffic Simulation Results Value

    Voice frame periodicity

    (VAF)

     

    20

    ms

    SID frame periodicity

     

    160 ms

    Voice Activity Factor

     

    0.5

    Probability

    of staying

    in

    state

    1

    ( 1

    -b

    )

    0.99

    Probability

    of staying

    in

    state

    0

    ( 1

    -

    a)

    0.99

    Probability

    of transitioning

    from

    state

    1 to

    0

    (b)

    0.01

    Probability

    of transitioning from state 0 to

    1

    ( a)

    0.01

    Mean talking duration E[T T ]

     

    100 voice packets (2 sec)

    Mean

    silence duration E[T S ]

    100

    voice packets (2 sec)

    Mean

    successive transitions into the 1 state E[T A E ]

    200 voice

    packets (4 sec)

    Mean

    rate of arrivals into the active state( R A E )

    0.25

    talk-spurts/second

    Chapter 4

    INTRODUCTION TO JITTER

    In wire-line systems, channels are typically clean and end-to-end transmissions are almost error-free, requiring no retransmissions. However, a wireless channel could be unfavorable, resulting in bit errors and corrupted packets. Packets may have to be retransmitted multiple times to ensure successful reception, and the number of retransmissions depends on the dynamic radio channel conditions. This could introduce significant delay variations. Furthermore, unlike the circuit channels which have a dedicated fixed bandwidth for continuous transmission, packet transmissions are typically bursty and share a common channel that allows multiplexing for efficient channel utilization. This operation results in loading-dependent delay (jitter).

    • 4.1 TH E JITTER CONCEPT

    In a packet-switched network, such as LTE, data is sent by the transmitter

    as a continuous stream of packets spaced evenly apart. However, due to network

    congestion, retransmissions, etc., this steady

    stream

    could

    vary

    over

    time.

    Consider Figure 4.1 showing the jitter concept. A transmitter sends packets

    sequentially

    and

    periodically

    every kT seconds,

    where k = 0, 1, 2,

    is

    the

    number

    t k = kT + t p , where t p is the propagation time. However, as the kth packet travels

     

    along the PS network it suffers

    delay variations t n , called jitter. Then, the

    packet

    would be received at a time t k = kT +t p + t.

    Jitter is defined as the variation in delay that the receiver experiences, or

    alternatively, a variation in the delivery

    rate. Jitter can

    be defined

    by

    using the

    known arrival intervals (20 ms for VoIP), and subtracting the consecutive delays of packets that were not lost. When jitter is a constant is can be filtered out or

    31

    Chapter 4. Introduction to Jitter

    32

    compensated in a deterministic way. However, often exhibits a random behavior

    [3].

    Transmitter

     

    T

    2T

    3T

    Packet 1

    Packet 2

    Packet 3

     

    Time

    Receiver

     

    Packet 1

    Packet 2

    Packet 3

    tp

    tp

    tp

    tp

     

    T1 = 0

    T2

    T3

    tp - Propagation time

     

    T - Packet periodicity

     

    Ti - Jitter

    4T

    Packet 4

    T4

    Figure 4.1: Jittered packets

    Packet 4

    Jitter is a source of speech intelligibility disruptions [4]; the end-to-end jitter has to be small enough not to be noticeable. Delay and jitter are not the same concept. However, as will be explained, there is a trade-off between jitter and delay, and that is the reason why commonly both terms are used.

    • 4.2 LT E JITTER SOURCES 4.2.1

    SCHEDULER BUFFER

    The generic function of a resource scheduler is to schedule data to a set of UEs on a shared set of physical resources. In general, scheduler algorithms can make use of two types of measurement information, channel-state information and traffic measurements (volume and priority). The algorithm used by the resource scheduler is closely related with the adaptive and modulation scheme and the retransmission protocol (Hybrid-ARQ).

    Chapter 4. Introduction to Jitter

    33

    As network load increases, the physical resources will become scarce and users will be placed in the scheduler buffer. The queues dynamics, which impact the throughput, delay and jitter characteristics of the link seen by the application, depend heavily on network congestion and the MCS (packet sizes). These concepts are shown in Figure 5.2.

    Figure 4.2: Resource scheduler

    Another jitter source in the scheduler is due to packet fragmentation. When a packet cannot be sent in one resource scheduling unit, it will be segmented, until the packet has been completely transmitted.

    • 4.2.2 HAR Q RETRANSMISSIONS

    Due to unfavorable instantaneous channel conditions, packets could arrive corrupted at the receiver. Consequently a retransmission would be requested. Consider Figure 4.3, showing a single HARQ process for the downlink direction. At the eNodeB, a packet is sent in the subframe n and received after a propagation time t p , in the subframe n of the receiver. Then the UE will attempt to decode the received signals during a time t UE , possible after soft combining. In the subframe n + 4 of the receiver, an ACK/NACK is sent by the uplink channel to the eNodeB. The eNodeB, processes this information during time t eNB and retransmits the packet at subframe n + 8. Thus, a retransmission occurs at least 8 ms after the previous transmission. For VoIP frames up to 6 retransmissions could be possible for a delay bound of 50 ms.

    Chapter 4. Introduction to Jitter

    34

    Figure 4.3: HARQ retransmission for the DL

    • 4.2.3 RADIO LINK CONTROL FUNCTIONS

    In LTE, retransmissions of missing or erroneous data units are handled

    primarily by the hybrid-ARQ mechanism in the MAC layer, complemented by the retransmission functionality of the RLC protocol. The reasons for having a two-level retransmission structure can be found in the trade-off between fast and reliable

    feedback of the status reports. A feedback

    error rate of around 1 % results common

    for hybrid-ARQ processes. Such an error rate is in many cases far too high; high

    data rates with TCP may require virtually error-free delivery of packets to the TCP protocol layer [8].

     

    The

    RLC protocol can be operated

    in three

    modes

    to

    adapt to the type

    of

    transmission:

     

    Transparent

    Mode

    (TM) bypasses the RLC functions. No retransmissions,

    no segmentation/reassembly, and no in-sequence delivery take place. This configuration is used for broadcast channels where the information should reach multiple users.

    Unacknowledged

    Mode

    (UM)

    supports

    segmentation/reassembly

    and

    in-

    sequence delivery, but not retransmissions. This mode is used when error- free delivery is not required, for example VoIP.

    Acknowledged Mode (AM) is the main operation mode for TCP/IP data transmission. Segmentation/reassembly, in-sequence delivery and retransmission of erroneous data are also supported.

    VoIP services are operated in unacknowledged mode, where certain packet lost is tolerated. That is, jitter produced by the RLC retransmission scheme is not

    Chapter 4. Introduction to Jitter

    35

    an issue. However, in-sequence delivery of packets is still a requirement which will introduce a fixed delay, the so-called jitter buffer size. This will be clarified in the next section.

    • 4.2.4 MOBILITY

    As a mobile terminal moves through the network, the propagation time will change. Furthermore, the link adaptation algorithm will adapt the transmission parameters, e.g. the modulation and coding scheme, to the radio channel conditions. This will result in fluctuating data rates.

    Mobility

    also

    implies

    handovers

    among

    sectors

    or

    cells.

    The

    handover

    process will introduce unexpected variations in delay, due to the unpredictable coordination time between eNodeBs.

    • 4.2.5 OTHER JITTER

    SOURCES

    There exist other jitter sources which are not considered for this thesis since they are negligible for network-level simulations, or because they are beyond the scope of this research. Some examples include:

    External Networks: If packets come from another packet-switched network such

    as Internet, they could already be jitter. Routing and buffers are typically the reasons behind this undesired impairment. Hardware: Because electronic circuits are not completely synchronized, small variation in the encoding times, processing speeds, etc., are present.

    4.3

    JITTER MANAGEMENT

    Although a jitter-free packet-switched network is unfeasible, the jitter phenomenon can be contained. Even real-time services can tolerate certain jitter as long as it is below an established delay bound. Some techniques to deal with jitter are presented in this section.

    • 4.3.1 JITTER BUFFER

    Since packets arrive at their destination at random times due to jitter, the user may perceive anomalies in the stream, experienced as static, strange noise

    effects, garbled words or even

    missed words or syllables.

    In

    Figure

    4.4

    a jitter

    Chapter 4. Introduction to Jitter

    36

    buffer is depicted, which is a common method used at the receiver side to counteract variations in the delay. Basically, incoming packets are stored for a predefined period of time and then they are played-out at the expected rate (often constant). In other words, the receiver holds the first packet in a buffer for a while before sending it to the voice decoder. The amount of time a packet is hold is known as jitter buffer size.

    Figure 4.4: Jitter buffer

    If the jitter buffer size is too short, packets will still experience jitter. On the other hand, if it is too long the delay will cause packet lost, and degradation for sensitive-delay applications such as interactive applications and real-time-services such as VoIP. Because of the jitter buffer, there is a trade-off between delay and jitter

    LTE assumes an end-to-end delay below 200 ms. Under this assumption, the delay budget available for radio interface is 50 ms (from eNodeB to UE). There is discussion about the ideal buffer size; even adaptive jitter buffers which change the size dynamically. For the discussion of this thesis, the buffer size will be assumed as 50 ms. Hence, if the jitter caused by the LTE air interface is less than 50 ms, then the jitter buffer will be exchanged for delay.

    4.3.2

    SCHEDULER STRATEGIES

    The LTE Scheduler can optimize over several metrics. However, a critical factor which must always be present is the queue dynamics. The proposed LTE scheduler used for this research takes decisions based on reducing delay - consequently jitter. Chapter 5 will give details about the proposed scheduler.

    Chapter 5

    VOIP JITTER IN LTE

    Previous Chapters provided to the reader the knowledge necessary to comprehend the jitter phenomenon in LTE networks, especially for VoIP services. From now on, VoIP jitter is quantified by means of physical layer and MAC layer simulations.

    • 5.1 SIMULATION SCENARIO

    Consider the simulation scenario shown in Figure 5.1 for the LTE downlink. It is assumed that VoIP packets and SID packets arrive to the eNodeB with no jitter. According to the traffic load and the instantaneous radio channel conditions, packets could be scheduled immediately or placed in the scheduler buffer until resources become available. Let us consider the first case. Once a packet has been selected to be transmitted, the appropriated modulation and coding scheme is determined by the base station. To help on the scheduler decisions, every UE sends periodical reports about the radio channel conditions (1 ms for this simulation). Feedback information is sent in the form of CQIs, indicating the maximum MCS that can be supported by the UE, such that the Block Error Rate does not exceed 10%. Once the scheduler has determined the resource allocation and MCS, the eNodeB notifies to the UE through the Physical Downlink Control Channel (PDCCH). Then the packet is sent through the Physical Downlink Shared- Channel (PDSCH), while storing a copy in the HARQ buffer in case a retransmission is requested.

    The UE will try to decode the information sent by the eNodeB. According to

    the result, the mobile terminal could

    send an

    ACK if the packet was received

    successfully or an NACK if the erroneous packet could not be recovered by the FEC. A HARQ retransmission takes at least 8 ms for the DL. Note that a

    37

    Chapter 5.VoIP Jitter in LTE

    38

    retransmission could also be placed in the scheduler queue in case of high traffic conditions.

    eNodeB Channel UEs UE1 Scheduler Buffer RBs, MCS CQI, ACK/NACK UE1 RBs, MCS VoIP Traffic UE2
    eNodeB
    Channel
    UEs
    UE1
    Scheduler Buffer
    RBs, MCS
    CQI, ACK/NACK
    UE1
    RBs, MCS
    VoIP Traffic
    UE2
    UE2
    Generator
    CQI, ACK/NACK
    UEn
    Hybrid-ARQ Buffer
    RBs, MCS
    CQI, ACK/NACK
    UEn

    Figure 5.1: Simulation Scenario

    The jitter sources under this scenario are:

    • 1. Scheduler buffer: Due to network congestion, some packets will have to wait in the queue. Besides, packets could also be placed in the scheduler buffer if the destination is experiencing poor channel conditions.

    • 2. Packet fragmentation: VoIP traffic is characterized by low bitrates, what implies small packets. However, in poor radio conditions a low order, more robust MCS would be chosen, so a single RB-pair could even not be enough to send a complete VoIP packet. The packet would be fragmented, requiring more than 1 TTI to be completely transmitted. The VoIP packet cannot be used by the voice decoder until it is complete.